Add the rest of the juce modules

source/modules/juce_audio_devices.h

@@ -0,0 +1,28 @@
+/*
+ * Carla Juce setup
+ * Copyright (C) 2013 Filipe Coelho <falktx@falktx.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as
+ * published by the Free Software Foundation; either version 2 of
+ * the License, or any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * For a full copy of the GNU General Public License see the doc/GPL.txt file.
+ */
+
+#ifndef CARLA_JUCE_AUDIO_DEVICES_H_INCLUDED
+#define CARLA_JUCE_AUDIO_DEVICES_H_INCLUDED
+
+#include "juce_events.h"
+#include "juce_audio_basics.h"
+#include "juce_audio_formats.h"
+
+#include "juce_audio_devices/AppConfig.h"
+#include "juce_audio_devices/juce_audio_devices.h"
+
+#endif // CARLA_JUCE_AUDIO_DEVICES_H_INCLUDED

source/modules/juce_audio_devices/AppConfig.h

@@ -0,0 +1,79 @@
+/*
+  ==============================================================================
+
+   Build options for juce_audio_devices static library
+
+  ==============================================================================
+*/
+
+#ifndef CARLA_JUCE_AUDIO_DEVICES_APPCONFIG_H_INCLUDED
+#define CARLA_JUCE_AUDIO_DEVICES_APPCONFIG_H_INCLUDED
+
+#include "../juce_events/AppConfig.h"
+#include "../juce_audio_basics/AppConfig.h"
+#include "../juce_audio_formats/AppConfig.h"
+
+//=============================================================================
+/** Config: JUCE_ASIO
+    Enables ASIO audio devices (MS Windows only).
+    Turning this on means that you'll need to have the Steinberg ASIO SDK installed
+    on your Windows build machine.
+
+    See the comments in the ASIOAudioIODevice class's header file for more
+    info about this.
+*/
+#if WINDOWS
+ #define JUCE_ASIO 1
+#else
+ #define JUCE_ASIO 0
+#endif
+
+/** Config: JUCE_WASAPI
+    Enables WASAPI audio devices (Windows Vista and above).
+*/
+#define JUCE_WASAPI 0
+
+/** Config: JUCE_DIRECTSOUND
+    Enables DirectSound audio (MS Windows only).
+*/
+#if WINDOWS
+ #define JUCE_DIRECTSOUND 1
+#else
+ #define JUCE_DIRECTSOUND 0
+#endif
+
+/** Config: JUCE_ALSA
+    Enables ALSA audio devices (Linux only).
+*/
+#if LINUX
+ #define JUCE_ALSA 1
+ #define JUCE_ALSA_MIDI_INPUT_NAME  "Carla"
+ #define JUCE_ALSA_MIDI_OUTPUT_NAME "Carla"
+ #define JUCE_ALSA_MIDI_INPUT_PORT_NAME  "Midi In"
+ #define JUCE_ALSA_MIDI_OUTPUT_PORT_NAME "Midi Out"
+#else
+ #define JUCE_ALSA 0
+#endif
+
+/** Config: JUCE_JACK
+    Enables JACK audio devices (Linux only).
+*/
+#if LINUX
+ #define JUCE_JACK 1
+ #define JUCE_JACK_CLIENT_NAME "Carla"
+#else
+ #define JUCE_JACK 0
+#endif
+
+//=============================================================================
+/** Config: JUCE_USE_CDREADER
+    Enables the AudioCDReader class (on supported platforms).
+*/
+#define JUCE_USE_CDREADER 0
+
+/** Config: JUCE_USE_CDBURNER
+    Enables the AudioCDBurner class (on supported platforms).
+*/
+#define JUCE_USE_CDBURNER 0
+
+#endif // CARLA_JUCE_AUDIO_DEVICES_APPCONFIG_H_INCLUDED

source/modules/juce_audio_devices/Makefile

@@ -0,0 +1,88 @@
+#!/usr/bin/make -f
+# Makefile for juce_audio_devices #
+# ------------------------------- #
+# Created by falkTX
+#
+
+include ../../Makefile.mk
+
+# --------------------------------------------------------------
+
+BUILD_CXX_FLAGS += -I.
+
+ifeq ($(MACOS),true)
+BUILD_CXX_FLAGS += -objc++
+OBJS         = juce_audio_devices.mm.o
+OBJS_posix32 = juce_audio_devices.mm.posix32.o
+OBJS_posix64 = juce_audio_devices.mm.posix64.o
+else
+OBJS         = juce_audio_devices.cpp.o
+OBJS_posix32 = juce_audio_devices.cpp.posix32.o
+OBJS_posix64 = juce_audio_devices.cpp.posix64.o
+endif
+
+OBJS_win32 = juce_audio_devices.cpp.win32.o
+OBJS_win64 = juce_audio_devices.cpp.win64.o
+
+# --------------------------------------------------------------
+
+all: ../juce_audio_devices.a
+
+posix32: ../juce_audio_devices.posix32.a
+posix64: ../juce_audio_devices.posix64.a
+win32:   ../juce_audio_devices.win32.a
+win64:   ../juce_audio_devices.win64.a
+
+# --------------------------------------------------------------
+
+../juce_audio_devices.a: $(OBJS)
+	$(AR) rs $@ $^
+
+../juce_audio_devices.posix32.a: $(OBJS_posix32)
+	$(AR) rs $@ $^
+
+../juce_audio_devices.posix64.a: $(OBJS_posix64)
+	$(AR) rs $@ $^
+
+../juce_audio_devices.win32.a: $(OBJS_win32)
+	$(AR) rs $@ $^
+
+../juce_audio_devices.win64.a: $(OBJS_win64)
+	$(AR) rs $@ $^
+
+../juce_audio_devices.dll: $(OBJS)
+	$(CXX) $^ -shared $(LINK_FLAGS) -o $@
+
+../juce_audio_devices.dylib: $(OBJS)
+	$(CXX) $^ -dynamiclib $(LINK_FLAGS) -o $@
+
+../juce_audio_devices.so: $(OBJS)
+	$(CXX) $^ -shared $(LINK_FLAGS) -o $@
+
+# --------------------------------------------------------------
+
+%.cpp.o: %.cpp
+	$(CXX) $< $(BUILD_CXX_FLAGS) -c -o $@
+
+%.mm.o: %.mm
+	$(CXX) $< $(BUILD_CXX_FLAGS) -c -o $@
+
+%.posix32.o: %
+	$(CXX) $< $(BUILD_CXX_FLAGS) $(32BIT_FLAGS) -c -o $@
+
+%.posix64.o: %
+	$(CXX) $< $(BUILD_CXX_FLAGS) $(64BIT_FLAGS) -c -o $@
+
+%.win32.o: %
+	$(CXX) $< $(BUILD_CXX_FLAGS) $(32BIT_FLAGS) -c -o $@
+
+%.win64.o: %
+	$(CXX) $< $(BUILD_CXX_FLAGS) $(64BIT_FLAGS) -c -o $@
+
+# --------------------------------------------------------------
+
+clean:
+	rm -f *.o ../juce_audio_devices*.a
+
+debug:
+	$(MAKE) DEBUG=true

source/modules/juce_audio_devices/audio_cd/juce_AudioCDBurner.h

@@ -0,0 +1,169 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#ifndef JUCE_AUDIOCDBURNER_H_INCLUDED
+#define JUCE_AUDIOCDBURNER_H_INCLUDED
+
+#if JUCE_USE_CDBURNER || DOXYGEN
+
+
+//==============================================================================
+/**
+*/
+class AudioCDBurner     : public ChangeBroadcaster
+{
+public:
+    //==============================================================================
+    /** Returns a list of available optical drives.
+
+        Use openDevice() to open one of the items from this list.
+    */
+    static StringArray findAvailableDevices();
+
+    /** Tries to open one of the optical drives.
+
+        The deviceIndex is an index into the array returned by findAvailableDevices().
+    */
+    static AudioCDBurner* openDevice (const int deviceIndex);
+
+    /** Destructor. */
+    ~AudioCDBurner();
+
+    //==============================================================================
+    enum DiskState
+    {
+        unknown,                /**< An error condition, if the device isn't responding. */
+        trayOpen,               /**< The drive is currently open. Note that a slot-loading drive
+                                     may seem to be permanently open. */
+        noDisc,                 /**< The drive has no disk in it. */
+        writableDiskPresent,    /**< The drive contains a writeable disk. */
+        readOnlyDiskPresent     /**< The drive contains a read-only disk. */
+    };
+
+    /** Returns the current status of the device.
+
+        To get informed when the drive's status changes, attach a ChangeListener to
+        the AudioCDBurner.
+    */
+    DiskState getDiskState() const;
+
+    /** Returns true if there's a writable disk in the drive. */
+    bool isDiskPresent() const;
+
+    /** Sends an eject signal to the drive.
+        The eject will happen asynchronously, so you can use getDiskState() and
+        waitUntilStateChange() to monitor its progress.
+    */
+    bool openTray();
+
+    /** Blocks the current thread until the drive's state changes, or until the timeout expires.
+        @returns    the device's new state
+    */
+    DiskState waitUntilStateChange (int timeOutMilliseconds);
+
+    //==============================================================================
+    /** Returns the set of possible write speeds that the device can handle.
+        These are as a multiple of 'normal' speed, so e.g. '24x' returns 24, etc.
+        Note that if there's no media present in the drive, this value may be unavailable!
+        @see setWriteSpeed, getWriteSpeed
+    */
+    Array<int> getAvailableWriteSpeeds() const;
+
+    //==============================================================================
+    /** Tries to enable or disable buffer underrun safety on devices that support it.
+        @returns    true if it's now enabled. If the device doesn't support it, this
+                    will always return false.
+    */
+    bool setBufferUnderrunProtection (bool shouldBeEnabled);
+
+    //==============================================================================
+    /** Returns the number of free blocks on the disk.
+
+        There are 75 blocks per second, at 44100Hz.
+    */
+    int getNumAvailableAudioBlocks() const;
+
+    /** Adds a track to be written.
+
+        The source passed-in here will be kept by this object, and it will
+        be used and deleted at some point in the future, either during the
+        burn() method or when this AudioCDBurner object is deleted. Your caller
+        method shouldn't keep a reference to it or use it again after passing
+        it in here.
+    */
+    bool addAudioTrack (AudioSource* source, int numSamples);
+
+    //==============================================================================
+    /** Receives progress callbacks during a cd-burn operation.
+        @see AudioCDBurner::burn()
+    */
+    class BurnProgressListener
+    {
+    public:
+        BurnProgressListener() noexcept {}
+        virtual ~BurnProgressListener() {}
+
+        /** Called at intervals to report on the progress of the AudioCDBurner.
+
+            To cancel the burn, return true from this method.
+        */
+        virtual bool audioCDBurnProgress (float proportionComplete) = 0;
+    };
+
+    /** Runs the burn process.
+        This method will block until the operation is complete.
+
+        @param listener             the object to receive callbacks about progress
+        @param ejectDiscAfterwards  whether to eject the disk after the burn completes
+        @param performFakeBurnForTesting    if true, no data will actually be written to the disk
+        @param writeSpeed           one of the write speeds from getAvailableWriteSpeeds(), or
+                                    0 or less to mean the fastest speed.
+    */
+    String burn (BurnProgressListener* listener,
+                 bool ejectDiscAfterwards,
+                 bool performFakeBurnForTesting,
+                 int writeSpeed);
+
+    /** If a burn operation is currently in progress, this tells it to stop
+        as soon as possible.
+
+        It's also possible to stop the burn process by returning true from
+        BurnProgressListener::audioCDBurnProgress()
+    */
+    void abortBurn();
+
+private:
+    //==============================================================================
+    AudioCDBurner (const int deviceIndex);
+
+    class Pimpl;
+    friend struct ContainerDeletePolicy<Pimpl>;
+    ScopedPointer<Pimpl> pimpl;
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AudioCDBurner)
+};
+
+
+#endif
+#endif   // JUCE_AUDIOCDBURNER_H_INCLUDED

source/modules/juce_audio_devices/audio_cd/juce_AudioCDReader.cpp

@@ -0,0 +1,57 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#if JUCE_USE_CDREADER
+
+int AudioCDReader::getNumTracks() const
+{
+    return trackStartSamples.size() - 1;
+}
+
+int AudioCDReader::getPositionOfTrackStart (int trackNum) const
+{
+    return trackStartSamples [trackNum];
+}
+
+const Array<int>& AudioCDReader::getTrackOffsets() const
+{
+    return trackStartSamples;
+}
+
+int AudioCDReader::getCDDBId()
+{
+    int checksum = 0;
+    const int numTracks = getNumTracks();
+
+    for (int i = 0; i < numTracks; ++i)
+        for (int offset = (trackStartSamples.getUnchecked(i) + 88200) / 44100; offset > 0; offset /= 10)
+            checksum += offset % 10;
+
+    const int length = (trackStartSamples.getLast() - trackStartSamples.getFirst()) / 44100;
+
+    // CCLLLLTT: checksum, length, tracks
+    return ((checksum & 0xff) << 24) | (length << 8) | numTracks;
+}
+
+#endif

source/modules/juce_audio_devices/audio_cd/juce_AudioCDReader.h

@@ -0,0 +1,174 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#ifndef JUCE_AUDIOCDREADER_H_INCLUDED
+#define JUCE_AUDIOCDREADER_H_INCLUDED
+
+#if JUCE_USE_CDREADER || DOXYGEN
+
+
+//==============================================================================
+/**
+    A type of AudioFormatReader that reads from an audio CD.
+
+    One of these can be used to read a CD as if it's one big audio stream. Use the
+    getPositionOfTrackStart() method to find where the individual tracks are
+    within the stream.
+
+    @see AudioFormatReader
+*/
+class JUCE_API  AudioCDReader  : public AudioFormatReader
+{
+public:
+    //==============================================================================
+    /** Returns a list of names of Audio CDs currently available for reading.
+
+        If there's a CD drive but no CD in it, this might return an empty list, or
+        possibly a device that can be opened but which has no tracks, depending
+        on the platform.
+
+        @see createReaderForCD
+    */
+    static StringArray getAvailableCDNames();
+
+    /** Tries to create an AudioFormatReader that can read from an Audio CD.
+
+        @param index    the index of one of the available CDs - use getAvailableCDNames()
+                        to find out how many there are.
+        @returns        a new AudioCDReader object, or nullptr if it couldn't be created. The
+                        caller will be responsible for deleting the object returned.
+    */
+    static AudioCDReader* createReaderForCD (const int index);
+
+    //==============================================================================
+    /** Destructor. */
+    ~AudioCDReader();
+
+    /** Implementation of the AudioFormatReader method. */
+    bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                      int64 startSampleInFile, int numSamples) override;
+
+    /** Checks whether the CD has been removed from the drive. */
+    bool isCDStillPresent() const;
+
+    /** Returns the total number of tracks (audio + data). */
+    int getNumTracks() const;
+
+    /** Finds the sample offset of the start of a track.
+        @param trackNum     the track number, where trackNum = 0 is the first track
+                            and trackNum = getNumTracks() means the end of the CD.
+    */
+    int getPositionOfTrackStart (int trackNum) const;
+
+    /** Returns true if a given track is an audio track.
+        @param trackNum     the track number, where 0 is the first track.
+    */
+    bool isTrackAudio (int trackNum) const;
+
+    /** Returns an array of sample offsets for the start of each track, followed by
+        the sample position of the end of the CD.
+    */
+    const Array<int>& getTrackOffsets() const;
+
+    /** Refreshes the object's table of contents.
+
+        If the disc has been ejected and a different one put in since this
+        object was created, this will cause it to update its idea of how many tracks
+        there are, etc.
+    */
+    void refreshTrackLengths();
+
+    /** Enables scanning for indexes within tracks.
+        @see getLastIndex
+    */
+    void enableIndexScanning (bool enabled);
+
+    /** Returns the index number found during the last read() call.
+
+        Index scanning is turned off by default - turn it on with enableIndexScanning().
+
+        Then when the read() method is called, if it comes across an index within that
+        block, the index number is stored and returned by this method.
+
+        Some devices might not support indexes, of course.
+
+        (If you don't know what CD indexes are, it's unlikely you'll ever need them).
+
+        @see enableIndexScanning
+    */
+    int getLastIndex() const;
+
+    /** Scans a track to find the position of any indexes within it.
+        @param trackNumber  the track to look in, where 0 is the first track on the disc
+        @returns    an array of sample positions of any index points found (not including
+                    the index that marks the start of the track)
+    */
+    Array<int> findIndexesInTrack (const int trackNumber);
+
+    /** Returns the CDDB id number for the CD.
+        It's not a great way of identifying a disc, but it's traditional.
+    */
+    int getCDDBId();
+
+    /** Tries to eject the disk.
+        Ejecting the disk might not actually be possible, e.g. if some other process is using it.
+    */
+    void ejectDisk();
+
+    //==============================================================================
+    enum
+    {
+        framesPerSecond = 75,
+        samplesPerFrame = 44100 / framesPerSecond
+    };
+
+private:
+    //==============================================================================
+    Array<int> trackStartSamples;
+
+   #if JUCE_MAC
+    File volumeDir;
+    Array<File> tracks;
+    int currentReaderTrack;
+    ScopedPointer <AudioFormatReader> reader;
+    AudioCDReader (const File& volume);
+
+   #elif JUCE_WINDOWS
+    bool audioTracks [100];
+    void* handle;
+    MemoryBlock buffer;
+    bool indexingEnabled;
+    int lastIndex, firstFrameInBuffer, samplesInBuffer;
+    AudioCDReader (void* handle);
+    int getIndexAt (int samplePos);
+
+   #elif JUCE_LINUX
+    AudioCDReader();
+   #endif
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AudioCDReader)
+};
+
+#endif
+#endif   // JUCE_AUDIOCDREADER_H_INCLUDED

source/modules/juce_audio_devices/audio_io/juce_AudioDeviceManager.cpp

@@ -0,0 +1,968 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+AudioDeviceManager::AudioDeviceSetup::AudioDeviceSetup()
+    : sampleRate (0),
+      bufferSize (0),
+      useDefaultInputChannels (true),
+      useDefaultOutputChannels (true)
+{
+}
+
+bool AudioDeviceManager::AudioDeviceSetup::operator== (const AudioDeviceManager::AudioDeviceSetup& other) const
+{
+    return outputDeviceName == other.outputDeviceName
+            && inputDeviceName == other.inputDeviceName
+            && sampleRate == other.sampleRate
+            && bufferSize == other.bufferSize
+            && inputChannels == other.inputChannels
+            && useDefaultInputChannels == other.useDefaultInputChannels
+            && outputChannels == other.outputChannels
+            && useDefaultOutputChannels == other.useDefaultOutputChannels;
+}
+
+//==============================================================================
+class AudioDeviceManager::CallbackHandler  : public AudioIODeviceCallback,
+                                             public MidiInputCallback,
+                                             public AudioIODeviceType::Listener
+{
+public:
+    CallbackHandler (AudioDeviceManager& adm) noexcept  : owner (adm) {}
+
+private:
+    void audioDeviceIOCallback (const float** ins, int numIns, float** outs, int numOuts, int numSamples) override
+    {
+        owner.audioDeviceIOCallbackInt (ins, numIns, outs, numOuts, numSamples);
+    }
+
+    void audioDeviceAboutToStart (AudioIODevice* device) override
+    {
+        owner.audioDeviceAboutToStartInt (device);
+    }
+
+    void audioDeviceStopped() override
+    {
+        owner.audioDeviceStoppedInt();
+    }
+
+    void audioDeviceError (const String& message) override
+    {
+        owner.audioDeviceErrorInt (message);
+    }
+
+    void handleIncomingMidiMessage (MidiInput* source, const MidiMessage& message) override
+    {
+        owner.handleIncomingMidiMessageInt (source, message);
+    }
+
+    void audioDeviceListChanged() override
+    {
+        owner.audioDeviceListChanged();
+    }
+
+    AudioDeviceManager& owner;
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (CallbackHandler)
+};
+
+
+//==============================================================================
+AudioDeviceManager::AudioDeviceManager()
+    : numInputChansNeeded (0),
+      numOutputChansNeeded (2),
+      listNeedsScanning (true),
+      useInputNames (false),
+      inputLevel (0),
+      testSoundPosition (0),
+      tempBuffer (2, 2),
+      cpuUsageMs (0),
+      timeToCpuScale (0)
+{
+    callbackHandler = new CallbackHandler (*this);
+}
+
+AudioDeviceManager::~AudioDeviceManager()
+{
+    currentAudioDevice = nullptr;
+    defaultMidiOutput = nullptr;
+}
+
+
+//==============================================================================
+void AudioDeviceManager::createDeviceTypesIfNeeded()
+{
+    if (availableDeviceTypes.size() == 0)
+    {
+        OwnedArray <AudioIODeviceType> types;
+        createAudioDeviceTypes (types);
+
+        for (int i = 0; i < types.size(); ++i)
+            addAudioDeviceType (types.getUnchecked(i));
+
+        types.clear (false);
+
+        if (AudioIODeviceType* first = availableDeviceTypes.getFirst())
+            currentDeviceType = first->getTypeName();
+    }
+}
+
+const OwnedArray <AudioIODeviceType>& AudioDeviceManager::getAvailableDeviceTypes()
+{
+    scanDevicesIfNeeded();
+    return availableDeviceTypes;
+}
+
+void AudioDeviceManager::audioDeviceListChanged()
+{
+    if (currentAudioDevice != nullptr)
+    {
+        currentSetup.sampleRate     = currentAudioDevice->getCurrentSampleRate();
+        currentSetup.bufferSize     = currentAudioDevice->getCurrentBufferSizeSamples();
+        currentSetup.inputChannels  = currentAudioDevice->getActiveInputChannels();
+        currentSetup.outputChannels = currentAudioDevice->getActiveOutputChannels();
+    }
+
+    sendChangeMessage();
+}
+
+//==============================================================================
+static void addIfNotNull (OwnedArray <AudioIODeviceType>& list, AudioIODeviceType* const device)
+{
+    if (device != nullptr)
+        list.add (device);
+}
+
+void AudioDeviceManager::createAudioDeviceTypes (OwnedArray <AudioIODeviceType>& list)
+{
+    addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_WASAPI());
+    addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_DirectSound());
+    addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_ASIO());
+    addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_CoreAudio());
+    addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_iOSAudio());
+    addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_JACK());
+    addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_ALSA());
+    addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_OpenSLES());
+    addIfNotNull (list, AudioIODeviceType::createAudioIODeviceType_Android());
+}
+
+void AudioDeviceManager::addAudioDeviceType (AudioIODeviceType* newDeviceType)
+{
+    if (newDeviceType != nullptr)
+    {
+        jassert (lastDeviceTypeConfigs.size() == availableDeviceTypes.size());
+        availableDeviceTypes.add (newDeviceType);
+        lastDeviceTypeConfigs.add (new AudioDeviceSetup());
+
+        newDeviceType->addListener (callbackHandler);
+    }
+}
+
+//==============================================================================
+String AudioDeviceManager::initialise (const int numInputChannelsNeeded,
+                                       const int numOutputChannelsNeeded,
+                                       const XmlElement* const e,
+                                       const bool selectDefaultDeviceOnFailure,
+                                       const String& preferredDefaultDeviceName,
+                                       const AudioDeviceSetup* preferredSetupOptions)
+{
+    scanDevicesIfNeeded();
+
+    numInputChansNeeded = numInputChannelsNeeded;
+    numOutputChansNeeded = numOutputChannelsNeeded;
+
+    if (e != nullptr && e->hasTagName ("DEVICESETUP"))
+    {
+        lastExplicitSettings = new XmlElement (*e);
+
+        String error;
+        AudioDeviceSetup setup;
+
+        if (preferredSetupOptions != nullptr)
+            setup = *preferredSetupOptions;
+
+        if (e->getStringAttribute ("audioDeviceName").isNotEmpty())
+        {
+            setup.inputDeviceName = setup.outputDeviceName
+                = e->getStringAttribute ("audioDeviceName");
+        }
+        else
+        {
+            setup.inputDeviceName = e->getStringAttribute ("audioInputDeviceName");
+            setup.outputDeviceName = e->getStringAttribute ("audioOutputDeviceName");
+        }
+
+        currentDeviceType = e->getStringAttribute ("deviceType");
+
+        if (findType (currentDeviceType) == nullptr)
+        {
+            if (AudioIODeviceType* const type = findType (setup.inputDeviceName, setup.outputDeviceName))
+                currentDeviceType = type->getTypeName();
+            else if (availableDeviceTypes.size() > 0)
+                currentDeviceType = availableDeviceTypes.getUnchecked(0)->getTypeName();
+        }
+
+        setup.bufferSize = e->getIntAttribute ("audioDeviceBufferSize");
+        setup.sampleRate = e->getDoubleAttribute ("audioDeviceRate");
+
+        setup.inputChannels .parseString (e->getStringAttribute ("audioDeviceInChans",  "11"), 2);
+        setup.outputChannels.parseString (e->getStringAttribute ("audioDeviceOutChans", "11"), 2);
+
+        setup.useDefaultInputChannels  = ! e->hasAttribute ("audioDeviceInChans");
+        setup.useDefaultOutputChannels = ! e->hasAttribute ("audioDeviceOutChans");
+
+        error = setAudioDeviceSetup (setup, true);
+
+        midiInsFromXml.clear();
+        forEachXmlChildElementWithTagName (*e, c, "MIDIINPUT")
+            midiInsFromXml.add (c->getStringAttribute ("name"));
+
+        const StringArray allMidiIns (MidiInput::getDevices());
+
+        for (int i = allMidiIns.size(); --i >= 0;)
+            setMidiInputEnabled (allMidiIns[i], midiInsFromXml.contains (allMidiIns[i]));
+
+        if (error.isNotEmpty() && selectDefaultDeviceOnFailure)
+            error = initialise (numInputChannelsNeeded, numOutputChannelsNeeded, 0,
+                                false, preferredDefaultDeviceName);
+
+        setDefaultMidiOutput (e->getStringAttribute ("defaultMidiOutput"));
+
+        return error;
+    }
+    else
+    {
+        AudioDeviceSetup setup;
+
+        if (preferredSetupOptions != nullptr)
+        {
+            setup = *preferredSetupOptions;
+        }
+        else if (preferredDefaultDeviceName.isNotEmpty())
+        {
+            for (int j = availableDeviceTypes.size(); --j >= 0;)
+            {
+                AudioIODeviceType* const type = availableDeviceTypes.getUnchecked(j);
+
+                const StringArray outs (type->getDeviceNames (false));
+
+                for (int i = 0; i < outs.size(); ++i)
+                {
+                    if (outs[i].matchesWildcard (preferredDefaultDeviceName, true))
+                    {
+                        setup.outputDeviceName = outs[i];
+                        break;
+                    }
+                }
+
+                const StringArray ins (type->getDeviceNames (true));
+
+                for (int i = 0; i < ins.size(); ++i)
+                {
+                    if (ins[i].matchesWildcard (preferredDefaultDeviceName, true))
+                    {
+                        setup.inputDeviceName = ins[i];
+                        break;
+                    }
+                }
+            }
+        }
+
+        insertDefaultDeviceNames (setup);
+        return setAudioDeviceSetup (setup, false);
+    }
+}
+
+void AudioDeviceManager::insertDefaultDeviceNames (AudioDeviceSetup& setup) const
+{
+    if (AudioIODeviceType* type = getCurrentDeviceTypeObject())
+    {
+        if (setup.outputDeviceName.isEmpty())
+            setup.outputDeviceName = type->getDeviceNames (false) [type->getDefaultDeviceIndex (false)];
+
+        if (setup.inputDeviceName.isEmpty())
+            setup.inputDeviceName = type->getDeviceNames (true) [type->getDefaultDeviceIndex (true)];
+    }
+}
+
+XmlElement* AudioDeviceManager::createStateXml() const
+{
+    return lastExplicitSettings.createCopy();
+}
+
+//==============================================================================
+void AudioDeviceManager::scanDevicesIfNeeded()
+{
+    if (listNeedsScanning)
+    {
+        listNeedsScanning = false;
+
+        createDeviceTypesIfNeeded();
+
+        for (int i = availableDeviceTypes.size(); --i >= 0;)
+            availableDeviceTypes.getUnchecked(i)->scanForDevices();
+    }
+}
+
+AudioIODeviceType* AudioDeviceManager::findType (const String& typeName)
+{
+    scanDevicesIfNeeded();
+
+    for (int i = availableDeviceTypes.size(); --i >= 0;)
+        if (availableDeviceTypes.getUnchecked(i)->getTypeName() == typeName)
+            return availableDeviceTypes.getUnchecked(i);
+
+    return nullptr;
+}
+
+AudioIODeviceType* AudioDeviceManager::findType (const String& inputName, const String& outputName)
+{
+    scanDevicesIfNeeded();
+
+    for (int i = availableDeviceTypes.size(); --i >= 0;)
+    {
+        AudioIODeviceType* const type = availableDeviceTypes.getUnchecked(i);
+
+        if ((inputName.isNotEmpty() && type->getDeviceNames (true).contains (inputName, true))
+            || (outputName.isNotEmpty() && type->getDeviceNames (false).contains (outputName, true)))
+        {
+            return type;
+        }
+    }
+
+    return nullptr;
+}
+
+void AudioDeviceManager::getAudioDeviceSetup (AudioDeviceSetup& setup)
+{
+    setup = currentSetup;
+}
+
+void AudioDeviceManager::deleteCurrentDevice()
+{
+    currentAudioDevice = nullptr;
+    currentSetup.inputDeviceName = String::empty;
+    currentSetup.outputDeviceName = String::empty;
+}
+
+void AudioDeviceManager::setCurrentAudioDeviceType (const String& type,
+                                                    const bool treatAsChosenDevice)
+{
+    for (int i = 0; i < availableDeviceTypes.size(); ++i)
+    {
+        if (availableDeviceTypes.getUnchecked(i)->getTypeName() == type
+             && currentDeviceType != type)
+        {
+            if (currentAudioDevice != nullptr)
+            {
+                closeAudioDevice();
+                Thread::sleep (1500); // allow a moment for OS devices to sort themselves out, to help
+                                      // avoid things like DirectSound/ASIO clashes
+            }
+
+            currentDeviceType = type;
+
+            AudioDeviceSetup s (*lastDeviceTypeConfigs.getUnchecked(i));
+            insertDefaultDeviceNames (s);
+
+            setAudioDeviceSetup (s, treatAsChosenDevice);
+
+            sendChangeMessage();
+            break;
+        }
+    }
+}
+
+AudioIODeviceType* AudioDeviceManager::getCurrentDeviceTypeObject() const
+{
+    for (int i = 0; i < availableDeviceTypes.size(); ++i)
+        if (availableDeviceTypes.getUnchecked(i)->getTypeName() == currentDeviceType)
+            return availableDeviceTypes.getUnchecked(i);
+
+    return availableDeviceTypes[0];
+}
+
+String AudioDeviceManager::setAudioDeviceSetup (const AudioDeviceSetup& newSetup,
+                                                const bool treatAsChosenDevice)
+{
+    jassert (&newSetup != &currentSetup);    // this will have no effect
+
+    if (newSetup == currentSetup && currentAudioDevice != nullptr)
+        return String::empty;
+
+    if (! (newSetup == currentSetup))
+        sendChangeMessage();
+
+    stopDevice();
+
+    const String newInputDeviceName (numInputChansNeeded == 0 ? String::empty : newSetup.inputDeviceName);
+    const String newOutputDeviceName (numOutputChansNeeded == 0 ? String::empty : newSetup.outputDeviceName);
+
+    String error;
+    AudioIODeviceType* type = getCurrentDeviceTypeObject();
+
+    if (type == nullptr || (newInputDeviceName.isEmpty() && newOutputDeviceName.isEmpty()))
+    {
+        deleteCurrentDevice();
+
+        if (treatAsChosenDevice)
+            updateXml();
+
+        return String::empty;
+    }
+
+    if (currentSetup.inputDeviceName != newInputDeviceName
+         || currentSetup.outputDeviceName != newOutputDeviceName
+         || currentAudioDevice == nullptr)
+    {
+        deleteCurrentDevice();
+        scanDevicesIfNeeded();
+
+        if (newOutputDeviceName.isNotEmpty()
+             && ! type->getDeviceNames (false).contains (newOutputDeviceName))
+        {
+            return "No such device: " + newOutputDeviceName;
+        }
+
+        if (newInputDeviceName.isNotEmpty()
+             && ! type->getDeviceNames (true).contains (newInputDeviceName))
+        {
+            return "No such device: " + newInputDeviceName;
+        }
+
+        currentAudioDevice = type->createDevice (newOutputDeviceName, newInputDeviceName);
+
+        if (currentAudioDevice == nullptr)
+            error = "Can't open the audio device!\n\n"
+                    "This may be because another application is currently using the same device - "
+                    "if so, you should close any other applications and try again!";
+        else
+            error = currentAudioDevice->getLastError();
+
+        if (error.isNotEmpty())
+        {
+            deleteCurrentDevice();
+            return error;
+        }
+
+        if (newSetup.useDefaultInputChannels)
+        {
+            inputChannels.clear();
+            inputChannels.setRange (0, numInputChansNeeded, true);
+        }
+
+        if (newSetup.useDefaultOutputChannels)
+        {
+            outputChannels.clear();
+            outputChannels.setRange (0, numOutputChansNeeded, true);
+        }
+
+        if (newInputDeviceName.isEmpty())  inputChannels.clear();
+        if (newOutputDeviceName.isEmpty()) outputChannels.clear();
+    }
+
+    if (! newSetup.useDefaultInputChannels)    inputChannels  = newSetup.inputChannels;
+    if (! newSetup.useDefaultOutputChannels)   outputChannels = newSetup.outputChannels;
+
+    currentSetup = newSetup;
+
+    currentSetup.sampleRate = chooseBestSampleRate (newSetup.sampleRate);
+    currentSetup.bufferSize = chooseBestBufferSize (newSetup.bufferSize);
+
+    error = currentAudioDevice->open (inputChannels,
+                                      outputChannels,
+                                      currentSetup.sampleRate,
+                                      currentSetup.bufferSize);
+
+    if (error.isEmpty())
+    {
+        currentDeviceType = currentAudioDevice->getTypeName();
+
+        currentAudioDevice->start (callbackHandler);
+
+        currentSetup.sampleRate     = currentAudioDevice->getCurrentSampleRate();
+        currentSetup.bufferSize     = currentAudioDevice->getCurrentBufferSizeSamples();
+        currentSetup.inputChannels  = currentAudioDevice->getActiveInputChannels();
+        currentSetup.outputChannels = currentAudioDevice->getActiveOutputChannels();
+
+        for (int i = 0; i < availableDeviceTypes.size(); ++i)
+            if (availableDeviceTypes.getUnchecked (i)->getTypeName() == currentDeviceType)
+                *(lastDeviceTypeConfigs.getUnchecked (i)) = currentSetup;
+
+        if (treatAsChosenDevice)
+            updateXml();
+    }
+    else
+    {
+        deleteCurrentDevice();
+    }
+
+    return error;
+}
+
+double AudioDeviceManager::chooseBestSampleRate (double rate) const
+{
+    jassert (currentAudioDevice != nullptr);
+
+    if (rate > 0)
+        for (int i = currentAudioDevice->getNumSampleRates(); --i >= 0;)
+            if (currentAudioDevice->getSampleRate (i) == rate)
+                return rate;
+
+    double lowestAbove44 = 0.0;
+
+    for (int i = currentAudioDevice->getNumSampleRates(); --i >= 0;)
+    {
+        const double sr = currentAudioDevice->getSampleRate (i);
+
+        if (sr >= 44100.0 && (lowestAbove44 < 1.0 || sr < lowestAbove44))
+            lowestAbove44 = sr;
+    }
+
+    if (lowestAbove44 > 0.0)
+        return lowestAbove44;
+
+    return currentAudioDevice->getSampleRate (0);
+}
+
+int AudioDeviceManager::chooseBestBufferSize (int bufferSize) const
+{
+    jassert (currentAudioDevice != nullptr);
+
+    if (bufferSize > 0)
+        for (int i = currentAudioDevice->getNumBufferSizesAvailable(); --i >= 0;)
+            if (currentAudioDevice->getBufferSizeSamples(i) == bufferSize)
+                return bufferSize;
+
+    return currentAudioDevice->getDefaultBufferSize();
+}
+
+void AudioDeviceManager::stopDevice()
+{
+    if (currentAudioDevice != nullptr)
+        currentAudioDevice->stop();
+
+    testSound = nullptr;
+}
+
+void AudioDeviceManager::closeAudioDevice()
+{
+    stopDevice();
+    currentAudioDevice = nullptr;
+}
+
+void AudioDeviceManager::restartLastAudioDevice()
+{
+    if (currentAudioDevice == nullptr)
+    {
+        if (currentSetup.inputDeviceName.isEmpty()
+              && currentSetup.outputDeviceName.isEmpty())
+        {
+            // This method will only reload the last device that was running
+            // before closeAudioDevice() was called - you need to actually open
+            // one first, with setAudioDevice().
+            jassertfalse;
+            return;
+        }
+
+        AudioDeviceSetup s (currentSetup);
+        setAudioDeviceSetup (s, false);
+    }
+}
+
+void AudioDeviceManager::updateXml()
+{
+    lastExplicitSettings = new XmlElement ("DEVICESETUP");
+
+    lastExplicitSettings->setAttribute ("deviceType", currentDeviceType);
+    lastExplicitSettings->setAttribute ("audioOutputDeviceName", currentSetup.outputDeviceName);
+    lastExplicitSettings->setAttribute ("audioInputDeviceName", currentSetup.inputDeviceName);
+
+    if (currentAudioDevice != nullptr)
+    {
+        lastExplicitSettings->setAttribute ("audioDeviceRate", currentAudioDevice->getCurrentSampleRate());
+
+        if (currentAudioDevice->getDefaultBufferSize() != currentAudioDevice->getCurrentBufferSizeSamples())
+            lastExplicitSettings->setAttribute ("audioDeviceBufferSize", currentAudioDevice->getCurrentBufferSizeSamples());
+
+        if (! currentSetup.useDefaultInputChannels)
+            lastExplicitSettings->setAttribute ("audioDeviceInChans", currentSetup.inputChannels.toString (2));
+
+        if (! currentSetup.useDefaultOutputChannels)
+            lastExplicitSettings->setAttribute ("audioDeviceOutChans", currentSetup.outputChannels.toString (2));
+    }
+
+    for (int i = 0; i < enabledMidiInputs.size(); ++i)
+        lastExplicitSettings->createNewChildElement ("MIDIINPUT")
+                            ->setAttribute ("name", enabledMidiInputs[i]->getName());
+
+    if (midiInsFromXml.size() > 0)
+    {
+        // Add any midi devices that have been enabled before, but which aren't currently
+        // open because the device has been disconnected.
+        const StringArray availableMidiDevices (MidiInput::getDevices());
+
+        for (int i = 0; i < midiInsFromXml.size(); ++i)
+            if (! availableMidiDevices.contains (midiInsFromXml[i], true))
+                lastExplicitSettings->createNewChildElement ("MIDIINPUT")
+                                    ->setAttribute ("name", midiInsFromXml[i]);
+    }
+
+    if (defaultMidiOutputName.isNotEmpty())
+        lastExplicitSettings->setAttribute ("defaultMidiOutput", defaultMidiOutputName);
+}
+
+//==============================================================================
+void AudioDeviceManager::addAudioCallback (AudioIODeviceCallback* newCallback)
+{
+    {
+        const ScopedLock sl (audioCallbackLock);
+        if (callbacks.contains (newCallback))
+            return;
+    }
+
+    if (currentAudioDevice != nullptr && newCallback != nullptr)
+        newCallback->audioDeviceAboutToStart (currentAudioDevice);
+
+    const ScopedLock sl (audioCallbackLock);
+    callbacks.add (newCallback);
+}
+
+void AudioDeviceManager::removeAudioCallback (AudioIODeviceCallback* callbackToRemove)
+{
+    if (callbackToRemove != nullptr)
+    {
+        bool needsDeinitialising = currentAudioDevice != nullptr;
+
+        {
+            const ScopedLock sl (audioCallbackLock);
+
+            needsDeinitialising = needsDeinitialising && callbacks.contains (callbackToRemove);
+            callbacks.removeFirstMatchingValue (callbackToRemove);
+        }
+
+        if (needsDeinitialising)
+            callbackToRemove->audioDeviceStopped();
+    }
+}
+
+void AudioDeviceManager::audioDeviceIOCallbackInt (const float** inputChannelData,
+                                                   int numInputChannels,
+                                                   float** outputChannelData,
+                                                   int numOutputChannels,
+                                                   int numSamples)
+{
+    const ScopedLock sl (audioCallbackLock);
+
+    if (inputLevelMeasurementEnabledCount.get() > 0 && numInputChannels > 0)
+    {
+        for (int j = 0; j < numSamples; ++j)
+        {
+            float s = 0;
+
+            for (int i = 0; i < numInputChannels; ++i)
+                s += std::abs (inputChannelData[i][j]);
+
+            s /= numInputChannels;
+
+            const double decayFactor = 0.99992;
+
+            if (s > inputLevel)
+                inputLevel = s;
+            else if (inputLevel > 0.001f)
+                inputLevel *= decayFactor;
+            else
+                inputLevel = 0;
+        }
+    }
+    else
+    {
+        inputLevel = 0;
+    }
+
+    if (callbacks.size() > 0)
+    {
+        const double callbackStartTime = Time::getMillisecondCounterHiRes();
+
+        tempBuffer.setSize (jmax (1, numOutputChannels), jmax (1, numSamples), false, false, true);
+
+        callbacks.getUnchecked(0)->audioDeviceIOCallback (inputChannelData, numInputChannels,
+                                                          outputChannelData, numOutputChannels, numSamples);
+
+        float** const tempChans = tempBuffer.getArrayOfChannels();
+
+        for (int i = callbacks.size(); --i > 0;)
+        {
+            callbacks.getUnchecked(i)->audioDeviceIOCallback (inputChannelData, numInputChannels,
+                                                              tempChans, numOutputChannels, numSamples);
+
+            for (int chan = 0; chan < numOutputChannels; ++chan)
+            {
+                if (const float* const src = tempChans [chan])
+                    if (float* const dst = outputChannelData [chan])
+                        for (int j = 0; j < numSamples; ++j)
+                            dst[j] += src[j];
+            }
+        }
+
+        const double msTaken = Time::getMillisecondCounterHiRes() - callbackStartTime;
+        const double filterAmount = 0.2;
+        cpuUsageMs += filterAmount * (msTaken - cpuUsageMs);
+    }
+    else
+    {
+        for (int i = 0; i < numOutputChannels; ++i)
+            zeromem (outputChannelData[i], sizeof (float) * (size_t) numSamples);
+    }
+
+    if (testSound != nullptr)
+    {
+        const int numSamps = jmin (numSamples, testSound->getNumSamples() - testSoundPosition);
+        const float* const src = testSound->getSampleData (0, testSoundPosition);
+
+        for (int i = 0; i < numOutputChannels; ++i)
+            for (int j = 0; j < numSamps; ++j)
+                outputChannelData [i][j] += src[j];
+
+        testSoundPosition += numSamps;
+        if (testSoundPosition >= testSound->getNumSamples())
+            testSound = nullptr;
+    }
+}
+
+void AudioDeviceManager::audioDeviceAboutToStartInt (AudioIODevice* const device)
+{
+    cpuUsageMs = 0;
+
+    const double sampleRate = device->getCurrentSampleRate();
+    const int blockSize = device->getCurrentBufferSizeSamples();
+
+    if (sampleRate > 0.0 && blockSize > 0)
+    {
+        const double msPerBlock = 1000.0 * blockSize / sampleRate;
+        timeToCpuScale = (msPerBlock > 0.0) ? (1.0 / msPerBlock) : 0.0;
+    }
+
+    {
+        const ScopedLock sl (audioCallbackLock);
+        for (int i = callbacks.size(); --i >= 0;)
+            callbacks.getUnchecked(i)->audioDeviceAboutToStart (device);
+    }
+
+    sendChangeMessage();
+}
+
+void AudioDeviceManager::audioDeviceStoppedInt()
+{
+    cpuUsageMs = 0;
+    timeToCpuScale = 0;
+    sendChangeMessage();
+
+    const ScopedLock sl (audioCallbackLock);
+    for (int i = callbacks.size(); --i >= 0;)
+        callbacks.getUnchecked(i)->audioDeviceStopped();
+}
+
+void AudioDeviceManager::audioDeviceErrorInt (const String& message)
+{
+    const ScopedLock sl (audioCallbackLock);
+    for (int i = callbacks.size(); --i >= 0;)
+        callbacks.getUnchecked(i)->audioDeviceError (message);
+}
+
+double AudioDeviceManager::getCpuUsage() const
+{
+    return jlimit (0.0, 1.0, timeToCpuScale * cpuUsageMs);
+}
+
+//==============================================================================
+void AudioDeviceManager::setMidiInputEnabled (const String& name, const bool enabled)
+{
+    if (enabled != isMidiInputEnabled (name))
+    {
+        if (enabled)
+        {
+            const int index = MidiInput::getDevices().indexOf (name);
+
+            if (index >= 0)
+            {
+                if (MidiInput* const midiIn = MidiInput::openDevice (index, callbackHandler))
+                {
+                    enabledMidiInputs.add (midiIn);
+                    midiIn->start();
+                }
+            }
+        }
+        else
+        {
+            for (int i = enabledMidiInputs.size(); --i >= 0;)
+                if (enabledMidiInputs[i]->getName() == name)
+                    enabledMidiInputs.remove (i);
+        }
+
+        updateXml();
+        sendChangeMessage();
+    }
+}
+
+bool AudioDeviceManager::isMidiInputEnabled (const String& name) const
+{
+    for (int i = enabledMidiInputs.size(); --i >= 0;)
+        if (enabledMidiInputs[i]->getName() == name)
+            return true;
+
+    return false;
+}
+
+void AudioDeviceManager::addMidiInputCallback (const String& name, MidiInputCallback* callbackToAdd)
+{
+    removeMidiInputCallback (name, callbackToAdd);
+
+    if (name.isEmpty() || isMidiInputEnabled (name))
+    {
+        const ScopedLock sl (midiCallbackLock);
+        midiCallbacks.add (callbackToAdd);
+        midiCallbackDevices.add (name);
+    }
+}
+
+void AudioDeviceManager::removeMidiInputCallback (const String& name, MidiInputCallback* callbackToRemove)
+{
+    for (int i = midiCallbacks.size(); --i >= 0;)
+    {
+        if (midiCallbackDevices[i] == name && midiCallbacks.getUnchecked(i) == callbackToRemove)
+        {
+            const ScopedLock sl (midiCallbackLock);
+            midiCallbacks.remove (i);
+            midiCallbackDevices.remove (i);
+        }
+    }
+}
+
+void AudioDeviceManager::handleIncomingMidiMessageInt (MidiInput* source, const MidiMessage& message)
+{
+    if (! message.isActiveSense())
+    {
+        const bool isDefaultSource = (source == nullptr || source == enabledMidiInputs.getFirst());
+
+        const ScopedLock sl (midiCallbackLock);
+
+        for (int i = midiCallbackDevices.size(); --i >= 0;)
+        {
+            const String name (midiCallbackDevices[i]);
+
+            if ((isDefaultSource && name.isEmpty()) || (name.isNotEmpty() && name == source->getName()))
+                midiCallbacks.getUnchecked(i)->handleIncomingMidiMessage (source, message);
+        }
+    }
+}
+
+//==============================================================================
+void AudioDeviceManager::setDefaultMidiOutput (const String& deviceName)
+{
+    if (defaultMidiOutputName != deviceName)
+    {
+        Array <AudioIODeviceCallback*> oldCallbacks;
+
+        {
+            const ScopedLock sl (audioCallbackLock);
+            oldCallbacks.swapWith (callbacks);
+        }
+
+        if (currentAudioDevice != nullptr)
+            for (int i = oldCallbacks.size(); --i >= 0;)
+                oldCallbacks.getUnchecked(i)->audioDeviceStopped();
+
+        defaultMidiOutput = nullptr;
+        defaultMidiOutputName = deviceName;
+
+        if (deviceName.isNotEmpty())
+            defaultMidiOutput = MidiOutput::openDevice (MidiOutput::getDevices().indexOf (deviceName));
+
+        if (currentAudioDevice != nullptr)
+            for (int i = oldCallbacks.size(); --i >= 0;)
+                oldCallbacks.getUnchecked(i)->audioDeviceAboutToStart (currentAudioDevice);
+
+        {
+            const ScopedLock sl (audioCallbackLock);
+            oldCallbacks.swapWith (callbacks);
+        }
+
+        updateXml();
+        sendChangeMessage();
+    }
+}
+
+//==============================================================================
+void AudioDeviceManager::playTestSound()
+{
+    { // cunningly nested to swap, unlock and delete in that order.
+        ScopedPointer <AudioSampleBuffer> oldSound;
+
+        {
+            const ScopedLock sl (audioCallbackLock);
+            oldSound = testSound;
+        }
+    }
+
+    testSoundPosition = 0;
+
+    if (currentAudioDevice != nullptr)
+    {
+        const double sampleRate = currentAudioDevice->getCurrentSampleRate();
+        const int soundLength = (int) sampleRate;
+
+        AudioSampleBuffer* const newSound = new AudioSampleBuffer (1, soundLength);
+        float* samples = newSound->getSampleData (0);
+
+        const double frequency = 440.0;
+        const float amplitude = 0.5f;
+
+        const double phasePerSample = double_Pi * 2.0 / (sampleRate / frequency);
+
+        for (int i = 0; i < soundLength; ++i)
+            samples[i] = amplitude * (float) std::sin (i * phasePerSample);
+
+        newSound->applyGainRamp (0, 0, soundLength / 10, 0.0f, 1.0f);
+        newSound->applyGainRamp (0, soundLength - soundLength / 4, soundLength / 4, 1.0f, 0.0f);
+
+        const ScopedLock sl (audioCallbackLock);
+        testSound = newSound;
+    }
+}
+
+void AudioDeviceManager::enableInputLevelMeasurement (const bool enableMeasurement)
+{
+    if (enableMeasurement)
+        ++inputLevelMeasurementEnabledCount;
+    else
+        --inputLevelMeasurementEnabledCount;
+
+    inputLevel = 0;
+}
+
+double AudioDeviceManager::getCurrentInputLevel() const
+{
+    jassert (inputLevelMeasurementEnabledCount.get() > 0); // you need to call enableInputLevelMeasurement() before using this!
+    return inputLevel;
+}

source/modules/juce_audio_devices/audio_io/juce_AudioDeviceManager.h

@@ -0,0 +1,510 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#ifndef JUCE_AUDIODEVICEMANAGER_H_INCLUDED
+#define JUCE_AUDIODEVICEMANAGER_H_INCLUDED
+
+
+//==============================================================================
+/**
+    Manages the state of some audio and midi i/o devices.
+
+    This class keeps tracks of a currently-selected audio device, through
+    with which it continuously streams data from an audio callback, as well as
+    one or more midi inputs.
+
+    The idea is that your application will create one global instance of this object,
+    and let it take care of creating and deleting specific types of audio devices
+    internally. So when the device is changed, your callbacks will just keep running
+    without having to worry about this.
+
+    The manager can save and reload all of its device settings as XML, which
+    makes it very easy for you to save and reload the audio setup of your
+    application.
+
+    And to make it easy to let the user change its settings, there's a component
+    to do just that - the AudioDeviceSelectorComponent class, which contains a set of
+    device selection/sample-rate/latency controls.
+
+    To use an AudioDeviceManager, create one, and use initialise() to set it up. Then
+    call addAudioCallback() to register your audio callback with it, and use that to process
+    your audio data.
+
+    The manager also acts as a handy hub for incoming midi messages, allowing a
+    listener to register for messages from either a specific midi device, or from whatever
+    the current default midi input device is. The listener then doesn't have to worry about
+    re-registering with different midi devices if they are changed or deleted.
+
+    And yet another neat trick is that amount of CPU time being used is measured and
+    available with the getCpuUsage() method.
+
+    The AudioDeviceManager is a ChangeBroadcaster, and will send a change message to
+    listeners whenever one of its settings is changed.
+
+    @see AudioDeviceSelectorComponent, AudioIODevice, AudioIODeviceType
+*/
+class JUCE_API  AudioDeviceManager  : public ChangeBroadcaster
+{
+public:
+    //==============================================================================
+    /** Creates a default AudioDeviceManager.
+
+        Initially no audio device will be selected. You should call the initialise() method
+        and register an audio callback with setAudioCallback() before it'll be able to
+        actually make any noise.
+    */
+    AudioDeviceManager();
+
+    /** Destructor. */
+    ~AudioDeviceManager();
+
+    //==============================================================================
+    /**
+        This structure holds a set of properties describing the current audio setup.
+
+        An AudioDeviceManager uses this class to save/load its current settings, and to
+        specify your preferred options when opening a device.
+
+        @see AudioDeviceManager::setAudioDeviceSetup(), AudioDeviceManager::initialise()
+    */
+    struct JUCE_API  AudioDeviceSetup
+    {
+        /** Creates an AudioDeviceSetup object.
+
+            The default constructor sets all the member variables to indicate default values.
+            You can then fill-in any values you want to before passing the object to
+            AudioDeviceManager::initialise().
+        */
+        AudioDeviceSetup();
+
+        bool operator== (const AudioDeviceSetup& other) const;
+
+        /** The name of the audio device used for output.
+            The name has to be one of the ones listed by the AudioDeviceManager's currently
+            selected device type.
+            This may be the same as the input device.
+            An empty string indicates the default device.
+        */
+        String outputDeviceName;
+
+        /** The name of the audio device used for input.
+            This may be the same as the output device.
+            An empty string indicates the default device.
+        */
+        String inputDeviceName;
+
+        /** The current sample rate.
+            This rate is used for both the input and output devices.
+            A value of 0 indicates that you don't care what rate is used, and the
+            device will choose a sensible rate for you.
+        */
+        double sampleRate;
+
+        /** The buffer size, in samples.
+            This buffer size is used for both the input and output devices.
+            A value of 0 indicates the default buffer size.
+        */
+        int bufferSize;
+
+        /** The set of active input channels.
+            The bits that are set in this array indicate the channels of the
+            input device that are active.
+            If useDefaultInputChannels is true, this value is ignored.
+        */
+        BigInteger inputChannels;
+
+        /** If this is true, it indicates that the inputChannels array
+            should be ignored, and instead, the device's default channels
+            should be used.
+        */
+        bool useDefaultInputChannels;
+
+        /** The set of active output channels.
+            The bits that are set in this array indicate the channels of the
+            input device that are active.
+            If useDefaultOutputChannels is true, this value is ignored.
+        */
+        BigInteger outputChannels;
+
+        /** If this is true, it indicates that the outputChannels array
+            should be ignored, and instead, the device's default channels
+            should be used.
+        */
+        bool useDefaultOutputChannels;
+    };
+
+
+    //==============================================================================
+    /** Opens a set of audio devices ready for use.
+
+        This will attempt to open either a default audio device, or one that was
+        previously saved as XML.
+
+        @param numInputChannelsNeeded       the maximum number of input channels your app would like to
+                                            use (the actual number of channels opened may be less than
+                                            the number requested)
+        @param numOutputChannelsNeeded      the maximum number of output channels your app would like to
+                                            use (the actual number of channels opened may be less than
+                                            the number requested)
+        @param savedState                   either a previously-saved state that was produced
+                                            by createStateXml(), or nullptr if you want the manager
+                                            to choose the best device to open.
+        @param selectDefaultDeviceOnFailure if true, then if the device specified in the XML
+                                            fails to open, then a default device will be used
+                                            instead. If false, then on failure, no device is
+                                            opened.
+        @param preferredDefaultDeviceName   if this is not empty, and there's a device with this
+                                            name, then that will be used as the default device
+                                            (assuming that there wasn't one specified in the XML).
+                                            The string can actually be a simple wildcard, containing "*"
+                                            and "?" characters
+        @param preferredSetupOptions        if this is non-null, the structure will be used as the
+                                            set of preferred settings when opening the device. If you
+                                            use this parameter, the preferredDefaultDeviceName
+                                            field will be ignored
+
+        @returns an error message if anything went wrong, or an empty string if it worked ok.
+    */
+    String initialise (int numInputChannelsNeeded,
+                       int numOutputChannelsNeeded,
+                       const XmlElement* savedState,
+                       bool selectDefaultDeviceOnFailure,
+                       const String& preferredDefaultDeviceName = String::empty,
+                       const AudioDeviceSetup* preferredSetupOptions = 0);
+
+    /** Returns some XML representing the current state of the manager.
+
+        This stores the current device, its samplerate, block size, etc, and
+        can be restored later with initialise().
+
+        Note that this can return a null pointer if no settings have been explicitly changed
+        (i.e. if the device manager has just been left in its default state).
+    */
+    XmlElement* createStateXml() const;
+
+    //==============================================================================
+    /** Returns the current device properties that are in use.
+
+        @see setAudioDeviceSetup
+    */
+    void getAudioDeviceSetup (AudioDeviceSetup& setup);
+
+    /** Changes the current device or its settings.
+
+        If you want to change a device property, like the current sample rate or
+        block size, you can call getAudioDeviceSetup() to retrieve the current
+        settings, then tweak the appropriate fields in the AudioDeviceSetup structure,
+        and pass it back into this method to apply the new settings.
+
+        @param newSetup             the settings that you'd like to use
+        @param treatAsChosenDevice  if this is true and if the device opens correctly, these new
+                                    settings will be taken as having been explicitly chosen by the
+                                    user, and the next time createStateXml() is called, these settings
+                                    will be returned. If it's false, then the device is treated as a
+                                    temporary or default device, and a call to createStateXml() will
+                                    return either the last settings that were made with treatAsChosenDevice
+                                    as true, or the last XML settings that were passed into initialise().
+        @returns an error message if anything went wrong, or an empty string if it worked ok.
+
+        @see getAudioDeviceSetup
+    */
+    String setAudioDeviceSetup (const AudioDeviceSetup& newSetup,
+                                bool treatAsChosenDevice);
+
+
+    /** Returns the currently-active audio device. */
+    AudioIODevice* getCurrentAudioDevice() const noexcept               { return currentAudioDevice; }
+
+    /** Returns the type of audio device currently in use.
+        @see setCurrentAudioDeviceType
+    */
+    String getCurrentAudioDeviceType() const                            { return currentDeviceType; }
+
+    /** Returns the currently active audio device type object.
+        Don't keep a copy of this pointer - it's owned by the device manager and could
+        change at any time.
+    */
+    AudioIODeviceType* getCurrentDeviceTypeObject() const;
+
+    /** Changes the class of audio device being used.
+
+        This switches between, e.g. ASIO and DirectSound. On the Mac you probably won't ever call
+        this because there's only one type: CoreAudio.
+
+        For a list of types, see getAvailableDeviceTypes().
+    */
+    void setCurrentAudioDeviceType (const String& type,
+                                    bool treatAsChosenDevice);
+
+
+    /** Closes the currently-open device.
+
+        You can call restartLastAudioDevice() later to reopen it in the same state
+        that it was just in.
+    */
+    void closeAudioDevice();
+
+    /** Tries to reload the last audio device that was running.
+
+        Note that this only reloads the last device that was running before
+        closeAudioDevice() was called - it doesn't reload any kind of saved-state,
+        and can only be called after a device has been opened with SetAudioDevice().
+
+        If a device is already open, this call will do nothing.
+    */
+    void restartLastAudioDevice();
+
+    //==============================================================================
+    /** Registers an audio callback to be used.
+
+        The manager will redirect callbacks from whatever audio device is currently
+        in use to all registered callback objects. If more than one callback is
+        active, they will all be given the same input data, and their outputs will
+        be summed.
+
+        If necessary, this method will invoke audioDeviceAboutToStart() on the callback
+        object before returning.
+
+        To remove a callback, use removeAudioCallback().
+    */
+    void addAudioCallback (AudioIODeviceCallback* newCallback);
+
+    /** Deregisters a previously added callback.
+
+        If necessary, this method will invoke audioDeviceStopped() on the callback
+        object before returning.
+
+        @see addAudioCallback
+    */
+    void removeAudioCallback (AudioIODeviceCallback* callback);
+
+    //==============================================================================
+    /** Returns the average proportion of available CPU being spent inside the audio callbacks.
+
+        Returns a value between 0 and 1.0
+    */
+    double getCpuUsage() const;
+
+    //==============================================================================
+    /** Enables or disables a midi input device.
+
+        The list of devices can be obtained with the MidiInput::getDevices() method.
+
+        Any incoming messages from enabled input devices will be forwarded on to all the
+        listeners that have been registered with the addMidiInputCallback() method. They
+        can either register for messages from a particular device, or from just the
+        "default" midi input.
+
+        Routing the midi input via an AudioDeviceManager means that when a listener
+        registers for the default midi input, this default device can be changed by the
+        manager without the listeners having to know about it or re-register.
+
+        It also means that a listener can stay registered for a midi input that is disabled
+        or not present, so that when the input is re-enabled, the listener will start
+        receiving messages again.
+
+        @see addMidiInputCallback, isMidiInputEnabled
+    */
+    void setMidiInputEnabled (const String& midiInputDeviceName, bool enabled);
+
+    /** Returns true if a given midi input device is being used.
+
+        @see setMidiInputEnabled
+    */
+    bool isMidiInputEnabled (const String& midiInputDeviceName) const;
+
+    /** Registers a listener for callbacks when midi events arrive from a midi input.
+
+        The device name can be empty to indicate that it wants events from whatever the
+        current "default" device is. Or it can be the name of one of the midi input devices
+        (see MidiInput::getDevices() for the names).
+
+        Only devices which are enabled (see the setMidiInputEnabled() method) will have their
+        events forwarded on to listeners.
+    */
+    void addMidiInputCallback (const String& midiInputDeviceName,
+                               MidiInputCallback* callback);
+
+    /** Removes a listener that was previously registered with addMidiInputCallback().
+    */
+    void removeMidiInputCallback (const String& midiInputDeviceName,
+                                  MidiInputCallback* callback);
+
+    //==============================================================================
+    /** Sets a midi output device to use as the default.
+
+        The list of devices can be obtained with the MidiOutput::getDevices() method.
+
+        The specified device will be opened automatically and can be retrieved with the
+        getDefaultMidiOutput() method.
+
+        Pass in an empty string to deselect all devices. For the default device, you
+        can use MidiOutput::getDevices() [MidiOutput::getDefaultDeviceIndex()].
+
+        @see getDefaultMidiOutput, getDefaultMidiOutputName
+    */
+    void setDefaultMidiOutput (const String& deviceName);
+
+    /** Returns the name of the default midi output.
+
+        @see setDefaultMidiOutput, getDefaultMidiOutput
+    */
+    String getDefaultMidiOutputName() const                         { return defaultMidiOutputName; }
+
+    /** Returns the current default midi output device.
+
+        If no device has been selected, or the device can't be opened, this will
+        return 0.
+
+        @see getDefaultMidiOutputName
+    */
+    MidiOutput* getDefaultMidiOutput() const noexcept               { return defaultMidiOutput; }
+
+    /** Returns a list of the types of device supported.
+    */
+    const OwnedArray <AudioIODeviceType>& getAvailableDeviceTypes();
+
+    //==============================================================================
+    /** Creates a list of available types.
+
+        This will add a set of new AudioIODeviceType objects to the specified list, to
+        represent each available types of device.
+
+        You can override this if your app needs to do something specific, like avoid
+        using DirectSound devices, etc.
+    */
+    virtual void createAudioDeviceTypes (OwnedArray <AudioIODeviceType>& types);
+
+    /** Adds a new device type to the list of types.
+        The manager will take ownership of the object that is passed-in.
+    */
+    void addAudioDeviceType (AudioIODeviceType* newDeviceType);
+
+    //==============================================================================
+    /** Plays a beep through the current audio device.
+
+        This is here to allow the audio setup UI panels to easily include a "test"
+        button so that the user can check where the audio is coming from.
+    */
+    void playTestSound();
+
+    /** Turns on level-measuring.
+
+        When enabled, the device manager will measure the peak input level
+        across all channels, and you can get this level by calling getCurrentInputLevel().
+
+        This is mainly intended for audio setup UI panels to use to create a mic
+        level display, so that the user can check that they've selected the right
+        device.
+
+        A simple filter is used to make the level decay smoothly, but this is
+        only intended for giving rough feedback, and not for any kind of accurate
+        measurement.
+    */
+    void enableInputLevelMeasurement (bool enableMeasurement);
+
+    /** Returns the current input level.
+
+        To use this, you must first enable it by calling enableInputLevelMeasurement().
+
+        See enableInputLevelMeasurement() for more info.
+    */
+    double getCurrentInputLevel() const;
+
+    /** Returns the a lock that can be used to synchronise access to the audio callback.
+        Obviously while this is locked, you're blocking the audio thread from running, so
+        it must only be used for very brief periods when absolutely necessary.
+    */
+    CriticalSection& getAudioCallbackLock() noexcept        { return audioCallbackLock; }
+
+    /** Returns the a lock that can be used to synchronise access to the midi callback.
+        Obviously while this is locked, you're blocking the midi system from running, so
+        it must only be used for very brief periods when absolutely necessary.
+    */
+    CriticalSection& getMidiCallbackLock() noexcept         { return midiCallbackLock; }
+
+private:
+    //==============================================================================
+    OwnedArray <AudioIODeviceType> availableDeviceTypes;
+    OwnedArray <AudioDeviceSetup> lastDeviceTypeConfigs;
+
+    AudioDeviceSetup currentSetup;
+    ScopedPointer <AudioIODevice> currentAudioDevice;
+    Array <AudioIODeviceCallback*> callbacks;
+    int numInputChansNeeded, numOutputChansNeeded;
+    String currentDeviceType;
+    BigInteger inputChannels, outputChannels;
+    ScopedPointer <XmlElement> lastExplicitSettings;
+    mutable bool listNeedsScanning;
+    bool useInputNames;
+    Atomic<int> inputLevelMeasurementEnabledCount;
+    double inputLevel;
+    ScopedPointer <AudioSampleBuffer> testSound;
+    int testSoundPosition;
+    AudioSampleBuffer tempBuffer;
+
+    StringArray midiInsFromXml;
+    OwnedArray <MidiInput> enabledMidiInputs;
+    Array <MidiInputCallback*> midiCallbacks;
+    StringArray midiCallbackDevices;
+    String defaultMidiOutputName;
+    ScopedPointer <MidiOutput> defaultMidiOutput;
+    CriticalSection audioCallbackLock, midiCallbackLock;
+
+    double cpuUsageMs, timeToCpuScale;
+
+    //==============================================================================
+    class CallbackHandler;
+    friend class CallbackHandler;
+    friend struct ContainerDeletePolicy<CallbackHandler>;
+    ScopedPointer<CallbackHandler> callbackHandler;
+
+    void audioDeviceIOCallbackInt (const float** inputChannelData, int totalNumInputChannels,
+                                   float** outputChannelData, int totalNumOutputChannels, int numSamples);
+    void audioDeviceAboutToStartInt (AudioIODevice*);
+    void audioDeviceStoppedInt();
+    void audioDeviceErrorInt (const String&);
+    void handleIncomingMidiMessageInt (MidiInput*, const MidiMessage&);
+    void audioDeviceListChanged();
+
+    String restartDevice (int blockSizeToUse, double sampleRateToUse,
+                          const BigInteger& ins, const BigInteger& outs);
+    void stopDevice();
+
+    void updateXml();
+
+    void createDeviceTypesIfNeeded();
+    void scanDevicesIfNeeded();
+    void deleteCurrentDevice();
+    double chooseBestSampleRate (double preferred) const;
+    int chooseBestBufferSize (int preferred) const;
+    void insertDefaultDeviceNames (AudioDeviceSetup&) const;
+
+    AudioIODeviceType* findType (const String& inputName, const String& outputName);
+    AudioIODeviceType* findType (const String& typeName);
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AudioDeviceManager)
+};
+
+#endif   // JUCE_AUDIODEVICEMANAGER_H_INCLUDED

source/modules/juce_audio_devices/audio_io/juce_AudioIODevice.cpp

@@ -0,0 +1,48 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+AudioIODevice::AudioIODevice (const String& deviceName, const String& typeName_)
+    : name (deviceName),
+      typeName (typeName_)
+{
+}
+
+AudioIODevice::~AudioIODevice()
+{
+}
+
+bool AudioIODevice::hasControlPanel() const
+{
+    return false;
+}
+
+bool AudioIODevice::showControlPanel()
+{
+    jassertfalse;    // this should only be called for devices which return true from
+                     // their hasControlPanel() method.
+    return false;
+}
+
+//==============================================================================
+void AudioIODeviceCallback::audioDeviceError (const String&) {}

source/modules/juce_audio_devices/audio_io/juce_AudioIODevice.h

@@ -0,0 +1,332 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#ifndef JUCE_AUDIOIODEVICE_H_INCLUDED
+#define JUCE_AUDIOIODEVICE_H_INCLUDED
+
+class AudioIODevice;
+
+
+//==============================================================================
+/**
+    One of these is passed to an AudioIODevice object to stream the audio data
+    in and out.
+
+    The AudioIODevice will repeatedly call this class's audioDeviceIOCallback()
+    method on its own high-priority audio thread, when it needs to send or receive
+    the next block of data.
+
+    @see AudioIODevice, AudioDeviceManager
+*/
+class JUCE_API  AudioIODeviceCallback
+{
+public:
+    /** Destructor. */
+    virtual ~AudioIODeviceCallback()  {}
+
+    /** Processes a block of incoming and outgoing audio data.
+
+        The subclass's implementation should use the incoming audio for whatever
+        purposes it needs to, and must fill all the output channels with the next
+        block of output data before returning.
+
+        The channel data is arranged with the same array indices as the channel name
+        array returned by AudioIODevice::getOutputChannelNames(), but those channels
+        that aren't specified in AudioIODevice::open() will have a null pointer for their
+        associated channel, so remember to check for this.
+
+        @param inputChannelData     a set of arrays containing the audio data for each
+                                    incoming channel - this data is valid until the function
+                                    returns. There will be one channel of data for each input
+                                    channel that was enabled when the audio device was opened
+                                    (see AudioIODevice::open())
+        @param numInputChannels     the number of pointers to channel data in the
+                                    inputChannelData array.
+        @param outputChannelData    a set of arrays which need to be filled with the data
+                                    that should be sent to each outgoing channel of the device.
+                                    There will be one channel of data for each output channel
+                                    that was enabled when the audio device was opened (see
+                                    AudioIODevice::open())
+                                    The initial contents of the array is undefined, so the
+                                    callback function must fill all the channels with zeros if
+                                    its output is silence. Failing to do this could cause quite
+                                    an unpleasant noise!
+        @param numOutputChannels    the number of pointers to channel data in the
+                                    outputChannelData array.
+        @param numSamples           the number of samples in each channel of the input and
+                                    output arrays. The number of samples will depend on the
+                                    audio device's buffer size and will usually remain constant,
+                                    although this isn't guaranteed, so make sure your code can
+                                    cope with reasonable changes in the buffer size from one
+                                    callback to the next.
+    */
+    virtual void audioDeviceIOCallback (const float** inputChannelData,
+                                        int numInputChannels,
+                                        float** outputChannelData,
+                                        int numOutputChannels,
+                                        int numSamples) = 0;
+
+    /** Called to indicate that the device is about to start calling back.
+
+        This will be called just before the audio callbacks begin, either when this
+        callback has just been added to an audio device, or after the device has been
+        restarted because of a sample-rate or block-size change.
+
+        You can use this opportunity to find out the sample rate and block size
+        that the device is going to use by calling the AudioIODevice::getCurrentSampleRate()
+        and AudioIODevice::getCurrentBufferSizeSamples() on the supplied pointer.
+
+        @param device       the audio IO device that will be used to drive the callback.
+                            Note that if you're going to store this this pointer, it is
+                            only valid until the next time that audioDeviceStopped is called.
+    */
+    virtual void audioDeviceAboutToStart (AudioIODevice* device) = 0;
+
+    /** Called to indicate that the device has stopped. */
+    virtual void audioDeviceStopped() = 0;
+
+    /** This can be overridden to be told if the device generates an error while operating.
+        Be aware that this could be called by any thread! And not all devices perform
+        this callback.
+    */
+    virtual void audioDeviceError (const String& errorMessage);
+};
+
+
+//==============================================================================
+/**
+    Base class for an audio device with synchronised input and output channels.
+
+    Subclasses of this are used to implement different protocols such as DirectSound,
+    ASIO, CoreAudio, etc.
+
+    To create one of these, you'll need to use the AudioIODeviceType class - see the
+    documentation for that class for more info.
+
+    For an easier way of managing audio devices and their settings, have a look at the
+    AudioDeviceManager class.
+
+    @see AudioIODeviceType, AudioDeviceManager
+*/
+class JUCE_API  AudioIODevice
+{
+public:
+    /** Destructor. */
+    virtual ~AudioIODevice();
+
+    //==============================================================================
+    /** Returns the device's name, (as set in the constructor). */
+    const String& getName() const noexcept                          { return name; }
+
+    /** Returns the type of the device.
+
+        E.g. "CoreAudio", "ASIO", etc. - this comes from the AudioIODeviceType that created it.
+    */
+    const String& getTypeName() const noexcept                      { return typeName; }
+
+    //==============================================================================
+    /** Returns the names of all the available output channels on this device.
+        To find out which of these are currently in use, call getActiveOutputChannels().
+    */
+    virtual StringArray getOutputChannelNames() = 0;
+
+    /** Returns the names of all the available input channels on this device.
+        To find out which of these are currently in use, call getActiveInputChannels().
+    */
+    virtual StringArray getInputChannelNames() = 0;
+
+    //==============================================================================
+    /** Returns the number of sample-rates this device supports.
+
+        To find out which rates are available on this device, use this method to
+        find out how many there are, and getSampleRate() to get the rates.
+
+        @see getSampleRate
+    */
+    virtual int getNumSampleRates() = 0;
+
+    /** Returns one of the sample-rates this device supports.
+
+        To find out which rates are available on this device, use getNumSampleRates() to
+        find out how many there are, and getSampleRate() to get the individual rates.
+
+        The sample rate is set by the open() method.
+
+        (Note that for DirectSound some rates might not work, depending on combinations
+        of i/o channels that are being opened).
+
+        @see getNumSampleRates
+    */
+    virtual double getSampleRate (int index) = 0;
+
+    /** Returns the number of sizes of buffer that are available.
+
+        @see getBufferSizeSamples, getDefaultBufferSize
+    */
+    virtual int getNumBufferSizesAvailable() = 0;
+
+    /** Returns one of the possible buffer-sizes.
+
+        @param index    the index of the buffer-size to use, from 0 to getNumBufferSizesAvailable() - 1
+        @returns a number of samples
+        @see getNumBufferSizesAvailable, getDefaultBufferSize
+    */
+    virtual int getBufferSizeSamples (int index) = 0;
+
+    /** Returns the default buffer-size to use.
+
+        @returns a number of samples
+        @see getNumBufferSizesAvailable, getBufferSizeSamples
+    */
+    virtual int getDefaultBufferSize() = 0;
+
+    //==============================================================================
+    /** Tries to open the device ready to play.
+
+        @param inputChannels        a BigInteger in which a set bit indicates that the corresponding
+                                    input channel should be enabled
+        @param outputChannels       a BigInteger in which a set bit indicates that the corresponding
+                                    output channel should be enabled
+        @param sampleRate           the sample rate to try to use - to find out which rates are
+                                    available, see getNumSampleRates() and getSampleRate()
+        @param bufferSizeSamples    the size of i/o buffer to use - to find out the available buffer
+                                    sizes, see getNumBufferSizesAvailable() and getBufferSizeSamples()
+        @returns    an error description if there's a problem, or an empty string if it succeeds in
+                    opening the device
+        @see close
+    */
+    virtual String open (const BigInteger& inputChannels,
+                         const BigInteger& outputChannels,
+                         double sampleRate,
+                         int bufferSizeSamples) = 0;
+
+    /** Closes and releases the device if it's open. */
+    virtual void close() = 0;
+
+    /** Returns true if the device is still open.
+
+        A device might spontaneously close itself if something goes wrong, so this checks if
+        it's still open.
+    */
+    virtual bool isOpen() = 0;
+
+    /** Starts the device actually playing.
+
+        This must be called after the device has been opened.
+
+        @param callback     the callback to use for streaming the data.
+        @see AudioIODeviceCallback, open
+    */
+    virtual void start (AudioIODeviceCallback* callback) = 0;
+
+    /** Stops the device playing.
+
+        Once a device has been started, this will stop it. Any pending calls to the
+        callback class will be flushed before this method returns.
+    */
+    virtual void stop() = 0;
+
+    /** Returns true if the device is still calling back.
+
+        The device might mysteriously stop, so this checks whether it's
+        still playing.
+    */
+    virtual bool isPlaying() = 0;
+
+    /** Returns the last error that happened if anything went wrong. */
+    virtual String getLastError() = 0;
+
+    //==============================================================================
+    /** Returns the buffer size that the device is currently using.
+
+        If the device isn't actually open, this value doesn't really mean much.
+    */
+    virtual int getCurrentBufferSizeSamples() = 0;
+
+    /** Returns the sample rate that the device is currently using.
+
+        If the device isn't actually open, this value doesn't really mean much.
+    */
+    virtual double getCurrentSampleRate() = 0;
+
+    /** Returns the device's current physical bit-depth.
+
+        If the device isn't actually open, this value doesn't really mean much.
+    */
+    virtual int getCurrentBitDepth() = 0;
+
+    /** Returns a mask showing which of the available output channels are currently
+        enabled.
+        @see getOutputChannelNames
+    */
+    virtual BigInteger getActiveOutputChannels() const = 0;
+
+    /** Returns a mask showing which of the available input channels are currently
+        enabled.
+        @see getInputChannelNames
+    */
+    virtual BigInteger getActiveInputChannels() const = 0;
+
+    /** Returns the device's output latency.
+
+        This is the delay in samples between a callback getting a block of data, and
+        that data actually getting played.
+    */
+    virtual int getOutputLatencyInSamples() = 0;
+
+    /** Returns the device's input latency.
+
+        This is the delay in samples between some audio actually arriving at the soundcard,
+        and the callback getting passed this block of data.
+    */
+    virtual int getInputLatencyInSamples() = 0;
+
+
+    //==============================================================================
+    /** True if this device can show a pop-up control panel for editing its settings.
+
+        This is generally just true of ASIO devices. If true, you can call showControlPanel()
+        to display it.
+    */
+    virtual bool hasControlPanel() const;
+
+    /** Shows a device-specific control panel if there is one.
+
+        This should only be called for devices which return true from hasControlPanel().
+    */
+    virtual bool showControlPanel();
+
+
+    //==============================================================================
+protected:
+    /** Creates a device, setting its name and type member variables. */
+    AudioIODevice (const String& deviceName,
+                   const String& typeName);
+
+    /** @internal */
+    String name, typeName;
+};
+
+
+#endif   // JUCE_AUDIOIODEVICE_H_INCLUDED

source/modules/juce_audio_devices/audio_io/juce_AudioIODeviceType.cpp

@@ -0,0 +1,78 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+AudioIODeviceType::AudioIODeviceType (const String& name)
+    : typeName (name)
+{
+}
+
+AudioIODeviceType::~AudioIODeviceType()
+{
+}
+
+//==============================================================================
+void AudioIODeviceType::addListener (Listener* l)      { listeners.add (l); }
+void AudioIODeviceType::removeListener (Listener* l)   { listeners.remove (l); }
+
+void AudioIODeviceType::callDeviceChangeListeners()
+{
+    listeners.call (&AudioIODeviceType::Listener::audioDeviceListChanged);
+}
+
+//==============================================================================
+#if ! JUCE_MAC
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_CoreAudio()       { return nullptr; }
+#endif
+
+#if ! JUCE_IOS
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_iOSAudio()        { return nullptr; }
+#endif
+
+#if ! (JUCE_WINDOWS && JUCE_WASAPI)
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_WASAPI()          { return nullptr; }
+#endif
+
+#if ! (JUCE_WINDOWS && JUCE_DIRECTSOUND)
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_DirectSound()     { return nullptr; }
+#endif
+
+#if ! (JUCE_WINDOWS && JUCE_ASIO)
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_ASIO()            { return nullptr; }
+#endif
+
+#if ! (JUCE_LINUX && JUCE_ALSA)
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_ALSA()            { return nullptr; }
+#endif
+
+#if ! (JUCE_LINUX && JUCE_JACK)
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_JACK()            { return nullptr; }
+#endif
+
+#if ! JUCE_ANDROID
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_Android()         { return nullptr; }
+#endif
+
+#if ! (JUCE_ANDROID && JUCE_USE_ANDROID_OPENSLES)
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_OpenSLES()        { return nullptr; }
+#endif

source/modules/juce_audio_devices/audio_io/juce_AudioIODeviceType.h

@@ -0,0 +1,182 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#ifndef JUCE_AUDIOIODEVICETYPE_H_INCLUDED
+#define JUCE_AUDIOIODEVICETYPE_H_INCLUDED
+
+
+//==============================================================================
+/**
+    Represents a type of audio driver, such as DirectSound, ASIO, CoreAudio, etc.
+
+    To get a list of available audio driver types, use the AudioDeviceManager::createAudioDeviceTypes()
+    method. Each of the objects returned can then be used to list the available
+    devices of that type. E.g.
+    @code
+    OwnedArray <AudioIODeviceType> types;
+    myAudioDeviceManager.createAudioDeviceTypes (types);
+
+    for (int i = 0; i < types.size(); ++i)
+    {
+        String typeName (types[i]->getTypeName());  // This will be things like "DirectSound", "CoreAudio", etc.
+
+        types[i]->scanForDevices();                 // This must be called before getting the list of devices
+
+        StringArray deviceNames (types[i]->getDeviceNames());  // This will now return a list of available devices of this type
+
+        for (int j = 0; j < deviceNames.size(); ++j)
+        {
+            AudioIODevice* device = types[i]->createDevice (deviceNames [j]);
+
+            ...
+        }
+    }
+    @endcode
+
+    For an easier way of managing audio devices and their settings, have a look at the
+    AudioDeviceManager class.
+
+    @see AudioIODevice, AudioDeviceManager
+*/
+class JUCE_API  AudioIODeviceType
+{
+public:
+    //==============================================================================
+    /** Returns the name of this type of driver that this object manages.
+
+        This will be something like "DirectSound", "ASIO", "CoreAudio", "ALSA", etc.
+    */
+    const String& getTypeName() const noexcept                      { return typeName; }
+
+    //==============================================================================
+    /** Refreshes the object's cached list of known devices.
+
+        This must be called at least once before calling getDeviceNames() or any of
+        the other device creation methods.
+    */
+    virtual void scanForDevices() = 0;
+
+    /** Returns the list of available devices of this type.
+
+        The scanForDevices() method must have been called to create this list.
+
+        @param wantInputNames     only really used by DirectSound where devices are split up
+                                  into inputs and outputs, this indicates whether to use
+                                  the input or output name to refer to a pair of devices.
+    */
+    virtual StringArray getDeviceNames (bool wantInputNames = false) const = 0;
+
+    /** Returns the name of the default device.
+
+        This will be one of the names from the getDeviceNames() list.
+
+        @param forInput     if true, this means that a default input device should be
+                            returned; if false, it should return the default output
+    */
+    virtual int getDefaultDeviceIndex (bool forInput) const = 0;
+
+    /** Returns the index of a given device in the list of device names.
+        If asInput is true, it shows the index in the inputs list, otherwise it
+        looks for it in the outputs list.
+    */
+    virtual int getIndexOfDevice (AudioIODevice* device, bool asInput) const = 0;
+
+    /** Returns true if two different devices can be used for the input and output.
+    */
+    virtual bool hasSeparateInputsAndOutputs() const = 0;
+
+    /** Creates one of the devices of this type.
+
+        The deviceName must be one of the strings returned by getDeviceNames(), and
+        scanForDevices() must have been called before this method is used.
+    */
+    virtual AudioIODevice* createDevice (const String& outputDeviceName,
+                                         const String& inputDeviceName) = 0;
+
+    //==============================================================================
+    /**
+        A class for receiving events when audio devices are inserted or removed.
+
+        You can register a AudioIODeviceType::Listener with an~AudioIODeviceType object
+        using the AudioIODeviceType::addListener() method, and it will be called when
+        devices of that type are added or removed.
+
+        @see AudioIODeviceType::addListener, AudioIODeviceType::removeListener
+    */
+    class Listener
+    {
+    public:
+        virtual ~Listener() {}
+
+        /** Called when the list of available audio devices changes. */
+        virtual void audioDeviceListChanged() = 0;
+    };
+
+    /** Adds a listener that will be called when this type of device is added or
+        removed from the system.
+    */
+    void addListener (Listener* listener);
+
+    /** Removes a listener that was previously added with addListener(). */
+    void removeListener (Listener* listener);
+
+    //==============================================================================
+    /** Destructor. */
+    virtual ~AudioIODeviceType();
+
+    //==============================================================================
+    /** Creates a CoreAudio device type if it's available on this platform, or returns null. */
+    static AudioIODeviceType* createAudioIODeviceType_CoreAudio();
+    /** Creates an iOS device type if it's available on this platform, or returns null. */
+    static AudioIODeviceType* createAudioIODeviceType_iOSAudio();
+    /** Creates a WASAPI device type if it's available on this platform, or returns null. */
+    static AudioIODeviceType* createAudioIODeviceType_WASAPI();
+    /** Creates a DirectSound device type if it's available on this platform, or returns null. */
+    static AudioIODeviceType* createAudioIODeviceType_DirectSound();
+    /** Creates an ASIO device type if it's available on this platform, or returns null. */
+    static AudioIODeviceType* createAudioIODeviceType_ASIO();
+    /** Creates an ALSA device type if it's available on this platform, or returns null. */
+    static AudioIODeviceType* createAudioIODeviceType_ALSA();
+    /** Creates a JACK device type if it's available on this platform, or returns null. */
+    static AudioIODeviceType* createAudioIODeviceType_JACK();
+    /** Creates an Android device type if it's available on this platform, or returns null. */
+    static AudioIODeviceType* createAudioIODeviceType_Android();
+    /** Creates an Android OpenSLES device type if it's available on this platform, or returns null. */
+    static AudioIODeviceType* createAudioIODeviceType_OpenSLES();
+
+protected:
+    explicit AudioIODeviceType (const String& typeName);
+
+    /** Synchronously calls all the registered device list change listeners. */
+    void callDeviceChangeListeners();
+
+private:
+    String typeName;
+    ListenerList<Listener> listeners;
+
+    JUCE_DECLARE_NON_COPYABLE (AudioIODeviceType)
+};
+
+
+#endif   // JUCE_AUDIOIODEVICETYPE_H_INCLUDED

source/modules/juce_audio_devices/audio_io/juce_SystemAudioVolume.h

@@ -0,0 +1,61 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#ifndef JUCE_SYSTEMAUDIOVOLUME_H_INCLUDED
+#define JUCE_SYSTEMAUDIOVOLUME_H_INCLUDED
+
+
+//==============================================================================
+/**
+    Contains functions to control the system's master volume.
+*/
+class JUCE_API  SystemAudioVolume
+{
+public:
+    //==============================================================================
+    /** Returns the operating system's current volume level in the range 0 to 1.0 */
+    static float JUCE_CALLTYPE getGain();
+
+    /** Attempts to set the operating system's current volume level.
+        @param newGain  the level, between 0 and 1.0
+        @returns true if the operation succeeds
+    */
+    static bool JUCE_CALLTYPE setGain (float newGain);
+
+    /** Returns true if the system's audio output is currently muted. */
+    static bool JUCE_CALLTYPE isMuted();
+
+    /** Attempts to mute the operating system's audio output.
+        @param shouldBeMuted    true if you want it to be muted
+        @returns true if the operation succeeds
+    */
+    static bool JUCE_CALLTYPE setMuted (bool shouldBeMuted);
+
+private:
+    SystemAudioVolume(); // Don't instantiate this class, just call its static fns.
+    JUCE_DECLARE_NON_COPYABLE (SystemAudioVolume)
+};
+
+
+#endif   // JUCE_SYSTEMAUDIOVOLUME_H_INCLUDED

source/modules/juce_audio_devices/juce_audio_devices.cpp

@@ -0,0 +1,226 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#if defined (JUCE_AUDIO_DEVICES_H_INCLUDED) && ! JUCE_AMALGAMATED_INCLUDE
+ /* When you add this cpp file to your project, you mustn't include it in a file where you've
+    already included any other headers - just put it inside a file on its own, possibly with your config
+    flags preceding it, but don't include anything else. That also includes avoiding any automatic prefix
+    header files that the compiler may be using.
+ */
+ #error "Incorrect use of JUCE cpp file"
+#endif
+
+// Your project must contain an AppConfig.h file with your project-specific settings in it,
+// and your header search path must make it accessible to the module's files.
+#include "AppConfig.h"
+
+#include "../juce_core/native/juce_BasicNativeHeaders.h"
+#include "juce_audio_devices.h"
+
+//==============================================================================
+#if JUCE_MAC
+ #define Point CarbonDummyPointName
+ #define Component CarbonDummyCompName
+ #import <CoreAudio/AudioHardware.h>
+ #import <CoreMIDI/MIDIServices.h>
+ #import <DiscRecording/DiscRecording.h>
+ #import <AudioToolbox/AudioServices.h>
+ #undef Point
+ #undef Component
+
+#elif JUCE_IOS
+ #import <AudioToolbox/AudioToolbox.h>
+ #import <AVFoundation/AVFoundation.h>
+ #import <CoreMIDI/MIDIServices.h>
+
+//==============================================================================
+#elif JUCE_WINDOWS
+ #if JUCE_WASAPI
+  #include <MMReg.h>
+ #endif
+
+ #if JUCE_ASIO
+  /* This is very frustrating - we only need to use a handful of definitions from
+     a couple of the header files in Steinberg's ASIO SDK, and it'd be easy to copy
+     about 30 lines of code into this cpp file to create a fully stand-alone ASIO
+     implementation...
+
+     ..unfortunately that would break Steinberg's license agreement for use of
+     their SDK, so I'm not allowed to do this.
+
+     This means that anyone who wants to use JUCE's ASIO abilities will have to:
+
+     1) Agree to Steinberg's licensing terms and download the ASIO SDK
+         (see www.steinberg.net/Steinberg/Developers.asp).
+
+     2) Enable this code with a global definition #define JUCE_ASIO 1.
+
+     3) Make sure that your header search path contains the iasiodrv.h file that
+        comes with the SDK. (Only about a handful of the SDK header files are actually
+        needed - so to simplify things, you could just copy these into your JUCE directory).
+  */
+  #include <iasiodrv.h>
+ #endif
+
+ #if JUCE_USE_CDBURNER
+  /* You'll need the Platform SDK for these headers - if you don't have it and don't
+     need to use CD-burning, then you might just want to set the JUCE_USE_CDBURNER flag
+     to 0, to avoid these includes.
+  */
+  #include <imapi.h>
+  #include <imapierror.h>
+ #endif
+
+//==============================================================================
+#elif JUCE_LINUX
+ #if JUCE_ALSA
+  /* Got an include error here? If so, you've either not got ALSA installed, or you've
+     not got your paths set up correctly to find its header files.
+
+     The package you need to install to get ASLA support is "libasound2-dev".
+
+     If you don't have the ALSA library and don't want to build Juce with audio support,
+     just set the JUCE_ALSA flag to 0.
+  */
+  #include <alsa/asoundlib.h>
+ #endif
+
+ #if JUCE_JACK
+  /* Got an include error here? If so, you've either not got jack-audio-connection-kit
+     installed, or you've not got your paths set up correctly to find its header files.
+
+     The package you need to install to get JACK support is "libjack-dev".
+
+     If you don't have the jack-audio-connection-kit library and don't want to build
+     Juce with low latency audio support, just set the JUCE_JACK flag to 0.
+  */
+  #include <jack/jack.h>
+ #endif
+ #undef SIZEOF
+
+//==============================================================================
+#elif JUCE_ANDROID
+
+ #if JUCE_USE_ANDROID_OPENSLES
+  #include <SLES/OpenSLES.h>
+  #include <SLES/OpenSLES_Android.h>
+ #endif
+
+#endif
+
+namespace juce
+{
+
+#include "audio_io/juce_AudioDeviceManager.cpp"
+#include "audio_io/juce_AudioIODevice.cpp"
+#include "audio_io/juce_AudioIODeviceType.cpp"
+#include "midi_io/juce_MidiMessageCollector.cpp"
+#include "midi_io/juce_MidiOutput.cpp"
+#include "audio_cd/juce_AudioCDReader.cpp"
+#include "sources/juce_AudioSourcePlayer.cpp"
+#include "sources/juce_AudioTransportSource.cpp"
+#include "native/juce_MidiDataConcatenator.h"
+
+//==============================================================================
+#if JUCE_MAC
+ #include "../juce_core/native/juce_osx_ObjCHelpers.h"
+ #include "native/juce_mac_CoreAudio.cpp"
+ #include "native/juce_mac_CoreMidi.cpp"
+
+ #if JUCE_USE_CDREADER
+  #include "native/juce_mac_AudioCDReader.mm"
+ #endif
+
+ #if JUCE_USE_CDBURNER
+  #include "native/juce_mac_AudioCDBurner.mm"
+ #endif
+
+//==============================================================================
+#elif JUCE_IOS
+ #include "native/juce_ios_Audio.cpp"
+ #include "native/juce_mac_CoreMidi.cpp"
+
+//==============================================================================
+#elif JUCE_WINDOWS
+ #include "../juce_core/native/juce_win32_ComSmartPtr.h"
+ #include "../juce_events/native/juce_win32_HiddenMessageWindow.h"
+
+ #if JUCE_WASAPI
+  #include "native/juce_win32_WASAPI.cpp"
+ #endif
+
+ #if JUCE_DIRECTSOUND
+  #include "native/juce_win32_DirectSound.cpp"
+ #endif
+
+ #include "native/juce_win32_Midi.cpp"
+
+ #if JUCE_ASIO
+  #include "native/juce_win32_ASIO.cpp"
+ #endif
+
+ #if JUCE_USE_CDREADER
+  #include "native/juce_win32_AudioCDReader.cpp"
+ #endif
+
+ #if JUCE_USE_CDBURNER
+  #include "native/juce_win32_AudioCDBurner.cpp"
+ #endif
+
+//==============================================================================
+#elif JUCE_LINUX
+ #if JUCE_ALSA
+  #include "native/juce_linux_ALSA.cpp"
+ #endif
+
+ #include "native/juce_linux_Midi.cpp"
+
+ #if JUCE_JACK
+  #include "native/juce_linux_JackAudio.cpp"
+ #endif
+
+ #if JUCE_USE_CDREADER
+  #include "native/juce_linux_AudioCDReader.cpp"
+ #endif
+
+//==============================================================================
+#elif JUCE_ANDROID
+ #include "../juce_core/native/juce_android_JNIHelpers.h"
+ #include "native/juce_android_Audio.cpp"
+ #include "native/juce_android_Midi.cpp"
+
+ #if JUCE_USE_ANDROID_OPENSLES
+  #include "native/juce_android_OpenSL.cpp"
+ #endif
+
+#endif
+
+#if ! JUCE_SYSTEMAUDIOVOL_IMPLEMENTED
+ // None of these methods are available. (On Windows you might need to enable WASAPI for this)
+ float JUCE_CALLTYPE SystemAudioVolume::getGain()         { jassertfalse; return 0.0f; }
+ bool  JUCE_CALLTYPE SystemAudioVolume::setGain (float)   { jassertfalse; return false; }
+ bool  JUCE_CALLTYPE SystemAudioVolume::isMuted()         { jassertfalse; return false; }
+ bool  JUCE_CALLTYPE SystemAudioVolume::setMuted (bool)   { jassertfalse; return false; }
+#endif
+}

source/modules/juce_audio_devices/juce_audio_devices.h

@@ -0,0 +1,117 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#ifndef JUCE_AUDIO_DEVICES_H_INCLUDED
+#define JUCE_AUDIO_DEVICES_H_INCLUDED
+
+#include "../juce_events/juce_events.h"
+#include "../juce_audio_basics/juce_audio_basics.h"
+#include "../juce_audio_formats/juce_audio_formats.h"
+
+//=============================================================================
+/** Config: JUCE_ASIO
+    Enables ASIO audio devices (MS Windows only).
+    Turning this on means that you'll need to have the Steinberg ASIO SDK installed
+    on your Windows build machine.
+
+    See the comments in the ASIOAudioIODevice class's header file for more
+    info about this.
+*/
+#ifndef JUCE_ASIO
+ #define JUCE_ASIO 0
+#endif
+
+/** Config: JUCE_WASAPI
+    Enables WASAPI audio devices (Windows Vista and above).
+*/
+#ifndef JUCE_WASAPI
+ #define JUCE_WASAPI 1
+#endif
+
+/** Config: JUCE_DIRECTSOUND
+    Enables DirectSound audio (MS Windows only).
+*/
+#ifndef JUCE_DIRECTSOUND
+ #define JUCE_DIRECTSOUND 1
+#endif
+
+/** Config: JUCE_ALSA
+    Enables ALSA audio devices (Linux only).
+*/
+#ifndef JUCE_ALSA
+ #define JUCE_ALSA 1
+#endif
+
+/** Config: JUCE_JACK
+    Enables JACK audio devices (Linux only).
+*/
+#ifndef JUCE_JACK
+ #define JUCE_JACK 0
+#endif
+
+/** Config: JUCE_USE_ANDROID_OPENSLES
+    Enables OpenSLES devices (Android only).
+*/
+#ifndef JUCE_USE_ANDROID_OPENSLES
+ #if JUCE_ANDROID_API_VERSION > 8
+  #define JUCE_USE_ANDROID_OPENSLES 1
+ #else
+  #define JUCE_USE_ANDROID_OPENSLES 0
+ #endif
+#endif
+
+//=============================================================================
+/** Config: JUCE_USE_CDREADER
+    Enables the AudioCDReader class (on supported platforms).
+*/
+#ifndef JUCE_USE_CDREADER
+ #define JUCE_USE_CDREADER 0
+#endif
+
+/** Config: JUCE_USE_CDBURNER
+    Enables the AudioCDBurner class (on supported platforms).
+*/
+#ifndef JUCE_USE_CDBURNER
+ #define JUCE_USE_CDBURNER 0
+#endif
+
+//=============================================================================
+namespace juce
+{
+
+#include "audio_io/juce_AudioIODevice.h"
+#include "audio_io/juce_AudioIODeviceType.h"
+#include "audio_io/juce_SystemAudioVolume.h"
+#include "midi_io/juce_MidiInput.h"
+#include "midi_io/juce_MidiMessageCollector.h"
+#include "midi_io/juce_MidiOutput.h"
+#include "sources/juce_AudioSourcePlayer.h"
+#include "sources/juce_AudioTransportSource.h"
+#include "audio_cd/juce_AudioCDBurner.h"
+#include "audio_cd/juce_AudioCDReader.h"
+#include "audio_io/juce_AudioDeviceManager.h"
+
+}
+
+#endif   // JUCE_AUDIO_DEVICES_H_INCLUDED

source/modules/juce_audio_devices/juce_audio_devices.mm

@@ -0,0 +1,25 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#include "juce_audio_devices.cpp"

source/modules/juce_audio_devices/juce_module_info

@@ -0,0 +1,28 @@
+{
+  "id":             "juce_audio_devices",
+  "name":           "JUCE audio and midi I/O device classes",
+  "version":        "2.1.3",
+  "description":    "Classes to play and record from audio and midi i/o devices.",
+  "website":        "http://www.juce.com/juce",
+  "license":        "GPL/Commercial",
+
+  "dependencies":   [ { "id": "juce_audio_basics",  "version": "matching" },
+                      { "id": "juce_audio_formats", "version": "matching" },
+                      { "id": "juce_events",        "version": "matching" } ],
+
+  "include":        "juce_audio_devices.h",
+
+  "compile":        [ { "file": "juce_audio_devices.cpp", "target": "! xcode" },
+                      { "file": "juce_audio_devices.mm",  "target": "xcode" } ],
+
+  "browse":         [ "audio_io/*",
+                      "midi_io/*",
+                      "sources/*",
+                      "audio_cd/*",
+                      "native/*" ],
+
+  "OSXFrameworks":  "CoreAudio CoreMIDI DiscRecording",
+  "iOSFrameworks":  "AudioToolbox CoreMIDI",
+  "LinuxLibs":      "asound",
+  "mingwLibs":      "winmm"
+}

source/modules/juce_audio_devices/midi_io/juce_MidiInput.h

@@ -0,0 +1,182 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#ifndef JUCE_MIDIINPUT_H_INCLUDED
+#define JUCE_MIDIINPUT_H_INCLUDED
+
+class MidiInput;
+
+
+//==============================================================================
+/**
+    Receives incoming messages from a physical MIDI input device.
+
+    This class is overridden to handle incoming midi messages. See the MidiInput
+    class for more details.
+
+    @see MidiInput
+*/
+class JUCE_API  MidiInputCallback
+{
+public:
+    /** Destructor. */
+    virtual ~MidiInputCallback()  {}
+
+
+    /** Receives an incoming message.
+
+        A MidiInput object will call this method when a midi event arrives. It'll be
+        called on a high-priority system thread, so avoid doing anything time-consuming
+        in here, and avoid making any UI calls. You might find the MidiBuffer class helpful
+        for queueing incoming messages for use later.
+
+        @param source   the MidiInput object that generated the message
+        @param message  the incoming message. The message's timestamp is set to a value
+                        equivalent to (Time::getMillisecondCounter() / 1000.0) to specify the
+                        time when the message arrived.
+    */
+    virtual void handleIncomingMidiMessage (MidiInput* source,
+                                            const MidiMessage& message) = 0;
+
+    /** Notification sent each time a packet of a multi-packet sysex message arrives.
+
+        If a long sysex message is broken up into multiple packets, this callback is made
+        for each packet that arrives until the message is finished, at which point
+        the normal handleIncomingMidiMessage() callback will be made with the entire
+        message.
+
+        The message passed in will contain the start of a sysex, but won't be finished
+        with the terminating 0xf7 byte.
+    */
+    virtual void handlePartialSysexMessage (MidiInput* source,
+                                            const uint8* messageData,
+                                            int numBytesSoFar,
+                                            double timestamp)
+    {
+        // (this bit is just to avoid compiler warnings about unused variables)
+        (void) source; (void) messageData; (void) numBytesSoFar; (void) timestamp;
+    }
+};
+
+//==============================================================================
+/**
+    Represents a midi input device.
+
+    To create one of these, use the static getDevices() method to find out what inputs are
+    available, and then use the openDevice() method to try to open one.
+
+    @see MidiOutput
+*/
+class JUCE_API  MidiInput
+{
+public:
+    //==============================================================================
+    /** Returns a list of the available midi input devices.
+
+        You can open one of the devices by passing its index into the
+        openDevice() method.
+
+        @see getDefaultDeviceIndex, openDevice
+    */
+    static StringArray getDevices();
+
+    /** Returns the index of the default midi input device to use.
+
+        This refers to the index in the list returned by getDevices().
+    */
+    static int getDefaultDeviceIndex();
+
+    /** Tries to open one of the midi input devices.
+
+        This will return a MidiInput object if it manages to open it. You can then
+        call start() and stop() on this device, and delete it when no longer needed.
+
+        If the device can't be opened, this will return a null pointer.
+
+        @param deviceIndex  the index of a device from the list returned by getDevices()
+        @param callback     the object that will receive the midi messages from this device.
+
+        @see MidiInputCallback, getDevices
+    */
+    static MidiInput* openDevice (int deviceIndex,
+                                  MidiInputCallback* callback);
+
+   #if JUCE_LINUX || JUCE_MAC || JUCE_IOS || DOXYGEN
+    /** This will try to create a new midi input device (Not available on Windows).
+
+        This will attempt to create a new midi input device with the specified name,
+        for other apps to connect to.
+
+        Returns nullptr if a device can't be created.
+
+        @param deviceName   the name to use for the new device
+        @param callback     the object that will receive the midi messages from this device.
+    */
+    static MidiInput* createNewDevice (const String& deviceName,
+                                       MidiInputCallback* callback);
+   #endif
+
+    //==============================================================================
+    /** Destructor. */
+    virtual ~MidiInput();
+
+    /** Returns the name of this device. */
+    const String& getName() const noexcept                      { return name; }
+
+    /** Allows you to set a custom name for the device, in case you don't like the name
+        it was given when created.
+    */
+    void setName (const String& newName) noexcept               { name = newName; }
+
+    //==============================================================================
+    /** Starts the device running.
+
+        After calling this, the device will start sending midi messages to the
+        MidiInputCallback object that was specified when the openDevice() method
+        was called.
+
+        @see stop
+    */
+    virtual void start();
+
+    /** Stops the device running.
+
+        @see start
+    */
+    virtual void stop();
+
+protected:
+    //==============================================================================
+    String name;
+    void* internal;
+
+    explicit MidiInput (const String& name);
+
+private:
+    //==============================================================================
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (MidiInput)
+};
+
+
+#endif   // JUCE_MIDIINPUT_H_INCLUDED

source/modules/juce_audio_devices/midi_io/juce_MidiMessageCollector.cpp

@@ -0,0 +1,153 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+MidiMessageCollector::MidiMessageCollector()
+    : lastCallbackTime (0),
+      sampleRate (44100.0001)
+{
+}
+
+MidiMessageCollector::~MidiMessageCollector()
+{
+}
+
+//==============================================================================
+void MidiMessageCollector::reset (const double sampleRate_)
+{
+    jassert (sampleRate_ > 0);
+
+    const ScopedLock sl (midiCallbackLock);
+    sampleRate = sampleRate_;
+    incomingMessages.clear();
+    lastCallbackTime = Time::getMillisecondCounterHiRes();
+}
+
+void MidiMessageCollector::addMessageToQueue (const MidiMessage& message)
+{
+    // you need to call reset() to set the correct sample rate before using this object
+    jassert (sampleRate != 44100.0001);
+
+    // the messages that come in here need to be time-stamped correctly - see MidiInput
+    // for details of what the number should be.
+    jassert (message.getTimeStamp() != 0);
+
+    const ScopedLock sl (midiCallbackLock);
+
+    const int sampleNumber
+        = (int) ((message.getTimeStamp() - 0.001 * lastCallbackTime) * sampleRate);
+
+    incomingMessages.addEvent (message, sampleNumber);
+
+    // if the messages don't get used for over a second, we'd better
+    // get rid of any old ones to avoid the queue getting too big
+    if (sampleNumber > sampleRate)
+        incomingMessages.clear (0, sampleNumber - (int) sampleRate);
+}
+
+void MidiMessageCollector::removeNextBlockOfMessages (MidiBuffer& destBuffer,
+                                                      const int numSamples)
+{
+    // you need to call reset() to set the correct sample rate before using this object
+    jassert (sampleRate != 44100.0001);
+    jassert (numSamples > 0);
+
+    const double timeNow = Time::getMillisecondCounterHiRes();
+    const double msElapsed = timeNow - lastCallbackTime;
+
+    const ScopedLock sl (midiCallbackLock);
+    lastCallbackTime = timeNow;
+
+    if (! incomingMessages.isEmpty())
+    {
+        int numSourceSamples = jmax (1, roundToInt (msElapsed * 0.001 * sampleRate));
+
+        int startSample = 0;
+        int scale = 1 << 16;
+
+        const uint8* midiData;
+        int numBytes, samplePosition;
+
+        MidiBuffer::Iterator iter (incomingMessages);
+
+        if (numSourceSamples > numSamples)
+        {
+            // if our list of events is longer than the buffer we're being
+            // asked for, scale them down to squeeze them all in..
+            const int maxBlockLengthToUse = numSamples << 5;
+
+            if (numSourceSamples > maxBlockLengthToUse)
+            {
+                startSample = numSourceSamples - maxBlockLengthToUse;
+                numSourceSamples = maxBlockLengthToUse;
+                iter.setNextSamplePosition (startSample);
+            }
+
+            scale = (numSamples << 10) / numSourceSamples;
+
+            while (iter.getNextEvent (midiData, numBytes, samplePosition))
+            {
+                samplePosition = ((samplePosition - startSample) * scale) >> 10;
+
+                destBuffer.addEvent (midiData, numBytes,
+                                     jlimit (0, numSamples - 1, samplePosition));
+            }
+        }
+        else
+        {
+            // if our event list is shorter than the number we need, put them
+            // towards the end of the buffer
+            startSample = numSamples - numSourceSamples;
+
+            while (iter.getNextEvent (midiData, numBytes, samplePosition))
+            {
+                destBuffer.addEvent (midiData, numBytes,
+                                     jlimit (0, numSamples - 1, samplePosition + startSample));
+            }
+        }
+
+        incomingMessages.clear();
+    }
+}
+
+//==============================================================================
+void MidiMessageCollector::handleNoteOn (MidiKeyboardState*, int midiChannel, int midiNoteNumber, float velocity)
+{
+    MidiMessage m (MidiMessage::noteOn (midiChannel, midiNoteNumber, velocity));
+    m.setTimeStamp (Time::getMillisecondCounterHiRes() * 0.001);
+
+    addMessageToQueue (m);
+}
+
+void MidiMessageCollector::handleNoteOff (MidiKeyboardState*, int midiChannel, int midiNoteNumber)
+{
+    MidiMessage m (MidiMessage::noteOff (midiChannel, midiNoteNumber));
+    m.setTimeStamp (Time::getMillisecondCounterHiRes() * 0.001);
+
+    addMessageToQueue (m);
+}
+
+void MidiMessageCollector::handleIncomingMidiMessage (MidiInput*, const MidiMessage& message)
+{
+    addMessageToQueue (message);
+}

source/modules/juce_audio_devices/midi_io/juce_MidiMessageCollector.h

@@ -0,0 +1,104 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#ifndef JUCE_MIDIMESSAGECOLLECTOR_H_INCLUDED
+#define JUCE_MIDIMESSAGECOLLECTOR_H_INCLUDED
+
+
+//==============================================================================
+/**
+    Collects incoming realtime MIDI messages and turns them into blocks suitable for
+    processing by a block-based audio callback.
+
+    The class can also be used as either a MidiKeyboardStateListener or a MidiInputCallback
+    so it can easily use a midi input or keyboard component as its source.
+
+    @see MidiMessage, MidiInput
+*/
+class JUCE_API  MidiMessageCollector    : public MidiKeyboardStateListener,
+                                          public MidiInputCallback
+{
+public:
+    //==============================================================================
+    /** Creates a MidiMessageCollector. */
+    MidiMessageCollector();
+
+    /** Destructor. */
+    ~MidiMessageCollector();
+
+    //==============================================================================
+    /** Clears any messages from the queue.
+
+        You need to call this method before starting to use the collector, so that
+        it knows the correct sample rate to use.
+    */
+    void reset (double sampleRate);
+
+    /** Takes an incoming real-time message and adds it to the queue.
+
+        The message's timestamp is taken, and it will be ready for retrieval as part
+        of the block returned by the next call to removeNextBlockOfMessages().
+
+        This method is fully thread-safe when overlapping calls are made with
+        removeNextBlockOfMessages().
+    */
+    void addMessageToQueue (const MidiMessage& message);
+
+    /** Removes all the pending messages from the queue as a buffer.
+
+        This will also correct the messages' timestamps to make sure they're in
+        the range 0 to numSamples - 1.
+
+        This call should be made regularly by something like an audio processing
+        callback, because the time that it happens is used in calculating the
+        midi event positions.
+
+        This method is fully thread-safe when overlapping calls are made with
+        addMessageToQueue().
+
+        Precondition: numSamples must be greater than 0.
+    */
+    void removeNextBlockOfMessages (MidiBuffer& destBuffer, int numSamples);
+
+
+    //==============================================================================
+    /** @internal */
+    void handleNoteOn (MidiKeyboardState*, int midiChannel, int midiNoteNumber, float velocity) override;
+    /** @internal */
+    void handleNoteOff (MidiKeyboardState*, int midiChannel, int midiNoteNumber) override;
+    /** @internal */
+    void handleIncomingMidiMessage (MidiInput*, const MidiMessage&) override;
+
+private:
+    //==============================================================================
+    double lastCallbackTime;
+    CriticalSection midiCallbackLock;
+    MidiBuffer incomingMessages;
+    double sampleRate;
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (MidiMessageCollector)
+};
+
+
+#endif   // JUCE_MIDIMESSAGECOLLECTOR_H_INCLUDED

source/modules/juce_audio_devices/midi_io/juce_MidiOutput.cpp

@@ -0,0 +1,162 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+struct MidiOutput::PendingMessage
+{
+    PendingMessage (const void* const data, const int len, const double timeStamp)
+        : message (data, len, timeStamp)
+    {}
+
+    MidiMessage message;
+    PendingMessage* next;
+};
+
+MidiOutput::MidiOutput()
+    : Thread ("midi out"),
+      internal (nullptr),
+      firstMessage (nullptr)
+{
+}
+
+void MidiOutput::sendBlockOfMessages (const MidiBuffer& buffer,
+                                      const double millisecondCounterToStartAt,
+                                      double samplesPerSecondForBuffer)
+{
+    // You've got to call startBackgroundThread() for this to actually work..
+    jassert (isThreadRunning());
+
+    // this needs to be a value in the future - RTFM for this method!
+    jassert (millisecondCounterToStartAt > 0);
+
+    const double timeScaleFactor = 1000.0 / samplesPerSecondForBuffer;
+
+    MidiBuffer::Iterator i (buffer);
+
+    const uint8* data;
+    int len, time;
+
+    while (i.getNextEvent (data, len, time))
+    {
+        const double eventTime = millisecondCounterToStartAt + timeScaleFactor * time;
+
+        PendingMessage* const m = new PendingMessage (data, len, eventTime);
+
+        const ScopedLock sl (lock);
+
+        if (firstMessage == nullptr || firstMessage->message.getTimeStamp() > eventTime)
+        {
+            m->next = firstMessage;
+            firstMessage = m;
+        }
+        else
+        {
+            PendingMessage* mm = firstMessage;
+
+            while (mm->next != nullptr && mm->next->message.getTimeStamp() <= eventTime)
+                mm = mm->next;
+
+            m->next = mm->next;
+            mm->next = m;
+        }
+    }
+
+    notify();
+}
+
+void MidiOutput::clearAllPendingMessages()
+{
+    const ScopedLock sl (lock);
+
+    while (firstMessage != nullptr)
+    {
+        PendingMessage* const m = firstMessage;
+        firstMessage = firstMessage->next;
+        delete m;
+    }
+}
+
+void MidiOutput::startBackgroundThread()
+{
+    startThread (9);
+}
+
+void MidiOutput::stopBackgroundThread()
+{
+    stopThread (5000);
+}
+
+void MidiOutput::run()
+{
+    while (! threadShouldExit())
+    {
+        uint32 now = Time::getMillisecondCounter();
+        uint32 eventTime = 0;
+        uint32 timeToWait = 500;
+
+        PendingMessage* message;
+
+        {
+            const ScopedLock sl (lock);
+            message = firstMessage;
+
+            if (message != nullptr)
+            {
+                eventTime = (uint32) roundToInt (message->message.getTimeStamp());
+
+                if (eventTime > now + 20)
+                {
+                    timeToWait = eventTime - (now + 20);
+                    message = nullptr;
+                }
+                else
+                {
+                    firstMessage = message->next;
+                }
+            }
+        }
+
+        if (message != nullptr)
+        {
+            const ScopedPointer<PendingMessage> messageDeleter (message);
+
+            if (eventTime > now)
+            {
+                Time::waitForMillisecondCounter (eventTime);
+
+                if (threadShouldExit())
+                    break;
+            }
+
+            if (eventTime > now - 200)
+                sendMessageNow (message->message);
+        }
+        else
+        {
+            jassert (timeToWait < 1000 * 30);
+            wait ((int) timeToWait);
+        }
+    }
+
+    clearAllPendingMessages();
+}

source/modules/juce_audio_devices/midi_io/juce_MidiOutput.h

@@ -0,0 +1,147 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#ifndef JUCE_MIDIOUTPUT_H_INCLUDED
+#define JUCE_MIDIOUTPUT_H_INCLUDED
+
+
+//==============================================================================
+/**
+    Controls a physical MIDI output device.
+
+    To create one of these, use the static getDevices() method to get a list of the
+    available output devices, then use the openDevice() method to try to open one.
+
+    @see MidiInput
+*/
+class JUCE_API  MidiOutput  : private Thread
+{
+public:
+    //==============================================================================
+    /** Returns a list of the available midi output devices.
+
+        You can open one of the devices by passing its index into the
+        openDevice() method.
+
+        @see getDefaultDeviceIndex, openDevice
+    */
+    static StringArray getDevices();
+
+    /** Returns the index of the default midi output device to use.
+
+        This refers to the index in the list returned by getDevices().
+    */
+    static int getDefaultDeviceIndex();
+
+    /** Tries to open one of the midi output devices.
+
+        This will return a MidiOutput object if it manages to open it. You can then
+        send messages to this device, and delete it when no longer needed.
+
+        If the device can't be opened, this will return a null pointer.
+
+        @param deviceIndex  the index of a device from the list returned by getDevices()
+        @see getDevices
+    */
+    static MidiOutput* openDevice (int deviceIndex);
+
+
+   #if JUCE_LINUX || JUCE_MAC || JUCE_IOS || DOXYGEN
+    /** This will try to create a new midi output device (Not available on Windows).
+
+        This will attempt to create a new midi output device that other apps can connect
+        to and use as their midi input.
+
+        Returns nullptr if a device can't be created.
+
+        @param deviceName   the name to use for the new device
+    */
+    static MidiOutput* createNewDevice (const String& deviceName);
+   #endif
+
+    //==============================================================================
+    /** Destructor. */
+    virtual ~MidiOutput();
+
+    /** Makes this device output a midi message.
+
+        @see MidiMessage
+    */
+    virtual void sendMessageNow (const MidiMessage& message);
+
+    //==============================================================================
+    /** This lets you supply a block of messages that will be sent out at some point
+        in the future.
+
+        The MidiOutput class has an internal thread that can send out timestamped
+        messages - this appends a set of messages to its internal buffer, ready for
+        sending.
+
+        This will only work if you've already started the thread with startBackgroundThread().
+
+        A time is supplied, at which the block of messages should be sent. This time uses
+        the same time base as Time::getMillisecondCounter(), and must be in the future.
+
+        The samplesPerSecondForBuffer parameter indicates the number of samples per second
+        used by the MidiBuffer. Each event in a MidiBuffer has a sample position, and the
+        samplesPerSecondForBuffer value is needed to convert this sample position to a
+        real time.
+    */
+    virtual void sendBlockOfMessages (const MidiBuffer& buffer,
+                                      double millisecondCounterToStartAt,
+                                      double samplesPerSecondForBuffer);
+
+    /** Gets rid of any midi messages that had been added by sendBlockOfMessages().
+    */
+    virtual void clearAllPendingMessages();
+
+    /** Starts up a background thread so that the device can send blocks of data.
+
+        Call this to get the device ready, before using sendBlockOfMessages().
+    */
+    virtual void startBackgroundThread();
+
+    /** Stops the background thread, and clears any pending midi events.
+
+        @see startBackgroundThread
+    */
+    virtual void stopBackgroundThread();
+
+
+protected:
+    //==============================================================================
+    void* internal;
+    CriticalSection lock;
+    struct PendingMessage;
+    PendingMessage* firstMessage;
+
+    MidiOutput();
+    void run() override;
+
+private:
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (MidiOutput)
+};
+
+
+#endif   // JUCE_MIDIOUTPUT_H_INCLUDED

source/modules/juce_audio_devices/native/juce_MidiDataConcatenator.h

@@ -0,0 +1,175 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#ifndef JUCE_MIDIDATACONCATENATOR_H_INCLUDED
+#define JUCE_MIDIDATACONCATENATOR_H_INCLUDED
+
+//==============================================================================
+/**
+    Helper class that takes chunks of incoming midi bytes, packages them into
+    messages, and dispatches them to a midi callback.
+*/
+class MidiDataConcatenator
+{
+public:
+    //==============================================================================
+    MidiDataConcatenator (const int initialBufferSize)
+        : pendingData ((size_t) initialBufferSize),
+          pendingDataTime (0), pendingBytes (0), runningStatus (0)
+    {
+    }
+
+    void reset()
+    {
+        pendingBytes = 0;
+        runningStatus = 0;
+        pendingDataTime = 0;
+    }
+
+    template <typename UserDataType, typename CallbackType>
+    void pushMidiData (const void* inputData, int numBytes, double time,
+                       UserDataType* input, CallbackType& callback)
+    {
+        const uint8* d = static_cast <const uint8*> (inputData);
+
+        while (numBytes > 0)
+        {
+            if (pendingBytes > 0 || d[0] == 0xf0)
+            {
+                processSysex (d, numBytes, time, input, callback);
+                runningStatus = 0;
+            }
+            else
+            {
+                int len = 0;
+                uint8 data[3];
+
+                while (numBytes > 0)
+                {
+                    // If there's a realtime message embedded in the middle of
+                    // the normal message, handle it now..
+                    if (*d >= 0xf8 && *d <= 0xfe)
+                    {
+                        const MidiMessage m (*d++, time);
+                        callback.handleIncomingMidiMessage (input, m);
+                        --numBytes;
+                    }
+                    else
+                    {
+                        if (len == 0 && *d < 0x80 && runningStatus >= 0x80)
+                            data[len++] = runningStatus;
+
+                        data[len++] = *d++;
+                        --numBytes;
+
+                        if (len >= MidiMessage::getMessageLengthFromFirstByte (data[0]))
+                            break;
+                    }
+                }
+
+                if (len > 0)
+                {
+                    int used = 0;
+                    const MidiMessage m (data, len, used, 0, time);
+
+                    if (used <= 0)
+                        break; // malformed message..
+
+                    jassert (used == len);
+                    callback.handleIncomingMidiMessage (input, m);
+                    runningStatus = data[0];
+                }
+            }
+        }
+    }
+
+private:
+    template <typename UserDataType, typename CallbackType>
+    void processSysex (const uint8*& d, int& numBytes, double time,
+                       UserDataType* input, CallbackType& callback)
+    {
+        if (*d == 0xf0)
+        {
+            pendingBytes = 0;
+            pendingDataTime = time;
+        }
+
+        pendingData.ensureSize ((size_t) (pendingBytes + numBytes), false);
+        uint8* totalMessage = static_cast<uint8*> (pendingData.getData());
+        uint8* dest = totalMessage + pendingBytes;
+
+        do
+        {
+            if (pendingBytes > 0 && *d >= 0x80)
+            {
+                if (*d >= 0xfa || *d == 0xf8)
+                {
+                    callback.handleIncomingMidiMessage (input, MidiMessage (*d, time));
+                    ++d;
+                    --numBytes;
+                }
+                else
+                {
+                    if (*d == 0xf7)
+                    {
+                        *dest++ = *d++;
+                        pendingBytes++;
+                        --numBytes;
+                    }
+
+                    break;
+                }
+            }
+            else
+            {
+                *dest++ = *d++;
+                pendingBytes++;
+                --numBytes;
+            }
+        }
+        while (numBytes > 0);
+
+        if (pendingBytes > 0)
+        {
+            if (totalMessage [pendingBytes - 1] == 0xf7)
+            {
+                callback.handleIncomingMidiMessage (input, MidiMessage (totalMessage, pendingBytes, pendingDataTime));
+                pendingBytes = 0;
+            }
+            else
+            {
+                callback.handlePartialSysexMessage (input, totalMessage, pendingBytes, pendingDataTime);
+            }
+        }
+    }
+
+    MemoryBlock pendingData;
+    double pendingDataTime;
+    int pendingBytes;
+    uint8 runningStatus;
+
+    JUCE_DECLARE_NON_COPYABLE (MidiDataConcatenator)
+};
+
+#endif   // JUCE_MIDIDATACONCATENATOR_H_INCLUDED

source/modules/juce_audio_devices/native/juce_android_Audio.cpp

@@ -0,0 +1,443 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+//==============================================================================
+#define JNI_CLASS_MEMBERS(METHOD, STATICMETHOD, FIELD, STATICFIELD) \
+ STATICMETHOD (getMinBufferSize,            "getMinBufferSize",             "(III)I") \
+ STATICMETHOD (getNativeOutputSampleRate,   "getNativeOutputSampleRate",    "(I)I") \
+ METHOD (constructor,   "<init>",   "(IIIIII)V") \
+ METHOD (getState,      "getState", "()I") \
+ METHOD (play,          "play",     "()V") \
+ METHOD (stop,          "stop",     "()V") \
+ METHOD (release,       "release",  "()V") \
+ METHOD (flush,         "flush",    "()V") \
+ METHOD (write,         "write",    "([SII)I") \
+
+DECLARE_JNI_CLASS (AudioTrack, "android/media/AudioTrack");
+#undef JNI_CLASS_MEMBERS
+
+//==============================================================================
+#define JNI_CLASS_MEMBERS(METHOD, STATICMETHOD, FIELD, STATICFIELD) \
+ STATICMETHOD (getMinBufferSize, "getMinBufferSize", "(III)I") \
+ METHOD (constructor,       "<init>",           "(IIIII)V") \
+ METHOD (getState,          "getState",         "()I") \
+ METHOD (startRecording,    "startRecording",   "()V") \
+ METHOD (stop,              "stop",             "()V") \
+ METHOD (read,              "read",             "([SII)I") \
+ METHOD (release,           "release",          "()V") \
+
+DECLARE_JNI_CLASS (AudioRecord, "android/media/AudioRecord");
+#undef JNI_CLASS_MEMBERS
+
+//==============================================================================
+enum
+{
+    CHANNEL_OUT_STEREO  = 12,
+    CHANNEL_IN_STEREO   = 12,
+    CHANNEL_IN_MONO     = 16,
+    ENCODING_PCM_16BIT  = 2,
+    STREAM_MUSIC        = 3,
+    MODE_STREAM         = 1,
+    STATE_UNINITIALIZED = 0
+};
+
+const char* const javaAudioTypeName = "Android Audio";
+
+//==============================================================================
+class AndroidAudioIODevice  : public AudioIODevice,
+                              public Thread
+{
+public:
+    //==============================================================================
+    AndroidAudioIODevice (const String& deviceName)
+        : AudioIODevice (deviceName, javaAudioTypeName),
+          Thread ("audio"),
+          minBufferSizeOut (0), minBufferSizeIn (0), callback (0), sampleRate (0),
+          numClientInputChannels (0), numDeviceInputChannels (0), numDeviceInputChannelsAvailable (2),
+          numClientOutputChannels (0), numDeviceOutputChannels (0),
+          actualBufferSize (0), isRunning (false),
+          inputChannelBuffer (1, 1),
+          outputChannelBuffer (1, 1)
+    {
+        JNIEnv* env = getEnv();
+        sampleRate = env->CallStaticIntMethod (AudioTrack, AudioTrack.getNativeOutputSampleRate, MODE_STREAM);
+
+        minBufferSizeOut = (int) env->CallStaticIntMethod (AudioTrack,  AudioTrack.getMinBufferSize,  sampleRate, CHANNEL_OUT_STEREO, ENCODING_PCM_16BIT);
+        minBufferSizeIn  = (int) env->CallStaticIntMethod (AudioRecord, AudioRecord.getMinBufferSize, sampleRate, CHANNEL_IN_STEREO,  ENCODING_PCM_16BIT);
+
+        if (minBufferSizeIn <= 0)
+        {
+            minBufferSizeIn = env->CallStaticIntMethod (AudioRecord, AudioRecord.getMinBufferSize, sampleRate, CHANNEL_IN_MONO, ENCODING_PCM_16BIT);
+
+            if (minBufferSizeIn > 0)
+                numDeviceInputChannelsAvailable = 1;
+            else
+                numDeviceInputChannelsAvailable = 0;
+        }
+
+        DBG ("Audio device - min buffers: " << minBufferSizeOut << ", " << minBufferSizeIn << "; "
+              << sampleRate << " Hz; input chans: " << numDeviceInputChannelsAvailable);
+    }
+
+    ~AndroidAudioIODevice()
+    {
+        close();
+    }
+
+    StringArray getOutputChannelNames()
+    {
+        StringArray s;
+        s.add ("Left");
+        s.add ("Right");
+        return s;
+    }
+
+    StringArray getInputChannelNames()
+    {
+        StringArray s;
+
+        if (numDeviceInputChannelsAvailable == 2)
+        {
+            s.add ("Left");
+            s.add ("Right");
+        }
+        else if (numDeviceInputChannelsAvailable == 1)
+        {
+            s.add ("Audio Input");
+        }
+
+        return s;
+    }
+
+    int getNumSampleRates()             { return 1;}
+    double getSampleRate (int index)    { return sampleRate; }
+
+    int getDefaultBufferSize()          { return 2048; }
+    int getNumBufferSizesAvailable()    { return 50; }
+
+    int getBufferSizeSamples (int index)
+    {
+        int n = 16;
+        for (int i = 0; i < index; ++i)
+            n += n < 64 ? 16
+                        : (n < 512 ? 32
+                                   : (n < 1024 ? 64
+                                               : (n < 2048 ? 128 : 256)));
+
+        return n;
+    }
+
+
+    String open (const BigInteger& inputChannels,
+                 const BigInteger& outputChannels,
+                 double requestedSampleRate,
+                 int bufferSize)
+    {
+        close();
+
+        if (sampleRate != (int) requestedSampleRate)
+            return "Sample rate not allowed";
+
+        lastError = String::empty;
+        int preferredBufferSize = (bufferSize <= 0) ? getDefaultBufferSize() : bufferSize;
+
+        numDeviceInputChannels = 0;
+        numDeviceOutputChannels = 0;
+
+        activeOutputChans = outputChannels;
+        activeOutputChans.setRange (2, activeOutputChans.getHighestBit(), false);
+        numClientOutputChannels = activeOutputChans.countNumberOfSetBits();
+
+        activeInputChans = inputChannels;
+        activeInputChans.setRange (2, activeInputChans.getHighestBit(), false);
+        numClientInputChannels = activeInputChans.countNumberOfSetBits();
+
+        actualBufferSize = preferredBufferSize;
+        inputChannelBuffer.setSize (2, actualBufferSize);
+        inputChannelBuffer.clear();
+        outputChannelBuffer.setSize (2, actualBufferSize);
+        outputChannelBuffer.clear();
+
+        JNIEnv* env = getEnv();
+
+        if (numClientOutputChannels > 0)
+        {
+            numDeviceOutputChannels = 2;
+            outputDevice = GlobalRef (env->NewObject (AudioTrack, AudioTrack.constructor,
+                                                      STREAM_MUSIC, sampleRate, CHANNEL_OUT_STEREO, ENCODING_PCM_16BIT,
+                                                      (jint) (minBufferSizeOut * numDeviceOutputChannels * sizeof (int16)), MODE_STREAM));
+
+            if (env->CallIntMethod (outputDevice, AudioTrack.getState) != STATE_UNINITIALIZED)
+                isRunning = true;
+            else
+                outputDevice.clear(); // failed to open the device
+        }
+
+        if (numClientInputChannels > 0 && numDeviceInputChannelsAvailable > 0)
+        {
+            numDeviceInputChannels = jmin (numClientInputChannels, numDeviceInputChannelsAvailable);
+            inputDevice = GlobalRef (env->NewObject (AudioRecord, AudioRecord.constructor,
+                                                     0 /* (default audio source) */, sampleRate,
+                                                     numDeviceInputChannelsAvailable > 1 ? CHANNEL_IN_STEREO : CHANNEL_IN_MONO,
+                                                     ENCODING_PCM_16BIT,
+                                                     (jint) (minBufferSizeIn * numDeviceInputChannels * sizeof (int16))));
+
+            if (env->CallIntMethod (inputDevice, AudioRecord.getState) != STATE_UNINITIALIZED)
+                isRunning = true;
+            else
+                inputDevice.clear(); // failed to open the device
+        }
+
+        if (isRunning)
+        {
+            if (outputDevice != nullptr)
+                env->CallVoidMethod (outputDevice, AudioTrack.play);
+
+            if (inputDevice != nullptr)
+                env->CallVoidMethod (inputDevice, AudioRecord.startRecording);
+
+            startThread (8);
+        }
+        else
+        {
+            closeDevices();
+        }
+
+        return lastError;
+    }
+
+    void close()
+    {
+        if (isRunning)
+        {
+            stopThread (2000);
+            isRunning = false;
+            closeDevices();
+        }
+    }
+
+    int getOutputLatencyInSamples()                     { return (minBufferSizeOut * 3) / 4; }
+    int getInputLatencyInSamples()                      { return (minBufferSizeIn * 3) / 4; }
+    bool isOpen()                                       { return isRunning; }
+    int getCurrentBufferSizeSamples()                   { return actualBufferSize; }
+    int getCurrentBitDepth()                            { return 16; }
+    double getCurrentSampleRate()                       { return sampleRate; }
+    BigInteger getActiveOutputChannels() const          { return activeOutputChans; }
+    BigInteger getActiveInputChannels() const           { return activeInputChans; }
+    String getLastError()                               { return lastError; }
+    bool isPlaying()                                    { return isRunning && callback != 0; }
+
+    void start (AudioIODeviceCallback* newCallback)
+    {
+        if (isRunning && callback != newCallback)
+        {
+            if (newCallback != nullptr)
+                newCallback->audioDeviceAboutToStart (this);
+
+            const ScopedLock sl (callbackLock);
+            callback = newCallback;
+        }
+    }
+
+    void stop()
+    {
+        if (isRunning)
+        {
+            AudioIODeviceCallback* lastCallback;
+
+            {
+                const ScopedLock sl (callbackLock);
+                lastCallback = callback;
+                callback = nullptr;
+            }
+
+            if (lastCallback != nullptr)
+                lastCallback->audioDeviceStopped();
+        }
+    }
+
+    void run() override
+    {
+        JNIEnv* env = getEnv();
+        jshortArray audioBuffer = env->NewShortArray (actualBufferSize * jmax (numDeviceOutputChannels, numDeviceInputChannels));
+
+        while (! threadShouldExit())
+        {
+            if (inputDevice != nullptr)
+            {
+                jint numRead = env->CallIntMethod (inputDevice, AudioRecord.read, audioBuffer, 0, actualBufferSize * numDeviceInputChannels);
+
+                if (numRead < actualBufferSize * numDeviceInputChannels)
+                {
+                    DBG ("Audio read under-run! " << numRead);
+                }
+
+                jshort* const src = env->GetShortArrayElements (audioBuffer, 0);
+
+                for (int chan = 0; chan < inputChannelBuffer.getNumChannels(); ++chan)
+                {
+                    AudioData::Pointer <AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::NonConst> d (inputChannelBuffer.getSampleData (chan));
+
+                    if (chan < numDeviceInputChannels)
+                    {
+                        AudioData::Pointer <AudioData::Int16, AudioData::NativeEndian, AudioData::Interleaved, AudioData::Const> s (src + chan, numDeviceInputChannels);
+                        d.convertSamples (s, actualBufferSize);
+                    }
+                    else
+                    {
+                        d.clearSamples (actualBufferSize);
+                    }
+                }
+
+                env->ReleaseShortArrayElements (audioBuffer, src, 0);
+            }
+
+            if (threadShouldExit())
+                break;
+
+            {
+                const ScopedLock sl (callbackLock);
+
+                if (callback != nullptr)
+                {
+                    callback->audioDeviceIOCallback ((const float**) inputChannelBuffer.getArrayOfChannels(), numClientInputChannels,
+                                                     outputChannelBuffer.getArrayOfChannels(), numClientOutputChannels,
+                                                     actualBufferSize);
+                }
+                else
+                {
+                    outputChannelBuffer.clear();
+                }
+            }
+
+            if (outputDevice != nullptr)
+            {
+                if (threadShouldExit())
+                    break;
+
+                jshort* const dest = env->GetShortArrayElements (audioBuffer, 0);
+
+                for (int chan = 0; chan < numDeviceOutputChannels; ++chan)
+                {
+                    AudioData::Pointer <AudioData::Int16, AudioData::NativeEndian, AudioData::Interleaved, AudioData::NonConst> d (dest + chan, numDeviceOutputChannels);
+
+                    const float* const sourceChanData = outputChannelBuffer.getSampleData (jmin (chan, outputChannelBuffer.getNumChannels() - 1));
+                    AudioData::Pointer <AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::Const> s (sourceChanData);
+                    d.convertSamples (s, actualBufferSize);
+                }
+
+                env->ReleaseShortArrayElements (audioBuffer, dest, 0);
+                jint numWritten = env->CallIntMethod (outputDevice, AudioTrack.write, audioBuffer, 0, actualBufferSize * numDeviceOutputChannels);
+
+                if (numWritten < actualBufferSize * numDeviceOutputChannels)
+                {
+                    DBG ("Audio write underrun! " << numWritten);
+                }
+            }
+        }
+    }
+
+    int minBufferSizeOut, minBufferSizeIn;
+
+private:
+    //==================================================================================================
+    CriticalSection callbackLock;
+    AudioIODeviceCallback* callback;
+    jint sampleRate;
+    int numClientInputChannels, numDeviceInputChannels, numDeviceInputChannelsAvailable;
+    int numClientOutputChannels, numDeviceOutputChannels;
+    int actualBufferSize;
+    bool isRunning;
+    String lastError;
+    BigInteger activeOutputChans, activeInputChans;
+    GlobalRef outputDevice, inputDevice;
+    AudioSampleBuffer inputChannelBuffer, outputChannelBuffer;
+
+    void closeDevices()
+    {
+        if (outputDevice != nullptr)
+        {
+            outputDevice.callVoidMethod (AudioTrack.stop);
+            outputDevice.callVoidMethod (AudioTrack.release);
+            outputDevice.clear();
+        }
+
+        if (inputDevice != nullptr)
+        {
+            inputDevice.callVoidMethod (AudioRecord.stop);
+            inputDevice.callVoidMethod (AudioRecord.release);
+            inputDevice.clear();
+        }
+    }
+
+    JUCE_DECLARE_NON_COPYABLE (AndroidAudioIODevice)
+};
+
+//==============================================================================
+class AndroidAudioIODeviceType  : public AudioIODeviceType
+{
+public:
+    AndroidAudioIODeviceType()  : AudioIODeviceType (javaAudioTypeName) {}
+
+    //==============================================================================
+    void scanForDevices() {}
+    StringArray getDeviceNames (bool wantInputNames) const              { return StringArray (javaAudioTypeName); }
+    int getDefaultDeviceIndex (bool forInput) const                     { return 0; }
+    int getIndexOfDevice (AudioIODevice* device, bool asInput) const    { return device != nullptr ? 0 : -1; }
+    bool hasSeparateInputsAndOutputs() const                            { return false; }
+
+    AudioIODevice* createDevice (const String& outputDeviceName,
+                                 const String& inputDeviceName)
+    {
+        ScopedPointer<AndroidAudioIODevice> dev;
+
+        if (outputDeviceName.isNotEmpty() || inputDeviceName.isNotEmpty())
+        {
+            dev = new AndroidAudioIODevice (outputDeviceName.isNotEmpty() ? outputDeviceName
+                                                                          : inputDeviceName);
+
+            if (dev->getCurrentSampleRate() <= 0 || dev->getDefaultBufferSize() <= 0)
+                dev = nullptr;
+        }
+
+        return dev.release();
+    }
+
+private:
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AndroidAudioIODeviceType)
+};
+
+
+//==============================================================================
+extern bool isOpenSLAvailable();
+
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_Android()
+{
+   #if JUCE_USE_ANDROID_OPENSLES
+    if (isOpenSLAvailable())
+        return nullptr;
+   #endif
+
+    return new AndroidAudioIODeviceType();
+}

source/modules/juce_audio_devices/native/juce_android_Midi.cpp

@@ -0,0 +1,84 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+StringArray MidiOutput::getDevices()
+{
+    StringArray devices;
+
+    return devices;
+}
+
+int MidiOutput::getDefaultDeviceIndex()
+{
+    return 0;
+}
+
+MidiOutput* MidiOutput::openDevice (int index)
+{
+    return nullptr;
+}
+
+MidiOutput::~MidiOutput()
+{
+}
+
+void MidiOutput::sendMessageNow (const MidiMessage&)
+{
+}
+
+//==============================================================================
+MidiInput::MidiInput (const String& name_)
+    : name (name_),
+      internal (0)
+{
+}
+
+MidiInput::~MidiInput()
+{
+}
+
+void MidiInput::start()
+{
+}
+
+void MidiInput::stop()
+{
+}
+
+int MidiInput::getDefaultDeviceIndex()
+{
+    return 0;
+}
+
+StringArray MidiInput::getDevices()
+{
+    StringArray devs;
+
+    return devs;
+}
+
+MidiInput* MidiInput::openDevice (int index, MidiInputCallback* callback)
+{
+    return nullptr;
+}

source/modules/juce_audio_devices/native/juce_android_OpenSL.cpp

@@ -0,0 +1,622 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+const char* const openSLTypeName = "Android OpenSL";
+
+bool isOpenSLAvailable()
+{
+    DynamicLibrary library;
+    return library.open ("libOpenSLES.so");
+}
+
+const unsigned short openSLRates[]       = { 8000, 16000, 32000, 44100, 48000 };
+const unsigned short openSLBufferSizes[] = { 256, 512, 768, 1024, 1280, 1600 }; // must all be multiples of the block size
+
+//==============================================================================
+class OpenSLAudioIODevice  : public AudioIODevice,
+                             public Thread
+{
+public:
+    OpenSLAudioIODevice (const String& deviceName)
+        : AudioIODevice (deviceName, openSLTypeName),
+          Thread ("OpenSL"),
+          callback (nullptr), sampleRate (0), deviceOpen (false),
+          inputBuffer (2, 2), outputBuffer (2, 2)
+    {
+        // OpenSL has piss-poor support for determining latency, so the only way I can find to
+        // get a number for this is by asking the AudioTrack/AudioRecord classes..
+        AndroidAudioIODevice javaDevice (String::empty);
+
+        // this is a total guess about how to calculate the latency, but seems to vaguely agree
+        // with the devices I've tested.. YMMV
+        inputLatency  = ((javaDevice.minBufferSizeIn  * 2) / 3);
+        outputLatency = ((javaDevice.minBufferSizeOut * 2) / 3);
+
+        const int longestLatency = jmax (inputLatency, outputLatency);
+        const int totalLatency = inputLatency + outputLatency;
+        inputLatency  = ((longestLatency * inputLatency)  / totalLatency) & ~15;
+        outputLatency = ((longestLatency * outputLatency) / totalLatency) & ~15;
+    }
+
+    ~OpenSLAudioIODevice()
+    {
+        close();
+    }
+
+    bool openedOk() const       { return engine.outputMixObject != nullptr; }
+
+    StringArray getOutputChannelNames()
+    {
+        StringArray s;
+        s.add ("Left");
+        s.add ("Right");
+        return s;
+    }
+
+    StringArray getInputChannelNames()
+    {
+        StringArray s;
+        s.add ("Audio Input");
+        return s;
+    }
+
+    int getNumSampleRates()                 { return numElementsInArray (openSLRates); }
+
+    double getSampleRate (int index)
+    {
+        jassert (index >= 0 && index < getNumSampleRates());
+        return (int) openSLRates [index];
+    }
+
+    int getDefaultBufferSize()              { return 1024; }
+    int getNumBufferSizesAvailable()        { return numElementsInArray (openSLBufferSizes); }
+
+    int getBufferSizeSamples (int index)
+    {
+        jassert (index >= 0 && index < getNumBufferSizesAvailable());
+        return (int) openSLBufferSizes [index];
+    }
+
+    String open (const BigInteger& inputChannels,
+                 const BigInteger& outputChannels,
+                 double requestedSampleRate,
+                 int bufferSize)
+    {
+        close();
+
+        lastError = String::empty;
+        sampleRate = (int) requestedSampleRate;
+
+        int preferredBufferSize = (bufferSize <= 0) ? getDefaultBufferSize() : bufferSize;
+
+        activeOutputChans = outputChannels;
+        activeOutputChans.setRange (2, activeOutputChans.getHighestBit(), false);
+        numOutputChannels = activeOutputChans.countNumberOfSetBits();
+
+        activeInputChans = inputChannels;
+        activeInputChans.setRange (1, activeInputChans.getHighestBit(), false);
+        numInputChannels = activeInputChans.countNumberOfSetBits();
+
+        actualBufferSize = preferredBufferSize;
+
+        inputBuffer.setSize  (jmax (1, numInputChannels),  actualBufferSize);
+        outputBuffer.setSize (jmax (1, numOutputChannels), actualBufferSize);
+        outputBuffer.clear();
+
+        recorder = engine.createRecorder (numInputChannels,  sampleRate);
+        player   = engine.createPlayer   (numOutputChannels, sampleRate);
+
+        startThread (8);
+
+        deviceOpen = true;
+        return lastError;
+    }
+
+    void close()
+    {
+        stop();
+        stopThread (6000);
+        deviceOpen = false;
+        recorder = nullptr;
+        player = nullptr;
+    }
+
+    int getOutputLatencyInSamples()                     { return outputLatency; }
+    int getInputLatencyInSamples()                      { return inputLatency; }
+    bool isOpen()                                       { return deviceOpen; }
+    int getCurrentBufferSizeSamples()                   { return actualBufferSize; }
+    int getCurrentBitDepth()                            { return 16; }
+    double getCurrentSampleRate()                       { return sampleRate; }
+    BigInteger getActiveOutputChannels() const          { return activeOutputChans; }
+    BigInteger getActiveInputChannels() const           { return activeInputChans; }
+    String getLastError()                               { return lastError; }
+    bool isPlaying()                                    { return callback != nullptr; }
+
+    void start (AudioIODeviceCallback* newCallback)
+    {
+        stop();
+
+        if (deviceOpen && callback != newCallback)
+        {
+            if (newCallback != nullptr)
+                newCallback->audioDeviceAboutToStart (this);
+
+            setCallback (newCallback);
+        }
+    }
+
+    void stop()
+    {
+        if (AudioIODeviceCallback* const oldCallback = setCallback (nullptr))
+            oldCallback->audioDeviceStopped();
+    }
+
+    void run() override
+    {
+        if (recorder != nullptr)    recorder->start();
+        if (player != nullptr)      player->start();
+
+        while (! threadShouldExit())
+        {
+            if (player != nullptr)      player->writeBuffer (outputBuffer, *this);
+            if (recorder != nullptr)    recorder->readNextBlock (inputBuffer, *this);
+
+            const ScopedLock sl (callbackLock);
+
+            if (callback != nullptr)
+            {
+                callback->audioDeviceIOCallback (numInputChannels > 0 ? (const float**) inputBuffer.getArrayOfChannels() : nullptr,
+                                                 numInputChannels,
+                                                 numOutputChannels > 0 ? outputBuffer.getArrayOfChannels() : nullptr,
+                                                 numOutputChannels,
+                                                 actualBufferSize);
+            }
+            else
+            {
+                outputBuffer.clear();
+            }
+        }
+    }
+
+private:
+    //==================================================================================================
+    CriticalSection callbackLock;
+    AudioIODeviceCallback* callback;
+    int actualBufferSize, sampleRate;
+    int inputLatency, outputLatency;
+    bool deviceOpen;
+    String lastError;
+    BigInteger activeOutputChans, activeInputChans;
+    int numInputChannels, numOutputChannels;
+    AudioSampleBuffer inputBuffer, outputBuffer;
+    struct Player;
+    struct Recorder;
+
+    AudioIODeviceCallback* setCallback (AudioIODeviceCallback* const newCallback)
+    {
+        const ScopedLock sl (callbackLock);
+        AudioIODeviceCallback* const oldCallback = callback;
+        callback = newCallback;
+        return oldCallback;
+    }
+
+    //==================================================================================================
+    struct Engine
+    {
+        Engine()
+            : engineObject (nullptr), engineInterface (nullptr), outputMixObject (nullptr)
+        {
+            if (library.open ("libOpenSLES.so"))
+            {
+                typedef SLresult (*CreateEngineFunc) (SLObjectItf*, SLuint32, const SLEngineOption*, SLuint32, const SLInterfaceID*, const SLboolean*);
+
+                if (CreateEngineFunc createEngine = (CreateEngineFunc) library.getFunction ("slCreateEngine"))
+                {
+                    check (createEngine (&engineObject, 0, nullptr, 0, nullptr, nullptr));
+
+                    SLInterfaceID* SL_IID_ENGINE    = (SLInterfaceID*) library.getFunction ("SL_IID_ENGINE");
+                    SL_IID_ANDROIDSIMPLEBUFFERQUEUE = (SLInterfaceID*) library.getFunction ("SL_IID_ANDROIDSIMPLEBUFFERQUEUE");
+                    SL_IID_PLAY                     = (SLInterfaceID*) library.getFunction ("SL_IID_PLAY");
+                    SL_IID_RECORD                   = (SLInterfaceID*) library.getFunction ("SL_IID_RECORD");
+
+                    check ((*engineObject)->Realize (engineObject, SL_BOOLEAN_FALSE));
+                    check ((*engineObject)->GetInterface (engineObject, *SL_IID_ENGINE, &engineInterface));
+
+                    check ((*engineInterface)->CreateOutputMix (engineInterface, &outputMixObject, 0, nullptr, nullptr));
+                    check ((*outputMixObject)->Realize (outputMixObject, SL_BOOLEAN_FALSE));
+                }
+            }
+        }
+
+        ~Engine()
+        {
+            if (outputMixObject != nullptr) (*outputMixObject)->Destroy (outputMixObject);
+            if (engineObject != nullptr)    (*engineObject)->Destroy (engineObject);
+        }
+
+        Player* createPlayer (const int numChannels, const int sampleRate)
+        {
+            if (numChannels <= 0)
+                return nullptr;
+
+            ScopedPointer<Player> player (new Player (numChannels, sampleRate, *this));
+            return player->openedOk() ? player.release() : nullptr;
+        }
+
+        Recorder* createRecorder (const int numChannels, const int sampleRate)
+        {
+            if (numChannels <= 0)
+                return nullptr;
+
+            ScopedPointer<Recorder> recorder (new Recorder (numChannels, sampleRate, *this));
+            return recorder->openedOk() ? recorder.release() : nullptr;
+        }
+
+        SLObjectItf engineObject;
+        SLEngineItf engineInterface;
+        SLObjectItf outputMixObject;
+
+        SLInterfaceID* SL_IID_ANDROIDSIMPLEBUFFERQUEUE;
+        SLInterfaceID* SL_IID_PLAY;
+        SLInterfaceID* SL_IID_RECORD;
+
+    private:
+        DynamicLibrary library;
+
+        JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (Engine)
+    };
+
+    //==================================================================================================
+    struct BufferList
+    {
+        BufferList (const int numChannels_)
+            : numChannels (numChannels_), bufferSpace (numChannels_ * numSamples * numBuffers), nextBlock (0)
+        {
+        }
+
+        int16* waitForFreeBuffer (Thread& threadToCheck)
+        {
+            while (numBlocksOut.get() == numBuffers)
+            {
+                dataArrived.wait (1);
+
+                if (threadToCheck.threadShouldExit())
+                    return nullptr;
+            }
+
+            return getNextBuffer();
+        }
+
+        int16* getNextBuffer()
+        {
+            if (++nextBlock == numBuffers)
+                nextBlock = 0;
+
+            return bufferSpace + nextBlock * numChannels * numSamples;
+        }
+
+        void bufferReturned()           { --numBlocksOut; dataArrived.signal(); }
+        void bufferSent()               { ++numBlocksOut; dataArrived.signal(); }
+
+        int getBufferSizeBytes() const  { return numChannels * numSamples * sizeof (int16); }
+
+        const int numChannels;
+        enum { numSamples = 256, numBuffers = 16 };
+
+    private:
+        HeapBlock<int16> bufferSpace;
+        int nextBlock;
+        Atomic<int> numBlocksOut;
+        WaitableEvent dataArrived;
+    };
+
+    //==================================================================================================
+    struct Player
+    {
+        Player (int numChannels, int sampleRate, Engine& engine)
+            : playerObject (nullptr), playerPlay (nullptr), playerBufferQueue (nullptr),
+              bufferList (numChannels)
+        {
+            jassert (numChannels == 2);
+
+            SLDataFormat_PCM pcmFormat =
+            {
+                SL_DATAFORMAT_PCM,
+                numChannels,
+                sampleRate * 1000, // (sample rate units are millihertz)
+                SL_PCMSAMPLEFORMAT_FIXED_16,
+                SL_PCMSAMPLEFORMAT_FIXED_16,
+                SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT,
+                SL_BYTEORDER_LITTLEENDIAN
+            };
+
+            SLDataLocator_AndroidSimpleBufferQueue bufferQueue = { SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, bufferList.numBuffers };
+            SLDataSource audioSrc = { &bufferQueue, &pcmFormat };
+
+            SLDataLocator_OutputMix outputMix = { SL_DATALOCATOR_OUTPUTMIX, engine.outputMixObject };
+            SLDataSink audioSink = { &outputMix, nullptr };
+
+            // (SL_IID_BUFFERQUEUE is not guaranteed to remain future-proof, so use SL_IID_ANDROIDSIMPLEBUFFERQUEUE)
+            const SLInterfaceID interfaceIDs[] = { *engine.SL_IID_ANDROIDSIMPLEBUFFERQUEUE };
+            const SLboolean flags[] = { SL_BOOLEAN_TRUE };
+
+            check ((*engine.engineInterface)->CreateAudioPlayer (engine.engineInterface, &playerObject, &audioSrc, &audioSink,
+                                                                 1, interfaceIDs, flags));
+
+            check ((*playerObject)->Realize (playerObject, SL_BOOLEAN_FALSE));
+            check ((*playerObject)->GetInterface (playerObject, *engine.SL_IID_PLAY, &playerPlay));
+            check ((*playerObject)->GetInterface (playerObject, *engine.SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &playerBufferQueue));
+            check ((*playerBufferQueue)->RegisterCallback (playerBufferQueue, staticCallback, this));
+        }
+
+        ~Player()
+        {
+            if (playerPlay != nullptr)
+                check ((*playerPlay)->SetPlayState (playerPlay, SL_PLAYSTATE_STOPPED));
+
+            if (playerBufferQueue != nullptr)
+                check ((*playerBufferQueue)->Clear (playerBufferQueue));
+
+            if (playerObject != nullptr)
+                (*playerObject)->Destroy (playerObject);
+        }
+
+        bool openedOk() const noexcept      { return playerBufferQueue != nullptr; }
+
+        void start()
+        {
+            jassert (openedOk());
+            check ((*playerPlay)->SetPlayState (playerPlay, SL_PLAYSTATE_PLAYING));
+        }
+
+        void writeBuffer (const AudioSampleBuffer& buffer, Thread& thread)
+        {
+            jassert (buffer.getNumChannels() == bufferList.numChannels);
+            jassert (buffer.getNumSamples() < bufferList.numSamples * bufferList.numBuffers);
+
+            int offset = 0;
+            int numSamples = buffer.getNumSamples();
+
+            while (numSamples > 0)
+            {
+                int16* const destBuffer = bufferList.waitForFreeBuffer (thread);
+
+                if (destBuffer == nullptr)
+                    break;
+
+                for (int i = 0; i < bufferList.numChannels; ++i)
+                {
+                    typedef AudioData::Pointer <AudioData::Int16,   AudioData::LittleEndian, AudioData::Interleaved, AudioData::NonConst> DstSampleType;
+                    typedef AudioData::Pointer <AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::Const> SrcSampleType;
+
+                    DstSampleType dstData (destBuffer + i, bufferList.numChannels);
+                    SrcSampleType srcData (buffer.getSampleData (i, offset));
+                    dstData.convertSamples (srcData, bufferList.numSamples);
+                }
+
+                check ((*playerBufferQueue)->Enqueue (playerBufferQueue, destBuffer, bufferList.getBufferSizeBytes()));
+                bufferList.bufferSent();
+
+                numSamples -= bufferList.numSamples;
+                offset += bufferList.numSamples;
+            }
+        }
+
+    private:
+        SLObjectItf playerObject;
+        SLPlayItf playerPlay;
+        SLAndroidSimpleBufferQueueItf playerBufferQueue;
+
+        BufferList bufferList;
+
+        static void staticCallback (SLAndroidSimpleBufferQueueItf queue, void* context)
+        {
+            jassert (queue == static_cast <Player*> (context)->playerBufferQueue); (void) queue;
+            static_cast <Player*> (context)->bufferList.bufferReturned();
+        }
+
+        JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (Player)
+    };
+
+    //==================================================================================================
+    struct Recorder
+    {
+        Recorder (int numChannels, int sampleRate, Engine& engine)
+            : recorderObject (nullptr), recorderRecord (nullptr), recorderBufferQueue (nullptr),
+              bufferList (numChannels)
+        {
+            jassert (numChannels == 1); // STEREO doesn't always work!!
+
+            SLDataFormat_PCM pcmFormat =
+            {
+                SL_DATAFORMAT_PCM,
+                numChannels,
+                sampleRate * 1000, // (sample rate units are millihertz)
+                SL_PCMSAMPLEFORMAT_FIXED_16,
+                SL_PCMSAMPLEFORMAT_FIXED_16,
+                (numChannels == 1) ? SL_SPEAKER_FRONT_CENTER : (SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT),
+                SL_BYTEORDER_LITTLEENDIAN
+            };
+
+            SLDataLocator_IODevice ioDevice = { SL_DATALOCATOR_IODEVICE, SL_IODEVICE_AUDIOINPUT, SL_DEFAULTDEVICEID_AUDIOINPUT, nullptr };
+            SLDataSource audioSrc = { &ioDevice, nullptr };
+
+            SLDataLocator_AndroidSimpleBufferQueue bufferQueue = { SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE, bufferList.numBuffers };
+            SLDataSink audioSink = { &bufferQueue, &pcmFormat };
+
+            const SLInterfaceID interfaceIDs[] = { *engine.SL_IID_ANDROIDSIMPLEBUFFERQUEUE };
+            const SLboolean flags[] = { SL_BOOLEAN_TRUE };
+
+            if (check ((*engine.engineInterface)->CreateAudioRecorder (engine.engineInterface, &recorderObject, &audioSrc,
+                                                                       &audioSink, 1, interfaceIDs, flags)))
+            {
+                if (check ((*recorderObject)->Realize (recorderObject, SL_BOOLEAN_FALSE)))
+                {
+                    check ((*recorderObject)->GetInterface (recorderObject, *engine.SL_IID_RECORD, &recorderRecord));
+                    check ((*recorderObject)->GetInterface (recorderObject, *engine.SL_IID_ANDROIDSIMPLEBUFFERQUEUE, &recorderBufferQueue));
+                    check ((*recorderBufferQueue)->RegisterCallback (recorderBufferQueue, staticCallback, this));
+                    check ((*recorderRecord)->SetRecordState (recorderRecord, SL_RECORDSTATE_STOPPED));
+
+                    for (int i = bufferList.numBuffers; --i >= 0;)
+                    {
+                        int16* const buffer = bufferList.getNextBuffer();
+                        jassert (buffer != nullptr);
+                        enqueueBuffer (buffer);
+                    }
+                }
+            }
+        }
+
+        ~Recorder()
+        {
+            if (recorderRecord != nullptr)
+                check ((*recorderRecord)->SetRecordState (recorderRecord, SL_RECORDSTATE_STOPPED));
+
+            if (recorderBufferQueue != nullptr)
+                check ((*recorderBufferQueue)->Clear (recorderBufferQueue));
+
+            if (recorderObject != nullptr)
+                (*recorderObject)->Destroy (recorderObject);
+        }
+
+        bool openedOk() const noexcept      { return recorderBufferQueue != nullptr; }
+
+        void start()
+        {
+            jassert (openedOk());
+            check ((*recorderRecord)->SetRecordState (recorderRecord, SL_RECORDSTATE_RECORDING));
+        }
+
+        void readNextBlock (AudioSampleBuffer& buffer, Thread& thread)
+        {
+            jassert (buffer.getNumChannels() == bufferList.numChannels);
+            jassert (buffer.getNumSamples() < bufferList.numSamples * bufferList.numBuffers);
+            jassert ((buffer.getNumSamples() % bufferList.numSamples) == 0);
+
+            int offset = 0;
+            int numSamples = buffer.getNumSamples();
+
+            while (numSamples > 0)
+            {
+                int16* const srcBuffer = bufferList.waitForFreeBuffer (thread);
+
+                if (srcBuffer == nullptr)
+                    break;
+
+                for (int i = 0; i < bufferList.numChannels; ++i)
+                {
+                    typedef AudioData::Pointer <AudioData::Float32, AudioData::NativeEndian, AudioData::NonInterleaved, AudioData::NonConst> DstSampleType;
+                    typedef AudioData::Pointer <AudioData::Int16,   AudioData::LittleEndian, AudioData::Interleaved, AudioData::Const> SrcSampleType;
+
+                    DstSampleType dstData (buffer.getSampleData (i, offset));
+                    SrcSampleType srcData (srcBuffer + i, bufferList.numChannels);
+                    dstData.convertSamples (srcData, bufferList.numSamples);
+                }
+
+                enqueueBuffer (srcBuffer);
+
+                numSamples -= bufferList.numSamples;
+                offset += bufferList.numSamples;
+            }
+        }
+
+    private:
+        SLObjectItf recorderObject;
+        SLRecordItf recorderRecord;
+        SLAndroidSimpleBufferQueueItf recorderBufferQueue;
+
+        BufferList bufferList;
+
+        void enqueueBuffer (int16* buffer)
+        {
+            check ((*recorderBufferQueue)->Enqueue (recorderBufferQueue, buffer, bufferList.getBufferSizeBytes()));
+            bufferList.bufferSent();
+        }
+
+        static void staticCallback (SLAndroidSimpleBufferQueueItf queue, void* context)
+        {
+            jassert (queue == static_cast <Recorder*> (context)->recorderBufferQueue); (void) queue;
+            static_cast <Recorder*> (context)->bufferList.bufferReturned();
+        }
+
+        JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (Recorder)
+    };
+
+
+    //==============================================================================
+    Engine engine;
+
+    ScopedPointer<Player> player;
+    ScopedPointer<Recorder> recorder;
+
+    //==============================================================================
+    static bool check (const SLresult result)
+    {
+        jassert (result == SL_RESULT_SUCCESS);
+        return result == SL_RESULT_SUCCESS;
+    }
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (OpenSLAudioIODevice)
+};
+
+
+//==============================================================================
+class OpenSLAudioDeviceType  : public AudioIODeviceType
+{
+public:
+    OpenSLAudioDeviceType()  : AudioIODeviceType (openSLTypeName) {}
+
+    //==============================================================================
+    void scanForDevices() {}
+    StringArray getDeviceNames (bool wantInputNames) const              { return StringArray (openSLTypeName); }
+    int getDefaultDeviceIndex (bool forInput) const                     { return 0; }
+    int getIndexOfDevice (AudioIODevice* device, bool asInput) const    { return device != nullptr ? 0 : -1; }
+    bool hasSeparateInputsAndOutputs() const                            { return false; }
+
+    AudioIODevice* createDevice (const String& outputDeviceName,
+                                 const String& inputDeviceName)
+    {
+        ScopedPointer<OpenSLAudioIODevice> dev;
+
+        if (outputDeviceName.isNotEmpty() || inputDeviceName.isNotEmpty())
+        {
+            dev = new OpenSLAudioIODevice (outputDeviceName.isNotEmpty() ? outputDeviceName
+                                                                         : inputDeviceName);
+            if (! dev->openedOk())
+                dev = nullptr;
+        }
+
+        return dev.release();
+    }
+
+private:
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (OpenSLAudioDeviceType)
+};
+
+
+//==============================================================================
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_OpenSLES()
+{
+    return isOpenSLAvailable() ? new OpenSLAudioDeviceType() : nullptr;
+}

source/modules/juce_audio_devices/native/juce_ios_Audio.cpp

@@ -0,0 +1,555 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+class iOSAudioIODevice  : public AudioIODevice
+{
+public:
+    iOSAudioIODevice (const String& deviceName)
+        : AudioIODevice (deviceName, "Audio"),
+          actualBufferSize (0),
+          isRunning (false),
+          audioUnit (0),
+          callback (nullptr),
+          floatData (1, 2)
+    {
+        getSessionHolder().activeDevices.add (this);
+
+        numInputChannels = 2;
+        numOutputChannels = 2;
+        preferredBufferSize = 0;
+
+        updateDeviceInfo();
+    }
+
+    ~iOSAudioIODevice()
+    {
+        getSessionHolder().activeDevices.removeFirstMatchingValue (this);
+        close();
+    }
+
+    StringArray getOutputChannelNames()
+    {
+        StringArray s;
+        s.add ("Left");
+        s.add ("Right");
+        return s;
+    }
+
+    StringArray getInputChannelNames()
+    {
+        StringArray s;
+        if (audioInputIsAvailable)
+        {
+            s.add ("Left");
+            s.add ("Right");
+        }
+        return s;
+    }
+
+    int getNumSampleRates()                 { return 1; }
+    double getSampleRate (int index)        { return sampleRate; }
+
+    int getNumBufferSizesAvailable()        { return 6; }
+    int getBufferSizeSamples (int index)    { return 1 << (jlimit (0, 5, index) + 6); }
+    int getDefaultBufferSize()              { return 1024; }
+
+    String open (const BigInteger& inputChannels,
+                 const BigInteger& outputChannels,
+                 double sampleRate,
+                 int bufferSize)
+    {
+        close();
+
+        lastError = String::empty;
+        preferredBufferSize = (bufferSize <= 0) ? getDefaultBufferSize() : bufferSize;
+
+        //  xxx set up channel mapping
+
+        activeOutputChans = outputChannels;
+        activeOutputChans.setRange (2, activeOutputChans.getHighestBit(), false);
+        numOutputChannels = activeOutputChans.countNumberOfSetBits();
+        monoOutputChannelNumber = activeOutputChans.findNextSetBit (0);
+
+        activeInputChans = inputChannels;
+        activeInputChans.setRange (2, activeInputChans.getHighestBit(), false);
+        numInputChannels = activeInputChans.countNumberOfSetBits();
+        monoInputChannelNumber = activeInputChans.findNextSetBit (0);
+
+        AudioSessionSetActive (true);
+
+        UInt32 audioCategory = (numInputChannels > 0 && audioInputIsAvailable) ? kAudioSessionCategory_PlayAndRecord
+                                                                               : kAudioSessionCategory_MediaPlayback;
+
+        AudioSessionSetProperty (kAudioSessionProperty_AudioCategory, sizeof (audioCategory), &audioCategory);
+
+        if (audioCategory == kAudioSessionCategory_PlayAndRecord)
+        {
+            // (note: mustn't set this until after the audio category property has been set)
+            UInt32 allowBluetoothInput = 1;
+            AudioSessionSetProperty (kAudioSessionProperty_OverrideCategoryEnableBluetoothInput,
+                                     sizeof (allowBluetoothInput), &allowBluetoothInput);
+        }
+
+        AudioSessionAddPropertyListener (kAudioSessionProperty_AudioRouteChange, routingChangedStatic, this);
+
+        fixAudioRouteIfSetToReceiver();
+        updateDeviceInfo();
+
+        Float32 bufferDuration = preferredBufferSize / sampleRate;
+        AudioSessionSetProperty (kAudioSessionProperty_PreferredHardwareIOBufferDuration, sizeof (bufferDuration), &bufferDuration);
+        actualBufferSize = preferredBufferSize;
+
+        prepareFloatBuffers (actualBufferSize);
+
+        isRunning = true;
+        routingChanged (nullptr);  // creates and starts the AU
+
+        lastError = audioUnit != 0 ? "" : "Couldn't open the device";
+        return lastError;
+    }
+
+    void close()
+    {
+        if (isRunning)
+        {
+            isRunning = false;
+            AudioSessionRemovePropertyListenerWithUserData (kAudioSessionProperty_AudioRouteChange, routingChangedStatic, this);
+            AudioSessionSetActive (false);
+
+            if (audioUnit != 0)
+            {
+                AudioComponentInstanceDispose (audioUnit);
+                audioUnit = 0;
+            }
+        }
+    }
+
+    bool isOpen()                       { return isRunning; }
+
+    int getCurrentBufferSizeSamples()   { return actualBufferSize; }
+    double getCurrentSampleRate()       { return sampleRate; }
+    int getCurrentBitDepth()            { return 16; }
+
+    BigInteger getActiveOutputChannels() const    { return activeOutputChans; }
+    BigInteger getActiveInputChannels() const     { return activeInputChans; }
+
+    int getOutputLatencyInSamples()               { return 0; } //xxx
+    int getInputLatencyInSamples()                { return 0; } //xxx
+
+    void start (AudioIODeviceCallback* newCallback)
+    {
+        if (isRunning && callback != newCallback)
+        {
+            if (newCallback != nullptr)
+                newCallback->audioDeviceAboutToStart (this);
+
+            const ScopedLock sl (callbackLock);
+            callback = newCallback;
+        }
+    }
+
+    void stop()
+    {
+        if (isRunning)
+        {
+            AudioIODeviceCallback* lastCallback;
+
+            {
+                const ScopedLock sl (callbackLock);
+                lastCallback = callback;
+                callback = nullptr;
+            }
+
+            if (lastCallback != nullptr)
+                lastCallback->audioDeviceStopped();
+        }
+    }
+
+    bool isPlaying()            { return isRunning && callback != nullptr; }
+    String getLastError()       { return lastError; }
+
+private:
+    //==================================================================================================
+    CriticalSection callbackLock;
+    Float64 sampleRate;
+    int numInputChannels, numOutputChannels;
+    int preferredBufferSize, actualBufferSize;
+    bool isRunning;
+    String lastError;
+
+    AudioStreamBasicDescription format;
+    AudioUnit audioUnit;
+    UInt32 audioInputIsAvailable;
+    AudioIODeviceCallback* callback;
+    BigInteger activeOutputChans, activeInputChans;
+
+    AudioSampleBuffer floatData;
+    float* inputChannels[3];
+    float* outputChannels[3];
+    bool monoInputChannelNumber, monoOutputChannelNumber;
+
+    void prepareFloatBuffers (int bufferSize)
+    {
+        if (numInputChannels + numOutputChannels > 0)
+        {
+            floatData.setSize (numInputChannels + numOutputChannels, bufferSize);
+            zeromem (inputChannels, sizeof (inputChannels));
+            zeromem (outputChannels, sizeof (outputChannels));
+
+            for (int i = 0; i < numInputChannels; ++i)
+                inputChannels[i] = floatData.getSampleData (i);
+
+            for (int i = 0; i < numOutputChannels; ++i)
+                outputChannels[i] = floatData.getSampleData (i + numInputChannels);
+        }
+    }
+
+    //==================================================================================================
+    OSStatus process (AudioUnitRenderActionFlags* flags, const AudioTimeStamp* time,
+                      const UInt32 numFrames, AudioBufferList* data)
+    {
+        OSStatus err = noErr;
+
+        if (audioInputIsAvailable && numInputChannels > 0)
+            err = AudioUnitRender (audioUnit, flags, time, 1, numFrames, data);
+
+        const ScopedLock sl (callbackLock);
+
+        if (callback != nullptr)
+        {
+            // This shouldn't ever get triggered, but please let me know if it does!
+            jassert (numFrames <= floatData.getNumSamples());
+
+            if (audioInputIsAvailable && numInputChannels > 0)
+            {
+                short* shortData = (short*) data->mBuffers[0].mData;
+
+                if (numInputChannels >= 2)
+                {
+                    for (UInt32 i = 0; i < numFrames; ++i)
+                    {
+                        inputChannels[0][i] = *shortData++ * (1.0f / 32768.0f);
+                        inputChannels[1][i] = *shortData++ * (1.0f / 32768.0f);
+                    }
+                }
+                else
+                {
+                    if (monoInputChannelNumber > 0)
+                        ++shortData;
+
+                    for (UInt32 i = 0; i < numFrames; ++i)
+                    {
+                        inputChannels[0][i] = *shortData++ * (1.0f / 32768.0f);
+                        ++shortData;
+                    }
+                }
+            }
+            else
+            {
+                for (int i = numInputChannels; --i >= 0;)
+                    zeromem (inputChannels[i], sizeof (float) * numFrames);
+            }
+
+            callback->audioDeviceIOCallback ((const float**) inputChannels, numInputChannels,
+                                             outputChannels, numOutputChannels, (int) numFrames);
+
+            short* shortData = (short*) data->mBuffers[0].mData;
+            int n = 0;
+
+            if (numOutputChannels >= 2)
+            {
+                for (UInt32 i = 0; i < numFrames; ++i)
+                {
+                    shortData [n++] = (short) (outputChannels[0][i] * 32767.0f);
+                    shortData [n++] = (short) (outputChannels[1][i] * 32767.0f);
+                }
+            }
+            else if (numOutputChannels == 1)
+            {
+                for (UInt32 i = 0; i < numFrames; ++i)
+                {
+                    const short s = (short) (outputChannels[monoOutputChannelNumber][i] * 32767.0f);
+                    shortData [n++] = s;
+                    shortData [n++] = s;
+                }
+            }
+            else
+            {
+                zeromem (data->mBuffers[0].mData, 2 * sizeof (short) * numFrames);
+            }
+        }
+        else
+        {
+            zeromem (data->mBuffers[0].mData, 2 * sizeof (short) * numFrames);
+        }
+
+        return err;
+    }
+
+    void updateDeviceInfo()
+    {
+        UInt32 size = sizeof (sampleRate);
+        AudioSessionGetProperty (kAudioSessionProperty_CurrentHardwareSampleRate, &size, &sampleRate);
+
+        size = sizeof (audioInputIsAvailable);
+        AudioSessionGetProperty (kAudioSessionProperty_AudioInputAvailable, &size, &audioInputIsAvailable);
+    }
+
+    void routingChanged (const void* propertyValue)
+    {
+        if (! isRunning)
+            return;
+
+        if (propertyValue != nullptr)
+        {
+            CFDictionaryRef routeChangeDictionary = (CFDictionaryRef) propertyValue;
+            CFNumberRef routeChangeReasonRef = (CFNumberRef) CFDictionaryGetValue (routeChangeDictionary,
+                                                                                   CFSTR (kAudioSession_AudioRouteChangeKey_Reason));
+
+            SInt32 routeChangeReason;
+            CFNumberGetValue (routeChangeReasonRef, kCFNumberSInt32Type, &routeChangeReason);
+
+            if (routeChangeReason == kAudioSessionRouteChangeReason_OldDeviceUnavailable)
+            {
+                const ScopedLock sl (callbackLock);
+
+                if (callback != nullptr)
+                    callback->audioDeviceError ("Old device unavailable");
+            }
+        }
+
+        updateDeviceInfo();
+        createAudioUnit();
+
+        AudioSessionSetActive (true);
+
+        if (audioUnit != 0)
+        {
+            UInt32 formatSize = sizeof (format);
+            AudioUnitGetProperty (audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &format, &formatSize);
+
+            Float32 bufferDuration = preferredBufferSize / sampleRate;
+            UInt32 bufferDurationSize = sizeof (bufferDuration);
+            AudioSessionGetProperty (kAudioSessionProperty_CurrentHardwareIOBufferDuration, &bufferDurationSize, &bufferDuration);
+            actualBufferSize = (int) (sampleRate * bufferDuration + 0.5);
+
+            AudioOutputUnitStart (audioUnit);
+        }
+    }
+
+    //==================================================================================================
+    struct AudioSessionHolder
+    {
+        AudioSessionHolder()
+        {
+            AudioSessionInitialize (0, 0, interruptionListenerCallback, this);
+        }
+
+        static void interruptionListenerCallback (void* client, UInt32 interruptionType)
+        {
+            const Array <iOSAudioIODevice*>& activeDevices = static_cast <AudioSessionHolder*> (client)->activeDevices;
+
+            for (int i = activeDevices.size(); --i >= 0;)
+                activeDevices.getUnchecked(i)->interruptionListener (interruptionType);
+        }
+
+        Array <iOSAudioIODevice*> activeDevices;
+    };
+
+    static AudioSessionHolder& getSessionHolder()
+    {
+        static AudioSessionHolder audioSessionHolder;
+        return audioSessionHolder;
+    }
+
+    void interruptionListener (const UInt32 interruptionType)
+    {
+        if (interruptionType == kAudioSessionBeginInterruption)
+        {
+            isRunning = false;
+            AudioOutputUnitStop (audioUnit);
+            AudioSessionSetActive (false);
+
+            const ScopedLock sl (callbackLock);
+
+            if (callback != nullptr)
+                callback->audioDeviceError ("iOS audio session interruption");
+        }
+
+        if (interruptionType == kAudioSessionEndInterruption)
+        {
+            isRunning = true;
+            AudioSessionSetActive (true);
+            AudioOutputUnitStart (audioUnit);
+        }
+    }
+
+    //==================================================================================================
+    static OSStatus processStatic (void* client, AudioUnitRenderActionFlags* flags, const AudioTimeStamp* time,
+                                   UInt32 /*busNumber*/, UInt32 numFrames, AudioBufferList* data)
+    {
+        return static_cast <iOSAudioIODevice*> (client)->process (flags, time, numFrames, data);
+    }
+
+    static void routingChangedStatic (void* client, AudioSessionPropertyID, UInt32 /*inDataSize*/, const void* propertyValue)
+    {
+        static_cast <iOSAudioIODevice*> (client)->routingChanged (propertyValue);
+    }
+
+    //==================================================================================================
+    void resetFormat (const int numChannels) noexcept
+    {
+        zerostruct (format);
+        format.mFormatID = kAudioFormatLinearPCM;
+        format.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked | kAudioFormatFlagsNativeEndian;
+        format.mBitsPerChannel = 8 * sizeof (short);
+        format.mChannelsPerFrame = numChannels;
+        format.mFramesPerPacket = 1;
+        format.mBytesPerFrame = format.mBytesPerPacket = numChannels * sizeof (short);
+    }
+
+    bool createAudioUnit()
+    {
+        if (audioUnit != 0)
+        {
+            AudioComponentInstanceDispose (audioUnit);
+            audioUnit = 0;
+        }
+
+        resetFormat (2);
+
+        AudioComponentDescription desc;
+        desc.componentType = kAudioUnitType_Output;
+        desc.componentSubType = kAudioUnitSubType_RemoteIO;
+        desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+        desc.componentFlags = 0;
+        desc.componentFlagsMask = 0;
+
+        AudioComponent comp = AudioComponentFindNext (0, &desc);
+        AudioComponentInstanceNew (comp, &audioUnit);
+
+        if (audioUnit == 0)
+            return false;
+
+        if (numInputChannels > 0)
+        {
+            const UInt32 one = 1;
+            AudioUnitSetProperty (audioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &one, sizeof (one));
+        }
+
+        {
+            AudioChannelLayout layout;
+            layout.mChannelBitmap = 0;
+            layout.mNumberChannelDescriptions = 0;
+            layout.mChannelLayoutTag = kAudioChannelLayoutTag_Stereo;
+            AudioUnitSetProperty (audioUnit, kAudioUnitProperty_AudioChannelLayout, kAudioUnitScope_Input,  0, &layout, sizeof (layout));
+            AudioUnitSetProperty (audioUnit, kAudioUnitProperty_AudioChannelLayout, kAudioUnitScope_Output, 0, &layout, sizeof (layout));
+        }
+
+        {
+            AURenderCallbackStruct inputProc;
+            inputProc.inputProc = processStatic;
+            inputProc.inputProcRefCon = this;
+            AudioUnitSetProperty (audioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &inputProc, sizeof (inputProc));
+        }
+
+        AudioUnitSetProperty (audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input,  0, &format, sizeof (format));
+        AudioUnitSetProperty (audioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &format, sizeof (format));
+
+        AudioUnitInitialize (audioUnit);
+        return true;
+    }
+
+    // If the routing is set to go through the receiver (i.e. the speaker, but quiet), this re-routes it
+    // to make it loud. Needed because by default when using an input + output, the output is kept quiet.
+    static void fixAudioRouteIfSetToReceiver()
+    {
+        CFStringRef audioRoute = 0;
+        UInt32 propertySize = sizeof (audioRoute);
+        if (AudioSessionGetProperty (kAudioSessionProperty_AudioRoute, &propertySize, &audioRoute) == noErr)
+        {
+            NSString* route = (NSString*) audioRoute;
+
+            //DBG ("audio route: " + nsStringToJuce (route));
+
+            if ([route hasPrefix: @"Receiver"])
+            {
+                UInt32 audioRouteOverride = kAudioSessionOverrideAudioRoute_Speaker;
+                AudioSessionSetProperty (kAudioSessionProperty_OverrideAudioRoute, sizeof (audioRouteOverride), &audioRouteOverride);
+            }
+
+            CFRelease (audioRoute);
+        }
+    }
+
+    JUCE_DECLARE_NON_COPYABLE (iOSAudioIODevice)
+};
+
+
+//==============================================================================
+class iOSAudioIODeviceType  : public AudioIODeviceType
+{
+public:
+    iOSAudioIODeviceType()  : AudioIODeviceType ("iOS Audio")
+    {
+    }
+
+    void scanForDevices() {}
+
+    StringArray getDeviceNames (bool wantInputNames) const
+    {
+        return StringArray ("iOS Audio");
+    }
+
+    int getDefaultDeviceIndex (bool forInput) const
+    {
+        return 0;
+    }
+
+    int getIndexOfDevice (AudioIODevice* device, bool asInput) const
+    {
+        return device != nullptr ? 0 : -1;
+    }
+
+    bool hasSeparateInputsAndOutputs() const    { return false; }
+
+    AudioIODevice* createDevice (const String& outputDeviceName,
+                                 const String& inputDeviceName)
+    {
+        if (outputDeviceName.isNotEmpty() || inputDeviceName.isNotEmpty())
+            return new iOSAudioIODevice (outputDeviceName.isNotEmpty() ? outputDeviceName
+                                                                       : inputDeviceName);
+
+        return nullptr;
+    }
+
+private:
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (iOSAudioIODeviceType)
+};
+
+//==============================================================================
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_iOSAudio()
+{
+    return new iOSAudioIODeviceType();
+}

source/modules/juce_audio_devices/native/juce_linux_AudioCDReader.cpp

@@ -0,0 +1,77 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+AudioCDReader::AudioCDReader()
+    : AudioFormatReader (0, "CD Audio")
+{
+}
+
+StringArray AudioCDReader::getAvailableCDNames()
+{
+    StringArray names;
+    return names;
+}
+
+AudioCDReader* AudioCDReader::createReaderForCD (const int index)
+{
+    return nullptr;
+}
+
+AudioCDReader::~AudioCDReader()
+{
+}
+
+void AudioCDReader::refreshTrackLengths()
+{
+}
+
+bool AudioCDReader::readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                                 int64 startSampleInFile, int numSamples)
+{
+    return false;
+}
+
+bool AudioCDReader::isCDStillPresent() const
+{
+    return false;
+}
+
+bool AudioCDReader::isTrackAudio (int trackNum) const
+{
+    return false;
+}
+
+void AudioCDReader::enableIndexScanning (bool b)
+{
+}
+
+int AudioCDReader::getLastIndex() const
+{
+    return 0;
+}
+
+Array<int> AudioCDReader::findIndexesInTrack (const int trackNumber)
+{
+    return Array<int>();
+}

source/modules/juce_audio_devices/native/juce_linux_JackAudio.cpp

@@ -0,0 +1,590 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+//==============================================================================
+static void* juce_libjackHandle = nullptr;
+
+static void* juce_loadJackFunction (const char* const name)
+{
+    if (juce_libjackHandle == nullptr)
+        return nullptr;
+
+    return dlsym (juce_libjackHandle, name);
+}
+
+#define JUCE_DECL_JACK_FUNCTION(return_type, fn_name, argument_types, arguments)  \
+  return_type fn_name argument_types                                              \
+  {                                                                               \
+      typedef return_type (*fn_type) argument_types;                              \
+      static fn_type fn = (fn_type) juce_loadJackFunction (#fn_name);             \
+      return (fn != nullptr) ? ((*fn) arguments) : (return_type) 0;               \
+  }
+
+#define JUCE_DECL_VOID_JACK_FUNCTION(fn_name, argument_types, arguments)          \
+  void fn_name argument_types                                                     \
+  {                                                                               \
+      typedef void (*fn_type) argument_types;                                     \
+      static fn_type fn = (fn_type) juce_loadJackFunction (#fn_name);             \
+      if (fn != nullptr) (*fn) arguments;                                         \
+  }
+
+//==============================================================================
+JUCE_DECL_JACK_FUNCTION (jack_client_t*, jack_client_open, (const char* client_name, jack_options_t options, jack_status_t* status, ...), (client_name, options, status));
+JUCE_DECL_JACK_FUNCTION (int, jack_client_close, (jack_client_t *client), (client));
+JUCE_DECL_JACK_FUNCTION (int, jack_activate, (jack_client_t* client), (client));
+JUCE_DECL_JACK_FUNCTION (int, jack_deactivate, (jack_client_t* client), (client));
+JUCE_DECL_JACK_FUNCTION (jack_nframes_t, jack_get_buffer_size, (jack_client_t* client), (client));
+JUCE_DECL_JACK_FUNCTION (jack_nframes_t, jack_get_sample_rate, (jack_client_t* client), (client));
+JUCE_DECL_VOID_JACK_FUNCTION (jack_on_shutdown, (jack_client_t* client, void (*function)(void* arg), void* arg), (client, function, arg));
+JUCE_DECL_JACK_FUNCTION (void* , jack_port_get_buffer, (jack_port_t* port, jack_nframes_t nframes), (port, nframes));
+JUCE_DECL_JACK_FUNCTION (jack_nframes_t, jack_port_get_total_latency, (jack_client_t* client, jack_port_t* port), (client, port));
+JUCE_DECL_JACK_FUNCTION (jack_port_t* , jack_port_register, (jack_client_t* client, const char* port_name, const char* port_type, unsigned long flags, unsigned long buffer_size), (client, port_name, port_type, flags, buffer_size));
+JUCE_DECL_VOID_JACK_FUNCTION (jack_set_error_function, (void (*func)(const char*)), (func));
+JUCE_DECL_JACK_FUNCTION (int, jack_set_process_callback, (jack_client_t* client, JackProcessCallback process_callback, void* arg), (client, process_callback, arg));
+JUCE_DECL_JACK_FUNCTION (const char**, jack_get_ports, (jack_client_t* client, const char* port_name_pattern, const char* type_name_pattern, unsigned long flags), (client, port_name_pattern, type_name_pattern, flags));
+JUCE_DECL_JACK_FUNCTION (int, jack_connect, (jack_client_t* client, const char* source_port, const char* destination_port), (client, source_port, destination_port));
+JUCE_DECL_JACK_FUNCTION (const char*, jack_port_name, (const jack_port_t* port), (port));
+JUCE_DECL_JACK_FUNCTION (void*, jack_set_port_connect_callback, (jack_client_t* client, JackPortConnectCallback connect_callback, void* arg), (client, connect_callback, arg));
+JUCE_DECL_JACK_FUNCTION (jack_port_t* , jack_port_by_id, (jack_client_t* client, jack_port_id_t port_id), (client, port_id));
+JUCE_DECL_JACK_FUNCTION (int, jack_port_connected, (const jack_port_t* port), (port));
+JUCE_DECL_JACK_FUNCTION (int, jack_port_connected_to, (const jack_port_t* port, const char* port_name), (port, port_name));
+
+#if JUCE_DEBUG
+ #define JACK_LOGGING_ENABLED 1
+#endif
+
+#if JACK_LOGGING_ENABLED
+namespace
+{
+    void jack_Log (const String& s)
+    {
+        std::cerr << s << std::endl;
+    }
+
+    const char* getJackErrorMessage (const jack_status_t status)
+    {
+        if (status & JackServerFailed
+             || status & JackServerError)   return "Unable to connect to JACK server";
+        if (status & JackVersionError)      return "Client's protocol version does not match";
+        if (status & JackInvalidOption)     return "The operation contained an invalid or unsupported option";
+        if (status & JackNameNotUnique)     return "The desired client name was not unique";
+        if (status & JackNoSuchClient)      return "Requested client does not exist";
+        if (status & JackInitFailure)       return "Unable to initialize client";
+        return nullptr;
+    }
+}
+ #define JUCE_JACK_LOG_STATUS(x)    { if (const char* m = getJackErrorMessage (x)) jack_Log (m); }
+ #define JUCE_JACK_LOG(x)           jack_Log(x)
+#else
+ #define JUCE_JACK_LOG_STATUS(x)    {}
+ #define JUCE_JACK_LOG(x)           {}
+#endif
+
+
+//==============================================================================
+#ifndef JUCE_JACK_CLIENT_NAME
+ #define JUCE_JACK_CLIENT_NAME "JUCEJack"
+#endif
+
+struct JackPortIterator
+{
+    JackPortIterator (jack_client_t* const client, const bool forInput)
+        : ports (nullptr), index (-1)
+    {
+        if (client != nullptr)
+            ports = juce::jack_get_ports (client, nullptr, nullptr,
+                                          forInput ? JackPortIsOutput : JackPortIsInput);
+                                            // (NB: This looks like it's the wrong way round, but it is correct!)
+    }
+
+    ~JackPortIterator()
+    {
+        ::free (ports);
+    }
+
+    bool next()
+    {
+        if (ports == nullptr || ports [index + 1] == nullptr)
+            return false;
+
+        name = CharPointer_UTF8 (ports[++index]);
+        clientName = name.upToFirstOccurrenceOf (":", false, false);
+        return true;
+    }
+
+    const char** ports;
+    int index;
+    String name;
+    String clientName;
+};
+
+class JackAudioIODeviceType;
+static Array<JackAudioIODeviceType*> activeDeviceTypes;
+
+//==============================================================================
+class JackAudioIODevice   : public AudioIODevice
+{
+public:
+    JackAudioIODevice (const String& deviceName,
+                       const String& inId,
+                       const String& outId)
+        : AudioIODevice (deviceName, "JACK"),
+          inputId (inId),
+          outputId (outId),
+          deviceIsOpen (false),
+          callback (nullptr),
+          totalNumberOfInputChannels (0),
+          totalNumberOfOutputChannels (0)
+    {
+        jassert (deviceName.isNotEmpty());
+
+        jack_status_t status;
+        client = juce::jack_client_open (JUCE_JACK_CLIENT_NAME, JackNoStartServer, &status);
+
+        if (client == nullptr)
+        {
+            JUCE_JACK_LOG_STATUS (status);
+        }
+        else
+        {
+            juce::jack_set_error_function (errorCallback);
+
+            // open input ports
+            const StringArray inputChannels (getInputChannelNames());
+            for (int i = 0; i < inputChannels.size(); ++i)
+            {
+                String inputName;
+                inputName << "in_" << ++totalNumberOfInputChannels;
+
+                inputPorts.add (juce::jack_port_register (client, inputName.toUTF8(),
+                                                          JACK_DEFAULT_AUDIO_TYPE, JackPortIsInput, 0));
+            }
+
+            // open output ports
+            const StringArray outputChannels (getOutputChannelNames());
+            for (int i = 0; i < outputChannels.size (); ++i)
+            {
+                String outputName;
+                outputName << "out_" << ++totalNumberOfOutputChannels;
+
+                outputPorts.add (juce::jack_port_register (client, outputName.toUTF8(),
+                                                           JACK_DEFAULT_AUDIO_TYPE, JackPortIsOutput, 0));
+            }
+
+            inChans.calloc (totalNumberOfInputChannels + 2);
+            outChans.calloc (totalNumberOfOutputChannels + 2);
+        }
+    }
+
+    ~JackAudioIODevice()
+    {
+        close();
+        if (client != nullptr)
+        {
+            juce::jack_client_close (client);
+            client = nullptr;
+        }
+    }
+
+    StringArray getChannelNames (bool forInput) const
+    {
+        StringArray names;
+
+        for (JackPortIterator i (client, forInput); i.next();)
+            if (i.clientName == getName())
+                names.add (i.name.fromFirstOccurrenceOf (":", false, false));
+
+        return names;
+    }
+
+    StringArray getOutputChannelNames()         { return getChannelNames (false); }
+    StringArray getInputChannelNames()          { return getChannelNames (true); }
+    int getNumSampleRates()                     { return client != nullptr ? 1 : 0; }
+    double getSampleRate (int /*index*/)        { return client != nullptr ? juce::jack_get_sample_rate (client) : 0; }
+    int getNumBufferSizesAvailable()            { return client != nullptr ? 1 : 0; }
+    int getBufferSizeSamples (int /*index*/)    { return getDefaultBufferSize(); }
+    int getDefaultBufferSize()                  { return client != nullptr ? juce::jack_get_buffer_size (client) : 0; }
+
+    String open (const BigInteger& inputChannels, const BigInteger& outputChannels,
+                 double /* sampleRate */, int /* bufferSizeSamples */)
+    {
+        if (client == nullptr)
+        {
+            lastError = "No JACK client running";
+            return lastError;
+        }
+
+        lastError = String::empty;
+        close();
+
+        juce::jack_set_process_callback (client, processCallback, this);
+        juce::jack_set_port_connect_callback (client, portConnectCallback, this);
+        juce::jack_on_shutdown (client, shutdownCallback, this);
+        juce::jack_activate (client);
+        deviceIsOpen = true;
+
+        if (! inputChannels.isZero())
+        {
+            for (JackPortIterator i (client, true); i.next();)
+            {
+                if (inputChannels [i.index] && i.clientName == getName())
+                {
+                    int error = juce::jack_connect (client, i.ports[i.index], juce::jack_port_name ((jack_port_t*) inputPorts[i.index]));
+                    if (error != 0)
+                        JUCE_JACK_LOG ("Cannot connect input port " + String (i.index) + " (" + i.name + "), error " + String (error));
+                }
+            }
+        }
+
+        if (! outputChannels.isZero())
+        {
+            for (JackPortIterator i (client, false); i.next();)
+            {
+                if (outputChannels [i.index] && i.clientName == getName())
+                {
+                    int error = juce::jack_connect (client, juce::jack_port_name ((jack_port_t*) outputPorts[i.index]), i.ports[i.index]);
+                    if (error != 0)
+                        JUCE_JACK_LOG ("Cannot connect output port " + String (i.index) + " (" + i.name + "), error " + String (error));
+                }
+            }
+        }
+
+        return lastError;
+    }
+
+    void close()
+    {
+        stop();
+
+        if (client != nullptr)
+        {
+            juce::jack_deactivate (client);
+            juce::jack_set_process_callback (client, processCallback, nullptr);
+            juce::jack_set_port_connect_callback (client, portConnectCallback, nullptr);
+            juce::jack_on_shutdown (client, shutdownCallback, nullptr);
+        }
+
+        deviceIsOpen = false;
+    }
+
+    void start (AudioIODeviceCallback* newCallback)
+    {
+        if (deviceIsOpen && newCallback != callback)
+        {
+            if (newCallback != nullptr)
+                newCallback->audioDeviceAboutToStart (this);
+
+            AudioIODeviceCallback* const oldCallback = callback;
+
+            {
+                const ScopedLock sl (callbackLock);
+                callback = newCallback;
+            }
+
+            if (oldCallback != nullptr)
+                oldCallback->audioDeviceStopped();
+        }
+    }
+
+    void stop()
+    {
+        start (nullptr);
+    }
+
+    bool isOpen()                           { return deviceIsOpen; }
+    bool isPlaying()                        { return callback != nullptr; }
+    int getCurrentBufferSizeSamples()       { return getBufferSizeSamples (0); }
+    double getCurrentSampleRate()           { return getSampleRate (0); }
+    int getCurrentBitDepth()                { return 32; }
+    String getLastError()                   { return lastError; }
+
+    BigInteger getActiveOutputChannels() const { return activeOutputChannels; }
+    BigInteger getActiveInputChannels()  const { return activeInputChannels;  }
+
+    int getOutputLatencyInSamples()
+    {
+        int latency = 0;
+
+        for (int i = 0; i < outputPorts.size(); i++)
+            latency = jmax (latency, (int) juce::jack_port_get_total_latency (client, (jack_port_t*) outputPorts [i]));
+
+        return latency;
+    }
+
+    int getInputLatencyInSamples()
+    {
+        int latency = 0;
+
+        for (int i = 0; i < inputPorts.size(); i++)
+            latency = jmax (latency, (int) juce::jack_port_get_total_latency (client, (jack_port_t*) inputPorts [i]));
+
+        return latency;
+    }
+
+    String inputId, outputId;
+
+private:
+    void process (const int numSamples)
+    {
+        int numActiveInChans = 0, numActiveOutChans = 0;
+
+        for (int i = 0; i < totalNumberOfInputChannels; ++i)
+        {
+            if (activeInputChannels[i])
+                if (jack_default_audio_sample_t* in
+                        = (jack_default_audio_sample_t*) juce::jack_port_get_buffer ((jack_port_t*) inputPorts.getUnchecked(i), numSamples))
+                    inChans [numActiveInChans++] = (float*) in;
+        }
+
+        for (int i = 0; i < totalNumberOfOutputChannels; ++i)
+        {
+            if (activeOutputChannels[i])
+                if (jack_default_audio_sample_t* out
+                        = (jack_default_audio_sample_t*) juce::jack_port_get_buffer ((jack_port_t*) outputPorts.getUnchecked(i), numSamples))
+                    outChans [numActiveOutChans++] = (float*) out;
+        }
+
+        const ScopedLock sl (callbackLock);
+
+        if (callback != nullptr)
+        {
+            if ((numActiveInChans + numActiveOutChans) > 0)
+                callback->audioDeviceIOCallback (const_cast <const float**> (inChans.getData()), numActiveInChans,
+                                                 outChans, numActiveOutChans, numSamples);
+        }
+        else
+        {
+            for (int i = 0; i < numActiveOutChans; ++i)
+                zeromem (outChans[i], sizeof (float) * numSamples);
+        }
+    }
+
+    static int processCallback (jack_nframes_t nframes, void* callbackArgument)
+    {
+        if (callbackArgument != nullptr)
+            ((JackAudioIODevice*) callbackArgument)->process (nframes);
+
+        return 0;
+    }
+
+    void updateActivePorts()
+    {
+        BigInteger newOutputChannels, newInputChannels;
+
+        for (int i = 0; i < outputPorts.size(); ++i)
+            if (juce::jack_port_connected ((jack_port_t*) outputPorts.getUnchecked(i)))
+                newOutputChannels.setBit (i);
+
+        for (int i = 0; i < inputPorts.size(); ++i)
+            if (juce::jack_port_connected ((jack_port_t*) inputPorts.getUnchecked(i)))
+                newInputChannels.setBit (i);
+
+        if (newOutputChannels != activeOutputChannels
+             || newInputChannels != activeInputChannels)
+        {
+            AudioIODeviceCallback* const oldCallback = callback;
+
+            stop();
+
+            activeOutputChannels = newOutputChannels;
+            activeInputChannels  = newInputChannels;
+
+            if (oldCallback != nullptr)
+                start (oldCallback);
+
+            sendDeviceChangedCallback();
+        }
+    }
+
+    static void portConnectCallback (jack_port_id_t, jack_port_id_t, int, void* arg)
+    {
+        if (JackAudioIODevice* device = static_cast <JackAudioIODevice*> (arg))
+            device->updateActivePorts();
+    }
+
+    static void threadInitCallback (void* /* callbackArgument */)
+    {
+        JUCE_JACK_LOG ("JackAudioIODevice::initialise");
+    }
+
+    static void shutdownCallback (void* callbackArgument)
+    {
+        JUCE_JACK_LOG ("JackAudioIODevice::shutdown");
+
+        if (JackAudioIODevice* device = (JackAudioIODevice*) callbackArgument)
+        {
+            device->client = nullptr;
+            device->close();
+        }
+    }
+
+    static void errorCallback (const char* msg)
+    {
+        JUCE_JACK_LOG ("JackAudioIODevice::errorCallback " + String (msg));
+    }
+
+    static void sendDeviceChangedCallback();
+
+    bool deviceIsOpen;
+    jack_client_t* client;
+    String lastError;
+    AudioIODeviceCallback* callback;
+    CriticalSection callbackLock;
+
+    HeapBlock <float*> inChans, outChans;
+    int totalNumberOfInputChannels;
+    int totalNumberOfOutputChannels;
+    Array<void*> inputPorts, outputPorts;
+    BigInteger activeInputChannels, activeOutputChannels;
+};
+
+
+//==============================================================================
+class JackAudioIODeviceType  : public AudioIODeviceType
+{
+public:
+    JackAudioIODeviceType()
+        : AudioIODeviceType ("JACK"),
+          hasScanned (false)
+    {
+        activeDeviceTypes.add (this);
+    }
+
+    ~JackAudioIODeviceType()
+    {
+        activeDeviceTypes.removeFirstMatchingValue (this);
+    }
+
+    void scanForDevices()
+    {
+        hasScanned = true;
+        inputNames.clear();
+        inputIds.clear();
+        outputNames.clear();
+        outputIds.clear();
+
+        if (juce_libjackHandle == nullptr)
+        {
+            juce_libjackHandle = dlopen ("libjack.so", RTLD_LAZY);
+
+            if (juce_libjackHandle == nullptr)
+                return;
+        }
+
+        jack_status_t status;
+
+        // open a dummy client
+        if (jack_client_t* const client = juce::jack_client_open ("JuceJackDummy", JackNoStartServer, &status))
+        {
+            // scan for output devices
+            for (JackPortIterator i (client, false); i.next();)
+            {
+                if (i.clientName != (JUCE_JACK_CLIENT_NAME) && ! inputNames.contains (i.clientName))
+                {
+                    inputNames.add (i.clientName);
+                    inputIds.add (i.ports [i.index]);
+                }
+            }
+
+            // scan for input devices
+            for (JackPortIterator i (client, true); i.next();)
+            {
+                if (i.clientName != (JUCE_JACK_CLIENT_NAME) && ! outputNames.contains (i.clientName))
+                {
+                    outputNames.add (i.clientName);
+                    outputIds.add (i.ports [i.index]);
+                }
+            }
+
+            juce::jack_client_close (client);
+        }
+        else
+        {
+            JUCE_JACK_LOG_STATUS (status);
+        }
+    }
+
+    StringArray getDeviceNames (bool wantInputNames) const
+    {
+        jassert (hasScanned); // need to call scanForDevices() before doing this
+        return wantInputNames ? inputNames : outputNames;
+    }
+
+    int getDefaultDeviceIndex (bool /* forInput */) const
+    {
+        jassert (hasScanned); // need to call scanForDevices() before doing this
+        return 0;
+    }
+
+    bool hasSeparateInputsAndOutputs() const    { return true; }
+
+    int getIndexOfDevice (AudioIODevice* device, bool asInput) const
+    {
+        jassert (hasScanned); // need to call scanForDevices() before doing this
+
+        if (JackAudioIODevice* d = dynamic_cast <JackAudioIODevice*> (device))
+            return asInput ? inputIds.indexOf (d->inputId)
+                           : outputIds.indexOf (d->outputId);
+
+        return -1;
+    }
+
+    AudioIODevice* createDevice (const String& outputDeviceName,
+                                 const String& inputDeviceName)
+    {
+        jassert (hasScanned); // need to call scanForDevices() before doing this
+
+        const int inputIndex = inputNames.indexOf (inputDeviceName);
+        const int outputIndex = outputNames.indexOf (outputDeviceName);
+
+        if (inputIndex >= 0 || outputIndex >= 0)
+            return new JackAudioIODevice (outputIndex >= 0 ? outputDeviceName
+                                                           : inputDeviceName,
+                                          inputIds [inputIndex],
+                                          outputIds [outputIndex]);
+
+        return nullptr;
+    }
+
+    void portConnectionChange()    { callDeviceChangeListeners(); }
+
+private:
+    StringArray inputNames, outputNames, inputIds, outputIds;
+    bool hasScanned;
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (JackAudioIODeviceType)
+};
+
+void JackAudioIODevice::sendDeviceChangedCallback()
+{
+    for (int i = activeDeviceTypes.size(); --i >= 0;)
+        if (JackAudioIODeviceType* d = activeDeviceTypes[i])
+            d->portConnectionChange();
+}
+
+//==============================================================================
+AudioIODeviceType* AudioIODeviceType::createAudioIODeviceType_JACK()
+{
+    return new JackAudioIODeviceType();
+}

source/modules/juce_audio_devices/native/juce_linux_Midi.cpp

@@ -0,0 +1,610 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#if JUCE_ALSA
+
+// You can define these strings in your app if you want to override the default names:
+#ifndef JUCE_ALSA_MIDI_INPUT_NAME
+ #define JUCE_ALSA_MIDI_INPUT_NAME  "Juce Midi Input"
+#endif
+
+#ifndef JUCE_ALSA_MIDI_OUTPUT_NAME
+ #define JUCE_ALSA_MIDI_OUTPUT_NAME "Juce Midi Output"
+#endif
+
+//==============================================================================
+namespace
+{
+
+class AlsaPortAndCallback;
+
+//==============================================================================
+class AlsaClient : public ReferenceCountedObject
+{
+public:
+    typedef ReferenceCountedObjectPtr<AlsaClient> Ptr;
+
+    AlsaClient (bool forInput)
+        : input (forInput), handle (nullptr)
+    {
+        snd_seq_open (&handle, "default", forInput ? SND_SEQ_OPEN_INPUT
+                                                   : SND_SEQ_OPEN_OUTPUT, 0);
+    }
+
+    ~AlsaClient()
+    {
+        if (handle != nullptr)
+        {
+            snd_seq_close (handle);
+            handle = nullptr;
+        }
+
+        jassert (activeCallbacks.size() == 0);
+
+        if (inputThread)
+        {
+            inputThread->stopThread (3000);
+            inputThread = nullptr;
+        }
+    }
+
+    bool isInput() const noexcept    { return input; }
+
+    void setName (const String& name)
+    {
+        snd_seq_set_client_name (handle, name.toUTF8());
+    }
+
+    void registerCallback (AlsaPortAndCallback* cb)
+    {
+        if (cb != nullptr)
+        {
+            {
+                const ScopedLock sl (callbackLock);
+                activeCallbacks.add (cb);
+
+                if (inputThread == nullptr)
+                    inputThread = new MidiInputThread (*this);
+            }
+
+            inputThread->startThread();
+        }
+    }
+
+    void unregisterCallback (AlsaPortAndCallback* cb)
+    {
+        const ScopedLock sl (callbackLock);
+
+        jassert (activeCallbacks.contains (cb));
+        activeCallbacks.removeAllInstancesOf (cb);
+
+        if (activeCallbacks.size() == 0 && inputThread->isThreadRunning())
+            inputThread->signalThreadShouldExit();
+    }
+
+    void handleIncomingMidiMessage (const MidiMessage& message, int port);
+
+    snd_seq_t* get() const noexcept     { return handle; }
+
+private:
+    bool input;
+    snd_seq_t* handle;
+
+    Array<AlsaPortAndCallback*> activeCallbacks;
+    CriticalSection callbackLock;
+
+    //==============================================================================
+    class MidiInputThread   : public Thread
+    {
+    public:
+        MidiInputThread (AlsaClient& c)
+            : Thread ("Juce MIDI Input"), client (c)
+        {
+            jassert (client.input && client.get() != nullptr);
+        }
+
+        void run() override
+        {
+            const int maxEventSize = 16 * 1024;
+            snd_midi_event_t* midiParser;
+            snd_seq_t* seqHandle = client.get();
+
+            if (snd_midi_event_new (maxEventSize, &midiParser) >= 0)
+            {
+                const int numPfds = snd_seq_poll_descriptors_count (seqHandle, POLLIN);
+                HeapBlock<pollfd> pfd (numPfds);
+                snd_seq_poll_descriptors (seqHandle, pfd, numPfds, POLLIN);
+
+                HeapBlock <uint8> buffer (maxEventSize);
+
+                while (! threadShouldExit())
+                {
+                    if (poll (pfd, numPfds, 100) > 0) // there was a "500" here which is a bit long when we exit the program and have to wait for a timeout on this poll call
+                    {
+                        if (threadShouldExit())
+                            break;
+
+                        snd_seq_nonblock (seqHandle, 1);
+
+                        do
+                        {
+                            snd_seq_event_t* inputEvent = nullptr;
+
+                            if (snd_seq_event_input (seqHandle, &inputEvent) >= 0)
+                            {
+                                // xxx what about SYSEXes that are too big for the buffer?
+                                const int numBytes = snd_midi_event_decode (midiParser, buffer,
+                                                                            maxEventSize, inputEvent);
+
+                                snd_midi_event_reset_decode (midiParser);
+
+                                if (numBytes > 0)
+                                {
+                                    const MidiMessage message ((const uint8*) buffer, numBytes,
+                                                               Time::getMillisecondCounter() * 0.001);
+
+                                    client.handleIncomingMidiMessage (message, inputEvent->dest.port);
+                                }
+
+                                snd_seq_free_event (inputEvent);
+                            }
+                        }
+                        while (snd_seq_event_input_pending (seqHandle, 0) > 0);
+                    }
+                }
+
+                snd_midi_event_free (midiParser);
+            }
+        };
+
+    private:
+        AlsaClient& client;
+    };
+
+    ScopedPointer<MidiInputThread> inputThread;
+};
+
+
+static AlsaClient::Ptr globalAlsaSequencerIn()
+{
+    static AlsaClient::Ptr global (new AlsaClient (true));
+    return global;
+}
+
+static AlsaClient::Ptr globalAlsaSequencerOut()
+{
+    static AlsaClient::Ptr global (new AlsaClient (false));
+    return global;
+}
+
+static AlsaClient::Ptr globalAlsaSequencer (bool input)
+{
+    return input ? globalAlsaSequencerIn()
+                 : globalAlsaSequencerOut();
+}
+
+//==============================================================================
+// represents an input or output port of the supplied AlsaClient
+class AlsaPort
+{
+public:
+    AlsaPort() noexcept  : portId (-1)  {}
+    AlsaPort (const AlsaClient::Ptr& c, int port) noexcept  : client (c), portId (port) {}
+
+    void createPort (const AlsaClient::Ptr& c, const String& name, bool forInput)
+    {
+        client = c;
+
+        if (snd_seq_t* handle = client->get())
+            portId = snd_seq_create_simple_port (handle, name.toUTF8(),
+                                                 forInput ? (SND_SEQ_PORT_CAP_WRITE | SND_SEQ_PORT_CAP_SUBS_WRITE)
+                                                          : (SND_SEQ_PORT_CAP_READ  | SND_SEQ_PORT_CAP_SUBS_READ),
+                                                 SND_SEQ_PORT_TYPE_MIDI_GENERIC);
+    }
+
+    void deletePort()
+    {
+        if (isValid())
+        {
+            snd_seq_delete_simple_port (client->get(), portId);
+            portId = -1;
+        }
+    }
+
+    void connectWith (int sourceClient, int sourcePort)
+    {
+        if (client->isInput())
+            snd_seq_connect_from (client->get(), portId, sourceClient, sourcePort);
+        else
+            snd_seq_connect_to (client->get(), portId, sourceClient, sourcePort);
+    }
+
+    bool isValid() const noexcept
+    {
+        return client != nullptr && client->get() != nullptr && portId >= 0;
+    }
+
+    AlsaClient::Ptr client;
+    int portId;
+};
+
+//==============================================================================
+class AlsaPortAndCallback
+{
+public:
+    AlsaPortAndCallback (AlsaPort p, MidiInput* in, MidiInputCallback* cb)
+        : port (p), midiInput (in), callback (cb), callbackEnabled (false)
+    {
+    }
+
+    ~AlsaPortAndCallback()
+    {
+        enableCallback (false);
+        port.deletePort();
+    }
+
+    void enableCallback (bool enable)
+    {
+        if (callbackEnabled != enable)
+        {
+            callbackEnabled = enable;
+
+            if (enable)
+                port.client->registerCallback (this);
+            else
+                port.client->unregisterCallback (this);
+        }
+    }
+
+    void handleIncomingMidiMessage (const MidiMessage& message) const
+    {
+        callback->handleIncomingMidiMessage (midiInput, message);
+    }
+
+private:
+    AlsaPort port;
+    MidiInput* midiInput;
+    MidiInputCallback* callback;
+    bool callbackEnabled;
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AlsaPortAndCallback)
+};
+
+void AlsaClient::handleIncomingMidiMessage (const MidiMessage& message, int port)
+{
+    const ScopedLock sl (callbackLock);
+
+    if (AlsaPortAndCallback* const cb = activeCallbacks[port])
+        cb->handleIncomingMidiMessage (message);
+}
+
+//==============================================================================
+static AlsaPort iterateMidiClient (const AlsaClient::Ptr& seq,
+                                   snd_seq_client_info_t* clientInfo,
+                                   const bool forInput,
+                                   StringArray& deviceNamesFound,
+                                   const int deviceIndexToOpen)
+{
+    AlsaPort port;
+
+    snd_seq_t* seqHandle = seq->get();
+    snd_seq_port_info_t* portInfo = nullptr;
+
+    if (snd_seq_port_info_malloc (&portInfo) == 0)
+    {
+        int numPorts = snd_seq_client_info_get_num_ports (clientInfo);
+        const int client = snd_seq_client_info_get_client (clientInfo);
+
+        snd_seq_port_info_set_client (portInfo, client);
+        snd_seq_port_info_set_port (portInfo, -1);
+
+        while (--numPorts >= 0)
+        {
+            if (snd_seq_query_next_port (seqHandle, portInfo) == 0
+                && (snd_seq_port_info_get_capability (portInfo) & (forInput ? SND_SEQ_PORT_CAP_READ
+                                                                            : SND_SEQ_PORT_CAP_WRITE)) != 0)
+            {
+                deviceNamesFound.add (snd_seq_client_info_get_name (clientInfo));
+
+                if (deviceNamesFound.size() == deviceIndexToOpen + 1)
+                {
+                    const int sourcePort   = snd_seq_port_info_get_port (portInfo);
+                    const int sourceClient = snd_seq_client_info_get_client (clientInfo);
+
+                    if (sourcePort != -1)
+                    {
+                        const String name (forInput ? JUCE_ALSA_MIDI_INPUT_NAME
+                                                    : JUCE_ALSA_MIDI_OUTPUT_NAME);
+                        seq->setName (name);
+                        port.createPort (seq, name, forInput);
+                        port.connectWith (sourceClient, sourcePort);
+                    }
+                }
+            }
+        }
+
+        snd_seq_port_info_free (portInfo);
+    }
+
+    return port;
+}
+
+static AlsaPort iterateMidiDevices (const bool forInput,
+                                    StringArray& deviceNamesFound,
+                                    const int deviceIndexToOpen)
+{
+    AlsaPort port;
+    const AlsaClient::Ptr client (globalAlsaSequencer (forInput));
+
+    if (snd_seq_t* const seqHandle = client->get())
+    {
+        snd_seq_system_info_t* systemInfo = nullptr;
+        snd_seq_client_info_t* clientInfo = nullptr;
+
+        if (snd_seq_system_info_malloc (&systemInfo) == 0)
+        {
+            if (snd_seq_system_info (seqHandle, systemInfo) == 0
+                 && snd_seq_client_info_malloc (&clientInfo) == 0)
+            {
+                int numClients = snd_seq_system_info_get_cur_clients (systemInfo);
+
+                while (--numClients >= 0 && ! port.isValid())
+                    if (snd_seq_query_next_client (seqHandle, clientInfo) == 0)
+                        port = iterateMidiClient (client, clientInfo, forInput,
+                                                  deviceNamesFound, deviceIndexToOpen);
+
+                snd_seq_client_info_free (clientInfo);
+            }
+
+            snd_seq_system_info_free (systemInfo);
+        }
+
+    }
+
+    deviceNamesFound.appendNumbersToDuplicates (true, true);
+
+    return port;
+}
+
+AlsaPort createMidiDevice (const bool forInput, const String& deviceNameToOpen)
+{
+    AlsaPort port;
+    AlsaClient::Ptr client (new AlsaClient (forInput));
+
+    if (client->get())
+    {
+        client->setName (deviceNameToOpen + (forInput ? " Input" : " Output"));
+        port.createPort (client, forInput ? "in" : "out", forInput);
+    }
+
+    return port;
+}
+
+//==============================================================================
+class MidiOutputDevice
+{
+public:
+    MidiOutputDevice (MidiOutput* const output, const AlsaPort& p)
+        : midiOutput (output), port (p),
+          maxEventSize (16 * 1024)
+    {
+        jassert (port.isValid() && midiOutput != nullptr);
+        snd_midi_event_new (maxEventSize, &midiParser);
+    }
+
+    ~MidiOutputDevice()
+    {
+        snd_midi_event_free (midiParser);
+        port.deletePort();
+    }
+
+    void sendMessageNow (const MidiMessage& message)
+    {
+        if (message.getRawDataSize() > maxEventSize)
+        {
+            maxEventSize = message.getRawDataSize();
+            snd_midi_event_free (midiParser);
+            snd_midi_event_new (maxEventSize, &midiParser);
+        }
+
+        snd_seq_event_t event;
+        snd_seq_ev_clear (&event);
+
+        long numBytes = (long) message.getRawDataSize();
+        const uint8* data = message.getRawData();
+
+        snd_seq_t* seqHandle = port.client->get();
+
+        while (numBytes > 0)
+        {
+            const long numSent = snd_midi_event_encode (midiParser, data, numBytes, &event);
+            if (numSent <= 0)
+                break;
+
+            numBytes -= numSent;
+            data += numSent;
+
+            snd_seq_ev_set_source (&event, 0);
+            snd_seq_ev_set_subs (&event);
+            snd_seq_ev_set_direct (&event);
+
+            snd_seq_event_output (seqHandle, &event);
+        }
+
+        snd_seq_drain_output (seqHandle);
+        snd_midi_event_reset_encode (midiParser);
+    }
+
+private:
+    MidiOutput* const midiOutput;
+    AlsaPort port;
+    snd_midi_event_t* midiParser;
+    int maxEventSize;
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (MidiOutputDevice);
+};
+
+} // namespace
+
+StringArray MidiOutput::getDevices()
+{
+    StringArray devices;
+    iterateMidiDevices (false, devices, -1);
+    return devices;
+}
+
+int MidiOutput::getDefaultDeviceIndex()
+{
+    return 0;
+}
+
+MidiOutput* MidiOutput::openDevice (int deviceIndex)
+{
+    MidiOutput* newDevice = nullptr;
+
+    StringArray devices;
+    AlsaPort port (iterateMidiDevices (false, devices, deviceIndex));
+
+    if (port.isValid())
+    {
+        newDevice = new MidiOutput();
+        newDevice->internal = new MidiOutputDevice (newDevice, port);
+    }
+
+    return newDevice;
+}
+
+MidiOutput* MidiOutput::createNewDevice (const String& deviceName)
+{
+    MidiOutput* newDevice = nullptr;
+
+    AlsaPort port (createMidiDevice (false, deviceName));
+
+    if (port.isValid())
+    {
+        newDevice = new MidiOutput();
+        newDevice->internal = new MidiOutputDevice (newDevice, port);
+    }
+
+    return newDevice;
+}
+
+MidiOutput::~MidiOutput()
+{
+    delete static_cast <MidiOutputDevice*> (internal);
+}
+
+void MidiOutput::sendMessageNow (const MidiMessage& message)
+{
+    static_cast <MidiOutputDevice*> (internal)->sendMessageNow (message);
+}
+
+//==============================================================================
+MidiInput::MidiInput (const String& nm)
+    : name (nm), internal (nullptr)
+{
+}
+
+MidiInput::~MidiInput()
+{
+    stop();
+    delete static_cast <AlsaPortAndCallback*> (internal);
+}
+
+void MidiInput::start()
+{
+    static_cast<AlsaPortAndCallback*> (internal)->enableCallback (true);
+}
+
+void MidiInput::stop()
+{
+    static_cast<AlsaPortAndCallback*> (internal)->enableCallback (false);
+}
+
+int MidiInput::getDefaultDeviceIndex()
+{
+    return 0;
+}
+
+StringArray MidiInput::getDevices()
+{
+    StringArray devices;
+    iterateMidiDevices (true, devices, -1);
+    return devices;
+}
+
+MidiInput* MidiInput::openDevice (int deviceIndex, MidiInputCallback* callback)
+{
+    MidiInput* newDevice = nullptr;
+
+    StringArray devices;
+    AlsaPort port (iterateMidiDevices (true, devices, deviceIndex));
+
+    if (port.isValid())
+    {
+        newDevice = new MidiInput (devices [deviceIndex]);
+        newDevice->internal = new AlsaPortAndCallback (port, newDevice, callback);
+    }
+
+    return newDevice;
+}
+
+MidiInput* MidiInput::createNewDevice (const String& deviceName, MidiInputCallback* callback)
+{
+    MidiInput* newDevice = nullptr;
+
+    AlsaPort port (createMidiDevice (true, deviceName));
+
+    if (port.isValid())
+    {
+        newDevice = new MidiInput (deviceName);
+        newDevice->internal = new AlsaPortAndCallback (port, newDevice, callback);
+    }
+
+    return newDevice;
+}
+
+
+//==============================================================================
+#else
+
+// (These are just stub functions if ALSA is unavailable...)
+
+StringArray MidiOutput::getDevices()                                { return StringArray(); }
+int MidiOutput::getDefaultDeviceIndex()                             { return 0; }
+MidiOutput* MidiOutput::openDevice (int)                            { return nullptr; }
+MidiOutput* MidiOutput::createNewDevice (const String&)             { return nullptr; }
+MidiOutput::~MidiOutput()   {}
+void MidiOutput::sendMessageNow (const MidiMessage&)    {}
+
+MidiInput::MidiInput (const String& nm) : name (nm), internal (nullptr)  {}
+MidiInput::~MidiInput() {}
+void MidiInput::start() {}
+void MidiInput::stop()  {}
+int MidiInput::getDefaultDeviceIndex()      { return 0; }
+StringArray MidiInput::getDevices()         { return StringArray(); }
+MidiInput* MidiInput::openDevice (int, MidiInputCallback*)                  { return nullptr; }
+MidiInput* MidiInput::createNewDevice (const String&, MidiInputCallback*)   { return nullptr; }
+
+#endif

source/modules/juce_audio_devices/native/juce_mac_AudioCDBurner.mm

@@ -0,0 +1,491 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+const int kilobytesPerSecond1x = 176;
+
+struct AudioTrackProducerClass  : public ObjCClass <NSObject>
+{
+    AudioTrackProducerClass()  : ObjCClass <NSObject> ("JUCEAudioTrackProducer_")
+    {
+        addIvar<AudioSourceHolder*> ("source");
+
+        addMethod (@selector (initWithAudioSourceHolder:),     initWithAudioSourceHolder,     "@@:^v");
+        addMethod (@selector (cleanupTrackAfterBurn:),         cleanupTrackAfterBurn,         "v@:@");
+        addMethod (@selector (cleanupTrackAfterVerification:), cleanupTrackAfterVerification, "c@:@");
+        addMethod (@selector (estimateLengthOfTrack:),         estimateLengthOfTrack,         "Q@:@");
+        addMethod (@selector (prepareTrack:forBurn:toMedia:),  prepareTrack,                  "c@:@@@");
+        addMethod (@selector (prepareTrackForVerification:),   prepareTrackForVerification,   "c@:@");
+        addMethod (@selector (produceDataForTrack:intoBuffer:length:atAddress:blockSize:ioFlags:),
+                                                               produceDataForTrack,           "I@:@^cIQI^I");
+        addMethod (@selector (producePreGapForTrack:intoBuffer:length:atAddress:blockSize:ioFlags:),
+                                                               produceDataForTrack,           "I@:@^cIQI^I");
+        addMethod (@selector (verifyDataForTrack:intoBuffer:length:atAddress:blockSize:ioFlags:),
+                                                               produceDataForTrack,           "I@:@^cIQI^I");
+
+        registerClass();
+    }
+
+    struct AudioSourceHolder
+    {
+        AudioSourceHolder (AudioSource* source_, int numFrames)
+            : source (source_), readPosition (0), lengthInFrames (numFrames)
+        {
+        }
+
+        ~AudioSourceHolder()
+        {
+            if (source != nullptr)
+                source->releaseResources();
+        }
+
+        ScopedPointer<AudioSource> source;
+        int readPosition, lengthInFrames;
+    };
+
+private:
+    static id initWithAudioSourceHolder (id self, SEL, AudioSourceHolder* source)
+    {
+        self = sendSuperclassMessage (self, @selector (init));
+        object_setInstanceVariable (self, "source", source);
+        return self;
+    }
+
+    static AudioSourceHolder* getSource (id self)
+    {
+        return getIvar<AudioSourceHolder*> (self, "source");
+    }
+
+    static void dealloc (id self, SEL)
+    {
+        delete getSource (self);
+        sendSuperclassMessage (self, @selector (dealloc));
+    }
+
+    static void cleanupTrackAfterBurn (id self, SEL, DRTrack*) {}
+    static BOOL cleanupTrackAfterVerification (id self, SEL, DRTrack*) { return true; }
+
+    static uint64_t estimateLengthOfTrack (id self, SEL, DRTrack*)
+    {
+        return getSource (self)->lengthInFrames;
+    }
+
+    static BOOL prepareTrack (id self, SEL, DRTrack*, DRBurn*, NSDictionary*)
+    {
+        if (AudioSourceHolder* const source = getSource (self))
+        {
+            source->source->prepareToPlay (44100 / 75, 44100);
+            source->readPosition = 0;
+        }
+
+        return true;
+    }
+
+    static BOOL prepareTrackForVerification (id self, SEL, DRTrack*)
+    {
+        if (AudioSourceHolder* const source = getSource (self))
+            source->source->prepareToPlay (44100 / 75, 44100);
+
+        return true;
+    }
+
+    static uint32_t produceDataForTrack (id self, SEL, DRTrack*, char* buffer,
+                                         uint32_t bufferLength, uint64_t /*address*/,
+                                         uint32_t /*blockSize*/, uint32_t* /*flags*/)
+    {
+        if (AudioSourceHolder* const source = getSource (self))
+        {
+            const int numSamples = jmin ((int) bufferLength / 4,
+                                         (source->lengthInFrames * (44100 / 75)) - source->readPosition);
+
+            if (numSamples > 0)
+            {
+                AudioSampleBuffer tempBuffer (2, numSamples);
+                AudioSourceChannelInfo info (tempBuffer);
+
+                source->source->getNextAudioBlock (info);
+
+                typedef AudioData::Pointer <AudioData::Int16,
+                                            AudioData::LittleEndian,
+                                            AudioData::Interleaved,
+                                            AudioData::NonConst> CDSampleFormat;
+
+                typedef AudioData::Pointer <AudioData::Float32,
+                                            AudioData::NativeEndian,
+                                            AudioData::NonInterleaved,
+                                            AudioData::Const> SourceSampleFormat;
+                CDSampleFormat left (buffer, 2);
+                left.convertSamples (SourceSampleFormat (tempBuffer.getSampleData (0)), numSamples);
+                CDSampleFormat right (buffer + 2, 2);
+                right.convertSamples (SourceSampleFormat (tempBuffer.getSampleData (1)), numSamples);
+
+                source->readPosition += numSamples;
+            }
+
+            return numSamples * 4;
+        }
+
+        return 0;
+    }
+
+    static uint32_t producePreGapForTrack (id self, SEL, DRTrack*, char* buffer,
+                                           uint32_t bufferLength, uint64_t /*address*/,
+                                           uint32_t /*blockSize*/, uint32_t* /*flags*/)
+    {
+        zeromem (buffer, bufferLength);
+        return bufferLength;
+    }
+
+    static BOOL verifyDataForTrack (id self, SEL, DRTrack*, const char*,
+                                    uint32_t /*bufferLength*/, uint64_t /*address*/,
+                                    uint32_t /*blockSize*/, uint32_t* /*flags*/)
+    {
+        return true;
+    }
+};
+
+struct OpenDiskDevice
+{
+    OpenDiskDevice (DRDevice* device_)
+        : device (device_),
+          tracks ([[NSMutableArray alloc] init]),
+          underrunProtection (true)
+    {
+    }
+
+    ~OpenDiskDevice()
+    {
+        [tracks release];
+    }
+
+    void addSourceTrack (AudioSource* source, int numSamples)
+    {
+        if (source != nullptr)
+        {
+            const int numFrames = (numSamples + 587) / 588;
+
+            static AudioTrackProducerClass cls;
+
+            NSObject* producer = [cls.createInstance()  performSelector: @selector (initWithAudioSourceHolder:)
+                                                             withObject: (id) new AudioTrackProducerClass::AudioSourceHolder (source, numFrames)];
+            DRTrack* track = [[DRTrack alloc] initWithProducer: producer];
+
+            {
+                NSMutableDictionary* p = [[track properties] mutableCopy];
+                [p setObject: [DRMSF msfWithFrames: numFrames] forKey: DRTrackLengthKey];
+                [p setObject: [NSNumber numberWithUnsignedShort: 2352] forKey: DRBlockSizeKey];
+                [p setObject: [NSNumber numberWithInt: 0] forKey: DRDataFormKey];
+                [p setObject: [NSNumber numberWithInt: 0] forKey: DRBlockTypeKey];
+                [p setObject: [NSNumber numberWithInt: 0] forKey: DRTrackModeKey];
+                [p setObject: [NSNumber numberWithInt: 0] forKey: DRSessionFormatKey];
+                [track setProperties: p];
+                [p release];
+            }
+
+            [tracks addObject: track];
+
+            [track release];
+            [producer release];
+        }
+    }
+
+    String burn (AudioCDBurner::BurnProgressListener* listener,
+                 bool shouldEject, bool peformFakeBurnForTesting, int burnSpeed)
+    {
+        DRBurn* burn = [DRBurn burnForDevice: device];
+
+        if (! [device acquireExclusiveAccess])
+            return "Couldn't open or write to the CD device";
+
+        [device acquireMediaReservation];
+
+        NSMutableDictionary* d = [[burn properties] mutableCopy];
+        [d autorelease];
+        [d setObject: [NSNumber numberWithBool: peformFakeBurnForTesting] forKey: DRBurnTestingKey];
+        [d setObject: [NSNumber numberWithBool: false] forKey: DRBurnVerifyDiscKey];
+        [d setObject: (shouldEject ? DRBurnCompletionActionEject : DRBurnCompletionActionMount) forKey: DRBurnCompletionActionKey];
+
+        if (burnSpeed > 0)
+            [d setObject: [NSNumber numberWithFloat: burnSpeed * kilobytesPerSecond1x] forKey: DRBurnRequestedSpeedKey];
+
+        if (! underrunProtection)
+            [d setObject: [NSNumber numberWithBool: false] forKey: DRBurnUnderrunProtectionKey];
+
+        [burn setProperties: d];
+
+        [burn writeLayout: tracks];
+
+        for (;;)
+        {
+            Thread::sleep (300);
+            float progress = [[[burn status] objectForKey: DRStatusPercentCompleteKey] floatValue];
+
+            if (listener != nullptr && listener->audioCDBurnProgress (progress))
+            {
+                [burn abort];
+                return "User cancelled the write operation";
+            }
+
+            if ([[[burn status] objectForKey: DRStatusStateKey] isEqualTo: DRStatusStateFailed])
+                return "Write operation failed";
+
+            if ([[[burn status] objectForKey: DRStatusStateKey] isEqualTo: DRStatusStateDone])
+                break;
+
+            NSString* err = (NSString*) [[[burn status] objectForKey: DRErrorStatusKey]
+                                                        objectForKey: DRErrorStatusErrorStringKey];
+
+            if ([err length] > 0)
+                return CharPointer_UTF8 ([err UTF8String]);
+        }
+
+        [device releaseMediaReservation];
+        [device releaseExclusiveAccess];
+        return String::empty;
+    }
+
+    DRDevice* device;
+    NSMutableArray* tracks;
+    bool underrunProtection;
+};
+
+//==============================================================================
+class AudioCDBurner::Pimpl  : public Timer
+{
+public:
+    Pimpl (AudioCDBurner& owner_, const int deviceIndex)
+        : device (0), owner (owner_)
+    {
+        DRDevice* dev = [[DRDevice devices] objectAtIndex: deviceIndex];
+
+        if (dev != nil)
+        {
+            device = new OpenDiskDevice (dev);
+            lastState = getDiskState();
+            startTimer (1000);
+        }
+    }
+
+    ~Pimpl()
+    {
+        stopTimer();
+    }
+
+    void timerCallback() override
+    {
+        const DiskState state = getDiskState();
+
+        if (state != lastState)
+        {
+            lastState = state;
+            owner.sendChangeMessage();
+        }
+    }
+
+    DiskState getDiskState() const
+    {
+        if ([device->device isValid])
+        {
+            NSDictionary* status = [device->device status];
+
+            NSString* state = [status objectForKey: DRDeviceMediaStateKey];
+
+            if ([state isEqualTo: DRDeviceMediaStateNone])
+            {
+                if ([[status objectForKey: DRDeviceIsTrayOpenKey] boolValue])
+                    return trayOpen;
+
+                return noDisc;
+            }
+
+            if ([state isEqualTo: DRDeviceMediaStateMediaPresent])
+            {
+                if ([[[status objectForKey: DRDeviceMediaInfoKey] objectForKey: DRDeviceMediaBlocksFreeKey] intValue] > 0)
+                    return writableDiskPresent;
+                else
+                    return readOnlyDiskPresent;
+            }
+        }
+
+        return unknown;
+    }
+
+    bool openTray()    { return [device->device isValid] && [device->device ejectMedia]; }
+
+    Array<int> getAvailableWriteSpeeds() const
+    {
+        Array<int> results;
+
+        if ([device->device isValid])
+        {
+            NSArray* speeds = [[[device->device status] objectForKey: DRDeviceMediaInfoKey] objectForKey: DRDeviceBurnSpeedsKey];
+            for (unsigned int i = 0; i < [speeds count]; ++i)
+            {
+                const int kbPerSec = [[speeds objectAtIndex: i] intValue];
+                results.add (kbPerSec / kilobytesPerSecond1x);
+            }
+        }
+
+        return results;
+    }
+
+    bool setBufferUnderrunProtection (const bool shouldBeEnabled)
+    {
+        if ([device->device isValid])
+        {
+            device->underrunProtection = shouldBeEnabled;
+            return shouldBeEnabled && [[[device->device status] objectForKey: DRDeviceCanUnderrunProtectCDKey] boolValue];
+        }
+
+        return false;
+    }
+
+    int getNumAvailableAudioBlocks() const
+    {
+        return [[[[device->device status] objectForKey: DRDeviceMediaInfoKey]
+                                          objectForKey: DRDeviceMediaBlocksFreeKey] intValue];
+    }
+
+    ScopedPointer<OpenDiskDevice> device;
+
+private:
+    DiskState lastState;
+    AudioCDBurner& owner;
+};
+
+//==============================================================================
+AudioCDBurner::AudioCDBurner (const int deviceIndex)
+{
+    pimpl = new Pimpl (*this, deviceIndex);
+}
+
+AudioCDBurner::~AudioCDBurner()
+{
+}
+
+AudioCDBurner* AudioCDBurner::openDevice (const int deviceIndex)
+{
+    ScopedPointer <AudioCDBurner> b (new AudioCDBurner (deviceIndex));
+
+    if (b->pimpl->device == nil)
+        b = 0;
+
+    return b.release();
+}
+
+namespace
+{
+    NSArray* findDiskBurnerDevices()
+    {
+        NSMutableArray* results = [NSMutableArray array];
+        NSArray* devs = [DRDevice devices];
+
+        for (int i = 0; i < [devs count]; ++i)
+        {
+            NSDictionary* dic = [[devs objectAtIndex: i] info];
+            NSString* name = [dic valueForKey: DRDeviceProductNameKey];
+            if (name != nil)
+                [results addObject: name];
+        }
+
+        return results;
+    }
+}
+
+StringArray AudioCDBurner::findAvailableDevices()
+{
+    NSArray* names = findDiskBurnerDevices();
+    StringArray s;
+
+    for (unsigned int i = 0; i < [names count]; ++i)
+        s.add (CharPointer_UTF8 ([[names objectAtIndex: i] UTF8String]));
+
+    return s;
+}
+
+AudioCDBurner::DiskState AudioCDBurner::getDiskState() const
+{
+    return pimpl->getDiskState();
+}
+
+bool AudioCDBurner::isDiskPresent() const
+{
+    return getDiskState() == writableDiskPresent;
+}
+
+bool AudioCDBurner::openTray()
+{
+    return pimpl->openTray();
+}
+
+AudioCDBurner::DiskState AudioCDBurner::waitUntilStateChange (int timeOutMilliseconds)
+{
+    const int64 timeout = Time::currentTimeMillis() + timeOutMilliseconds;
+    DiskState oldState = getDiskState();
+    DiskState newState = oldState;
+
+    while (newState == oldState && Time::currentTimeMillis() < timeout)
+    {
+        newState = getDiskState();
+        Thread::sleep (100);
+    }
+
+    return newState;
+}
+
+Array<int> AudioCDBurner::getAvailableWriteSpeeds() const
+{
+    return pimpl->getAvailableWriteSpeeds();
+}
+
+bool AudioCDBurner::setBufferUnderrunProtection (const bool shouldBeEnabled)
+{
+    return pimpl->setBufferUnderrunProtection (shouldBeEnabled);
+}
+
+int AudioCDBurner::getNumAvailableAudioBlocks() const
+{
+    return pimpl->getNumAvailableAudioBlocks();
+}
+
+bool AudioCDBurner::addAudioTrack (AudioSource* source, int numSamps)
+{
+    if ([pimpl->device->device isValid])
+    {
+        pimpl->device->addSourceTrack (source, numSamps);
+        return true;
+    }
+
+    return false;
+}
+
+String AudioCDBurner::burn (AudioCDBurner::BurnProgressListener* listener,
+                            bool ejectDiscAfterwards,
+                            bool performFakeBurnForTesting,
+                            int writeSpeed)
+{
+    if ([pimpl->device->device isValid])
+        return pimpl->device->burn (listener, ejectDiscAfterwards, performFakeBurnForTesting, writeSpeed);
+
+    return "Couldn't open or write to the CD device";
+}

source/modules/juce_audio_devices/native/juce_mac_AudioCDReader.mm

@@ -0,0 +1,261 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+namespace CDReaderHelpers
+{
+    inline const XmlElement* getElementForKey (const XmlElement& xml, const String& key)
+    {
+        forEachXmlChildElementWithTagName (xml, child, "key")
+            if (child->getAllSubText().trim() == key)
+                return child->getNextElement();
+
+        return nullptr;
+    }
+
+    static int getIntValueForKey (const XmlElement& xml, const String& key, int defaultValue = -1)
+    {
+        const XmlElement* const block = getElementForKey (xml, key);
+        return block != nullptr ? block->getAllSubText().trim().getIntValue() : defaultValue;
+    }
+
+    // Get the track offsets for a CD given an XmlElement representing its TOC.Plist.
+    // Returns NULL on success, otherwise a const char* representing an error.
+    static const char* getTrackOffsets (XmlDocument& xmlDocument, Array<int>& offsets)
+    {
+        const ScopedPointer<XmlElement> xml (xmlDocument.getDocumentElement());
+        if (xml == nullptr)
+            return "Couldn't parse XML in file";
+
+        const XmlElement* const dict = xml->getChildByName ("dict");
+        if (dict == nullptr)
+            return "Couldn't get top level dictionary";
+
+        const XmlElement* const sessions = getElementForKey (*dict, "Sessions");
+        if (sessions == nullptr)
+            return "Couldn't find sessions key";
+
+        const XmlElement* const session = sessions->getFirstChildElement();
+        if (session == nullptr)
+            return "Couldn't find first session";
+
+        const int leadOut = getIntValueForKey (*session, "Leadout Block");
+        if (leadOut < 0)
+            return "Couldn't find Leadout Block";
+
+        const XmlElement* const trackArray = getElementForKey (*session, "Track Array");
+        if (trackArray == nullptr)
+            return "Couldn't find Track Array";
+
+        forEachXmlChildElement (*trackArray, track)
+        {
+            const int trackValue = getIntValueForKey (*track, "Start Block");
+            if (trackValue < 0)
+                return "Couldn't find Start Block in the track";
+
+            offsets.add (trackValue * AudioCDReader::samplesPerFrame - 88200);
+        }
+
+        offsets.add (leadOut * AudioCDReader::samplesPerFrame - 88200);
+        return nullptr;
+    }
+
+    static void findDevices (Array<File>& cds)
+    {
+        File volumes ("/Volumes");
+        volumes.findChildFiles (cds, File::findDirectories, false);
+
+        for (int i = cds.size(); --i >= 0;)
+            if (! cds.getReference(i).getChildFile (".TOC.plist").exists())
+                cds.remove (i);
+    }
+
+    struct TrackSorter
+    {
+        static int getCDTrackNumber (const File& file)
+        {
+            return file.getFileName().initialSectionContainingOnly ("0123456789").getIntValue();
+        }
+
+        static int compareElements (const File& first, const File& second)
+        {
+            const int firstTrack  = getCDTrackNumber (first);
+            const int secondTrack = getCDTrackNumber (second);
+
+            jassert (firstTrack > 0 && secondTrack > 0);
+
+            return firstTrack - secondTrack;
+        }
+    };
+}
+
+//==============================================================================
+StringArray AudioCDReader::getAvailableCDNames()
+{
+    Array<File> cds;
+    CDReaderHelpers::findDevices (cds);
+
+    StringArray names;
+
+    for (int i = 0; i < cds.size(); ++i)
+        names.add (cds.getReference(i).getFileName());
+
+    return names;
+}
+
+AudioCDReader* AudioCDReader::createReaderForCD (const int index)
+{
+    Array<File> cds;
+    CDReaderHelpers::findDevices (cds);
+
+    if (cds[index].exists())
+        return new AudioCDReader (cds[index]);
+
+    return nullptr;
+}
+
+AudioCDReader::AudioCDReader (const File& volume)
+   : AudioFormatReader (0, "CD Audio"),
+     volumeDir (volume),
+     currentReaderTrack (-1),
+     reader (0)
+{
+     sampleRate = 44100.0;
+     bitsPerSample = 16;
+     numChannels = 2;
+     usesFloatingPointData = false;
+
+     refreshTrackLengths();
+}
+
+AudioCDReader::~AudioCDReader()
+{
+}
+
+void AudioCDReader::refreshTrackLengths()
+{
+    tracks.clear();
+    trackStartSamples.clear();
+    lengthInSamples = 0;
+
+    volumeDir.findChildFiles (tracks, File::findFiles | File::ignoreHiddenFiles, false, "*.aiff");
+
+    CDReaderHelpers::TrackSorter sorter;
+    tracks.sort (sorter);
+
+    const File toc (volumeDir.getChildFile (".TOC.plist"));
+
+    if (toc.exists())
+    {
+        XmlDocument doc (toc);
+        const char* error = CDReaderHelpers::getTrackOffsets (doc, trackStartSamples);
+        (void) error; // could be logged..
+
+        lengthInSamples = trackStartSamples.getLast() - trackStartSamples.getFirst();
+    }
+}
+
+bool AudioCDReader::readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                                 int64 startSampleInFile, int numSamples)
+{
+    while (numSamples > 0)
+    {
+        int track = -1;
+
+        for (int i = 0; i < trackStartSamples.size() - 1; ++i)
+        {
+            if (startSampleInFile < trackStartSamples.getUnchecked (i + 1))
+            {
+                track = i;
+                break;
+            }
+        }
+
+        if (track < 0)
+            return false;
+
+        if (track != currentReaderTrack)
+        {
+            reader = nullptr;
+
+            if (FileInputStream* const in = tracks [track].createInputStream())
+            {
+                BufferedInputStream* const bin = new BufferedInputStream (in, 65536, true);
+
+                AiffAudioFormat format;
+                reader = format.createReaderFor (bin, true);
+
+                if (reader == nullptr)
+                    currentReaderTrack = -1;
+                else
+                    currentReaderTrack = track;
+            }
+        }
+
+        if (reader == nullptr)
+            return false;
+
+        const int startPos = (int) (startSampleInFile - trackStartSamples.getUnchecked (track));
+        const int numAvailable = (int) jmin ((int64) numSamples, reader->lengthInSamples - startPos);
+
+        reader->readSamples (destSamples, numDestChannels, startOffsetInDestBuffer, startPos, numAvailable);
+
+        numSamples -= numAvailable;
+        startSampleInFile += numAvailable;
+    }
+
+    return true;
+}
+
+bool AudioCDReader::isCDStillPresent() const
+{
+    return volumeDir.exists();
+}
+
+void AudioCDReader::ejectDisk()
+{
+    JUCE_AUTORELEASEPOOL
+    {
+        [[NSWorkspace sharedWorkspace] unmountAndEjectDeviceAtPath: juceStringToNS (volumeDir.getFullPathName())];
+    }
+}
+
+bool AudioCDReader::isTrackAudio (int trackNum) const
+{
+    return tracks [trackNum].hasFileExtension (".aiff");
+}
+
+void AudioCDReader::enableIndexScanning (bool)
+{
+    // any way to do this on a Mac??
+}
+
+int AudioCDReader::getLastIndex() const
+{
+    return 0;
+}
+
+Array<int> AudioCDReader::findIndexesInTrack (const int /*trackNumber*/)
+{
+    return Array<int>();
+}

source/modules/juce_audio_devices/native/juce_mac_CoreMidi.cpp

@@ -0,0 +1,528 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#ifndef JUCE_LOG_COREMIDI_ERRORS
+ #define JUCE_LOG_COREMIDI_ERRORS 1
+#endif
+
+namespace CoreMidiHelpers
+{
+    static bool checkError (const OSStatus err, const int lineNum)
+    {
+        if (err == noErr)
+            return true;
+
+       #if JUCE_LOG_COREMIDI_ERRORS
+        Logger::writeToLog ("CoreMIDI error: " + String (lineNum) + " - " + String::toHexString ((int) err));
+       #endif
+
+        (void) lineNum;
+        return false;
+    }
+
+    #undef CHECK_ERROR
+    #define CHECK_ERROR(a) CoreMidiHelpers::checkError (a, __LINE__)
+
+    //==============================================================================
+    static String getMidiObjectName (MIDIObjectRef entity)
+    {
+        String result;
+        CFStringRef str = nullptr;
+        MIDIObjectGetStringProperty (entity, kMIDIPropertyName, &str);
+
+        if (str != nullptr)
+        {
+            result = String::fromCFString (str);
+            CFRelease (str);
+        }
+
+        return result;
+    }
+
+    static String getEndpointName (MIDIEndpointRef endpoint, bool isExternal)
+    {
+        String result (getMidiObjectName (endpoint));
+
+        MIDIEntityRef entity = 0;  // NB: don't attempt to use nullptr for refs - it fails in some types of build.
+        MIDIEndpointGetEntity (endpoint, &entity);
+
+        if (entity == 0)
+            return result; // probably virtual
+
+        if (result.isEmpty())
+            result = getMidiObjectName (entity);  // endpoint name is empty - try the entity
+
+        // now consider the device's name
+        MIDIDeviceRef device = 0;
+        MIDIEntityGetDevice (entity, &device);
+
+        if (device != 0)
+        {
+            const String deviceName (getMidiObjectName (device));
+
+            if (deviceName.isNotEmpty())
+            {
+                // if an external device has only one entity, throw away
+                // the endpoint name and just use the device name
+                if (isExternal && MIDIDeviceGetNumberOfEntities (device) < 2)
+                {
+                    result = deviceName;
+                }
+                else if (! result.startsWithIgnoreCase (deviceName))
+                {
+                    // prepend the device name to the entity name
+                    result = (deviceName + " " + result).trimEnd();
+                }
+            }
+        }
+
+        return result;
+    }
+
+    static String getConnectedEndpointName (MIDIEndpointRef endpoint)
+    {
+        String result;
+
+        // Does the endpoint have connections?
+        CFDataRef connections = nullptr;
+        int numConnections = 0;
+
+        MIDIObjectGetDataProperty (endpoint, kMIDIPropertyConnectionUniqueID, &connections);
+
+        if (connections != nullptr)
+        {
+            numConnections = ((int) CFDataGetLength (connections)) / (int) sizeof (MIDIUniqueID);
+
+            if (numConnections > 0)
+            {
+                const SInt32* pid = reinterpret_cast <const SInt32*> (CFDataGetBytePtr (connections));
+
+                for (int i = 0; i < numConnections; ++i, ++pid)
+                {
+                    MIDIUniqueID uid = (MIDIUniqueID) ByteOrder::swapIfLittleEndian ((uint32) *pid);
+                    MIDIObjectRef connObject;
+                    MIDIObjectType connObjectType;
+                    OSStatus err = MIDIObjectFindByUniqueID (uid, &connObject, &connObjectType);
+
+                    if (err == noErr)
+                    {
+                        String s;
+
+                        if (connObjectType == kMIDIObjectType_ExternalSource
+                             || connObjectType == kMIDIObjectType_ExternalDestination)
+                        {
+                            // Connected to an external device's endpoint (10.3 and later).
+                            s = getEndpointName (static_cast <MIDIEndpointRef> (connObject), true);
+                        }
+                        else
+                        {
+                            // Connected to an external device (10.2) (or something else, catch-all)
+                            s = getMidiObjectName (connObject);
+                        }
+
+                        if (s.isNotEmpty())
+                        {
+                            if (result.isNotEmpty())
+                                result += ", ";
+
+                            result += s;
+                        }
+                    }
+                }
+            }
+
+            CFRelease (connections);
+        }
+
+        if (result.isEmpty())  // Here, either the endpoint had no connections, or we failed to obtain names for them.
+            result = getEndpointName (endpoint, false);
+
+        return result;
+    }
+
+    static StringArray findDevices (const bool forInput)
+    {
+        const ItemCount num = forInput ? MIDIGetNumberOfSources()
+                                       : MIDIGetNumberOfDestinations();
+        StringArray s;
+
+        for (ItemCount i = 0; i < num; ++i)
+        {
+            MIDIEndpointRef dest = forInput ? MIDIGetSource (i)
+                                            : MIDIGetDestination (i);
+            String name;
+
+            if (dest != 0)
+                name = getConnectedEndpointName (dest);
+
+            if (name.isEmpty())
+                name = "<error>";
+
+            s.add (name);
+        }
+
+        return s;
+    }
+
+    static void globalSystemChangeCallback (const MIDINotification*, void*)
+    {
+        // TODO.. Should pass-on this notification..
+    }
+
+    static String getGlobalMidiClientName()
+    {
+        if (JUCEApplicationBase* const app = JUCEApplicationBase::getInstance())
+            return app->getApplicationName();
+
+        return "JUCE";
+    }
+
+    static MIDIClientRef getGlobalMidiClient()
+    {
+        static MIDIClientRef globalMidiClient = 0;
+
+        if (globalMidiClient == 0)
+        {
+            // Since OSX 10.6, the MIDIClientCreate function will only work
+            // correctly when called from the message thread!
+            jassert (MessageManager::getInstance()->isThisTheMessageThread());
+
+            CFStringRef name = getGlobalMidiClientName().toCFString();
+            CHECK_ERROR (MIDIClientCreate (name, &globalSystemChangeCallback, nullptr, &globalMidiClient));
+            CFRelease (name);
+        }
+
+        return globalMidiClient;
+    }
+
+    //==============================================================================
+    class MidiPortAndEndpoint
+    {
+    public:
+        MidiPortAndEndpoint (MIDIPortRef p, MIDIEndpointRef ep)
+            : port (p), endPoint (ep)
+        {
+        }
+
+        ~MidiPortAndEndpoint()
+        {
+            if (port != 0)
+                MIDIPortDispose (port);
+
+            if (port == 0 && endPoint != 0) // if port == nullptr, it means we created the endpoint, so it's safe to delete it
+                MIDIEndpointDispose (endPoint);
+        }
+
+        void send (const MIDIPacketList* const packets)
+        {
+            if (port != 0)
+                MIDISend (port, endPoint, packets);
+            else
+                MIDIReceived (endPoint, packets);
+        }
+
+        MIDIPortRef port;
+        MIDIEndpointRef endPoint;
+    };
+
+    //==============================================================================
+    class MidiPortAndCallback;
+    CriticalSection callbackLock;
+    Array<MidiPortAndCallback*> activeCallbacks;
+
+    class MidiPortAndCallback
+    {
+    public:
+        MidiPortAndCallback (MidiInputCallback& cb)
+            : input (nullptr), active (false), callback (cb), concatenator (2048)
+        {
+        }
+
+        ~MidiPortAndCallback()
+        {
+            active = false;
+
+            {
+                const ScopedLock sl (callbackLock);
+                activeCallbacks.removeFirstMatchingValue (this);
+            }
+
+            if (portAndEndpoint != 0 && portAndEndpoint->port != 0)
+                CHECK_ERROR (MIDIPortDisconnectSource (portAndEndpoint->port, portAndEndpoint->endPoint));
+        }
+
+        void handlePackets (const MIDIPacketList* const pktlist)
+        {
+            const double time = Time::getMillisecondCounterHiRes() * 0.001;
+
+            const ScopedLock sl (callbackLock);
+            if (activeCallbacks.contains (this) && active)
+            {
+                const MIDIPacket* packet = &pktlist->packet[0];
+
+                for (unsigned int i = 0; i < pktlist->numPackets; ++i)
+                {
+                    concatenator.pushMidiData (packet->data, (int) packet->length, time,
+                                               input, callback);
+
+                    packet = MIDIPacketNext (packet);
+                }
+            }
+        }
+
+        MidiInput* input;
+        ScopedPointer<MidiPortAndEndpoint> portAndEndpoint;
+        volatile bool active;
+
+    private:
+        MidiInputCallback& callback;
+        MidiDataConcatenator concatenator;
+    };
+
+    static void midiInputProc (const MIDIPacketList* pktlist, void* readProcRefCon, void* /*srcConnRefCon*/)
+    {
+        static_cast <MidiPortAndCallback*> (readProcRefCon)->handlePackets (pktlist);
+    }
+}
+
+//==============================================================================
+StringArray MidiOutput::getDevices()        { return CoreMidiHelpers::findDevices (false); }
+int MidiOutput::getDefaultDeviceIndex()     { return 0; }
+
+MidiOutput* MidiOutput::openDevice (int index)
+{
+    MidiOutput* mo = nullptr;
+
+    if (isPositiveAndBelow (index, (int) MIDIGetNumberOfDestinations()))
+    {
+        MIDIEndpointRef endPoint = MIDIGetDestination ((ItemCount) index);
+
+        CFStringRef pname;
+        if (CHECK_ERROR (MIDIObjectGetStringProperty (endPoint, kMIDIPropertyName, &pname)))
+        {
+            MIDIClientRef client = CoreMidiHelpers::getGlobalMidiClient();
+            MIDIPortRef port;
+
+            if (client != 0 && CHECK_ERROR (MIDIOutputPortCreate (client, pname, &port)))
+            {
+                mo = new MidiOutput();
+                mo->internal = new CoreMidiHelpers::MidiPortAndEndpoint (port, endPoint);
+            }
+
+            CFRelease (pname);
+        }
+    }
+
+    return mo;
+}
+
+MidiOutput* MidiOutput::createNewDevice (const String& deviceName)
+{
+    MidiOutput* mo = nullptr;
+    MIDIClientRef client = CoreMidiHelpers::getGlobalMidiClient();
+
+    MIDIEndpointRef endPoint;
+    CFStringRef name = deviceName.toCFString();
+
+    if (client != 0 && CHECK_ERROR (MIDISourceCreate (client, name, &endPoint)))
+    {
+        mo = new MidiOutput();
+        mo->internal = new CoreMidiHelpers::MidiPortAndEndpoint (0, endPoint);
+    }
+
+    CFRelease (name);
+    return mo;
+}
+
+MidiOutput::~MidiOutput()
+{
+    delete static_cast<CoreMidiHelpers::MidiPortAndEndpoint*> (internal);
+}
+
+void MidiOutput::sendMessageNow (const MidiMessage& message)
+{
+   #if JUCE_IOS
+    const MIDITimeStamp timeStamp = mach_absolute_time();
+   #else
+    const MIDITimeStamp timeStamp = AudioGetCurrentHostTime();
+   #endif
+
+    HeapBlock <MIDIPacketList> allocatedPackets;
+    MIDIPacketList stackPacket;
+    MIDIPacketList* packetToSend = &stackPacket;
+    const size_t dataSize = (size_t) message.getRawDataSize();
+
+    if (message.isSysEx())
+    {
+        const int maxPacketSize = 256;
+        int pos = 0, bytesLeft = (int) dataSize;
+        const int numPackets = (bytesLeft + maxPacketSize - 1) / maxPacketSize;
+        allocatedPackets.malloc ((size_t) (32 * (size_t) numPackets + dataSize), 1);
+        packetToSend = allocatedPackets;
+        packetToSend->numPackets = (UInt32) numPackets;
+
+        MIDIPacket* p = packetToSend->packet;
+
+        for (int i = 0; i < numPackets; ++i)
+        {
+            p->timeStamp = timeStamp;
+            p->length = (UInt16) jmin (maxPacketSize, bytesLeft);
+            memcpy (p->data, message.getRawData() + pos, p->length);
+            pos += p->length;
+            bytesLeft -= p->length;
+            p = MIDIPacketNext (p);
+        }
+    }
+    else if (dataSize < 65536) // max packet size
+    {
+        const size_t stackCapacity = sizeof (stackPacket.packet->data);
+
+        if (dataSize > stackCapacity)
+        {
+            allocatedPackets.malloc ((sizeof (MIDIPacketList) - stackCapacity) + dataSize, 1);
+            packetToSend = allocatedPackets;
+        }
+
+        packetToSend->numPackets = 1;
+        MIDIPacket& p = *(packetToSend->packet);
+        p.timeStamp = timeStamp;
+        p.length = (UInt16) dataSize;
+        memcpy (p.data, message.getRawData(), dataSize);
+    }
+    else
+    {
+        jassertfalse; // packet too large to send!
+        return;
+    }
+
+    static_cast<CoreMidiHelpers::MidiPortAndEndpoint*> (internal)->send (packetToSend);
+}
+
+//==============================================================================
+StringArray MidiInput::getDevices()     { return CoreMidiHelpers::findDevices (true); }
+int MidiInput::getDefaultDeviceIndex()  { return 0; }
+
+MidiInput* MidiInput::openDevice (int index, MidiInputCallback* callback)
+{
+    jassert (callback != nullptr);
+
+    using namespace CoreMidiHelpers;
+    MidiInput* newInput = nullptr;
+
+    if (isPositiveAndBelow (index, (int) MIDIGetNumberOfSources()))
+    {
+        if (MIDIEndpointRef endPoint = MIDIGetSource ((ItemCount) index))
+        {
+            CFStringRef name;
+
+            if (CHECK_ERROR (MIDIObjectGetStringProperty (endPoint, kMIDIPropertyName, &name)))
+            {
+                if (MIDIClientRef client = getGlobalMidiClient())
+                {
+                    MIDIPortRef port;
+                    ScopedPointer <MidiPortAndCallback> mpc (new MidiPortAndCallback (*callback));
+
+                    if (CHECK_ERROR (MIDIInputPortCreate (client, name, midiInputProc, mpc, &port)))
+                    {
+                        if (CHECK_ERROR (MIDIPortConnectSource (port, endPoint, nullptr)))
+                        {
+                            mpc->portAndEndpoint = new MidiPortAndEndpoint (port, endPoint);
+
+                            newInput = new MidiInput (getDevices() [index]);
+                            mpc->input = newInput;
+                            newInput->internal = mpc;
+
+                            const ScopedLock sl (callbackLock);
+                            activeCallbacks.add (mpc.release());
+                        }
+                        else
+                        {
+                            CHECK_ERROR (MIDIPortDispose (port));
+                        }
+                    }
+                }
+            }
+
+            CFRelease (name);
+        }
+    }
+
+    return newInput;
+}
+
+MidiInput* MidiInput::createNewDevice (const String& deviceName, MidiInputCallback* callback)
+{
+    jassert (callback != nullptr);
+
+    using namespace CoreMidiHelpers;
+    MidiInput* mi = nullptr;
+
+    if (MIDIClientRef client = getGlobalMidiClient())
+    {
+        ScopedPointer <MidiPortAndCallback> mpc (new MidiPortAndCallback (*callback));
+        mpc->active = false;
+
+        MIDIEndpointRef endPoint;
+        CFStringRef name = deviceName.toCFString();
+
+        if (CHECK_ERROR (MIDIDestinationCreate (client, name, midiInputProc, mpc, &endPoint)))
+        {
+            mpc->portAndEndpoint = new MidiPortAndEndpoint (0, endPoint);
+
+            mi = new MidiInput (deviceName);
+            mpc->input = mi;
+            mi->internal = mpc;
+
+            const ScopedLock sl (callbackLock);
+            activeCallbacks.add (mpc.release());
+        }
+
+        CFRelease (name);
+    }
+
+    return mi;
+}
+
+MidiInput::MidiInput (const String& nm)  : name (nm)
+{
+}
+
+MidiInput::~MidiInput()
+{
+    delete static_cast<CoreMidiHelpers::MidiPortAndCallback*> (internal);
+}
+
+void MidiInput::start()
+{
+    const ScopedLock sl (CoreMidiHelpers::callbackLock);
+    static_cast<CoreMidiHelpers::MidiPortAndCallback*> (internal)->active = true;
+}
+
+void MidiInput::stop()
+{
+    const ScopedLock sl (CoreMidiHelpers::callbackLock);
+    static_cast<CoreMidiHelpers::MidiPortAndCallback*> (internal)->active = false;
+}
+
+#undef CHECK_ERROR

source/modules/juce_audio_devices/native/juce_win32_AudioCDBurner.cpp

@@ -0,0 +1,411 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+namespace CDBurnerHelpers
+{
+    IDiscRecorder* enumCDBurners (StringArray* list, int indexToOpen, IDiscMaster** master)
+    {
+        CoInitialize (0);
+
+        IDiscMaster* dm;
+        IDiscRecorder* result = nullptr;
+
+        if (SUCCEEDED (CoCreateInstance (CLSID_MSDiscMasterObj, 0,
+                                         CLSCTX_INPROC_SERVER | CLSCTX_LOCAL_SERVER,
+                                         IID_IDiscMaster,
+                                         (void**) &dm)))
+        {
+            if (SUCCEEDED (dm->Open()))
+            {
+                IEnumDiscRecorders* drEnum = nullptr;
+
+                if (SUCCEEDED (dm->EnumDiscRecorders (&drEnum)))
+                {
+                    IDiscRecorder* dr = nullptr;
+                    DWORD dummy;
+                    int index = 0;
+
+                    while (drEnum->Next (1, &dr, &dummy) == S_OK)
+                    {
+                        if (indexToOpen == index)
+                        {
+                            result = dr;
+                            break;
+                        }
+                        else if (list != nullptr)
+                        {
+                            BSTR path;
+
+                            if (SUCCEEDED (dr->GetPath (&path)))
+                                list->add ((const WCHAR*) path);
+                        }
+
+                        ++index;
+                        dr->Release();
+                    }
+
+                    drEnum->Release();
+                }
+
+                if (master == 0)
+                    dm->Close();
+            }
+
+            if (master != nullptr)
+                *master = dm;
+            else
+                dm->Release();
+        }
+
+        return result;
+    }
+}
+
+//==============================================================================
+class AudioCDBurner::Pimpl  : public ComBaseClassHelper <IDiscMasterProgressEvents>,
+                              public Timer
+{
+public:
+    Pimpl (AudioCDBurner& owner_, IDiscMaster* discMaster_, IDiscRecorder* discRecorder_)
+      : owner (owner_), discMaster (discMaster_), discRecorder (discRecorder_), redbook (0),
+        listener (0), progress (0), shouldCancel (false)
+    {
+        HRESULT hr = discMaster->SetActiveDiscMasterFormat (IID_IRedbookDiscMaster, (void**) &redbook);
+        jassert (SUCCEEDED (hr));
+        hr = discMaster->SetActiveDiscRecorder (discRecorder);
+        //jassert (SUCCEEDED (hr));
+
+        lastState = getDiskState();
+        startTimer (2000);
+    }
+
+    ~Pimpl()  {}
+
+    void releaseObjects()
+    {
+        discRecorder->Close();
+        if (redbook != nullptr)
+            redbook->Release();
+        discRecorder->Release();
+        discMaster->Release();
+        Release();
+    }
+
+    JUCE_COMRESULT QueryCancel (boolean* pbCancel)
+    {
+        if (listener != nullptr && ! shouldCancel)
+            shouldCancel = listener->audioCDBurnProgress (progress);
+
+        *pbCancel = shouldCancel;
+
+        return S_OK;
+    }
+
+    JUCE_COMRESULT NotifyBlockProgress (long nCompleted, long nTotal)
+    {
+        progress = nCompleted / (float) nTotal;
+        shouldCancel = listener != nullptr && listener->audioCDBurnProgress (progress);
+
+        return E_NOTIMPL;
+    }
+
+    JUCE_COMRESULT NotifyPnPActivity (void)                              { return E_NOTIMPL; }
+    JUCE_COMRESULT NotifyAddProgress (long /*nCompletedSteps*/, long /*nTotalSteps*/)    { return E_NOTIMPL; }
+    JUCE_COMRESULT NotifyTrackProgress (long /*nCurrentTrack*/, long /*nTotalTracks*/)   { return E_NOTIMPL; }
+    JUCE_COMRESULT NotifyPreparingBurn (long /*nEstimatedSeconds*/)      { return E_NOTIMPL; }
+    JUCE_COMRESULT NotifyClosingDisc (long /*nEstimatedSeconds*/)        { return E_NOTIMPL; }
+    JUCE_COMRESULT NotifyBurnComplete (HRESULT /*status*/)               { return E_NOTIMPL; }
+    JUCE_COMRESULT NotifyEraseComplete (HRESULT /*status*/)              { return E_NOTIMPL; }
+
+    class ScopedDiscOpener
+    {
+    public:
+        ScopedDiscOpener (Pimpl& p) : pimpl (p) { pimpl.discRecorder->OpenExclusive(); }
+        ~ScopedDiscOpener()                     { pimpl.discRecorder->Close(); }
+
+    private:
+        Pimpl& pimpl;
+
+        JUCE_DECLARE_NON_COPYABLE (ScopedDiscOpener)
+    };
+
+    DiskState getDiskState()
+    {
+        const ScopedDiscOpener opener (*this);
+
+        long type, flags;
+        HRESULT hr = discRecorder->QueryMediaType (&type, &flags);
+
+        if (FAILED (hr))
+            return unknown;
+
+        if (type != 0 && (flags & MEDIA_WRITABLE) != 0)
+            return writableDiskPresent;
+
+        if (type == 0)
+            return noDisc;
+
+        return readOnlyDiskPresent;
+    }
+
+    int getIntProperty (const LPOLESTR name, const int defaultReturn) const
+    {
+        ComSmartPtr<IPropertyStorage> prop;
+        if (FAILED (discRecorder->GetRecorderProperties (prop.resetAndGetPointerAddress())))
+            return defaultReturn;
+
+        PROPSPEC iPropSpec;
+        iPropSpec.ulKind = PRSPEC_LPWSTR;
+        iPropSpec.lpwstr = name;
+
+        PROPVARIANT iPropVariant;
+        return FAILED (prop->ReadMultiple (1, &iPropSpec, &iPropVariant))
+                   ? defaultReturn : (int) iPropVariant.lVal;
+    }
+
+    bool setIntProperty (const LPOLESTR name, const int value) const
+    {
+        ComSmartPtr<IPropertyStorage> prop;
+        if (FAILED (discRecorder->GetRecorderProperties (prop.resetAndGetPointerAddress())))
+            return false;
+
+        PROPSPEC iPropSpec;
+        iPropSpec.ulKind = PRSPEC_LPWSTR;
+        iPropSpec.lpwstr = name;
+
+        PROPVARIANT iPropVariant;
+        if (FAILED (prop->ReadMultiple (1, &iPropSpec, &iPropVariant)))
+            return false;
+
+        iPropVariant.lVal = (long) value;
+        return SUCCEEDED (prop->WriteMultiple (1, &iPropSpec, &iPropVariant, iPropVariant.vt))
+                && SUCCEEDED (discRecorder->SetRecorderProperties (prop));
+    }
+
+    void timerCallback() override
+    {
+        const DiskState state = getDiskState();
+
+        if (state != lastState)
+        {
+            lastState = state;
+            owner.sendChangeMessage();
+        }
+    }
+
+    AudioCDBurner& owner;
+    DiskState lastState;
+    IDiscMaster* discMaster;
+    IDiscRecorder* discRecorder;
+    IRedbookDiscMaster* redbook;
+    AudioCDBurner::BurnProgressListener* listener;
+    float progress;
+    bool shouldCancel;
+};
+
+//==============================================================================
+AudioCDBurner::AudioCDBurner (const int deviceIndex)
+{
+    IDiscMaster* discMaster = nullptr;
+    IDiscRecorder* discRecorder = CDBurnerHelpers::enumCDBurners (0, deviceIndex, &discMaster);
+
+    if (discRecorder != nullptr)
+        pimpl = new Pimpl (*this, discMaster, discRecorder);
+}
+
+AudioCDBurner::~AudioCDBurner()
+{
+    if (pimpl != nullptr)
+        pimpl.release()->releaseObjects();
+}
+
+StringArray AudioCDBurner::findAvailableDevices()
+{
+    StringArray devs;
+    CDBurnerHelpers::enumCDBurners (&devs, -1, 0);
+    return devs;
+}
+
+AudioCDBurner* AudioCDBurner::openDevice (const int deviceIndex)
+{
+    ScopedPointer<AudioCDBurner> b (new AudioCDBurner (deviceIndex));
+
+    if (b->pimpl == 0)
+        b = nullptr;
+
+    return b.release();
+}
+
+AudioCDBurner::DiskState AudioCDBurner::getDiskState() const
+{
+    return pimpl->getDiskState();
+}
+
+bool AudioCDBurner::isDiskPresent() const
+{
+    return getDiskState() == writableDiskPresent;
+}
+
+bool AudioCDBurner::openTray()
+{
+    const Pimpl::ScopedDiscOpener opener (*pimpl);
+    return SUCCEEDED (pimpl->discRecorder->Eject());
+}
+
+AudioCDBurner::DiskState AudioCDBurner::waitUntilStateChange (int timeOutMilliseconds)
+{
+    const int64 timeout = Time::currentTimeMillis() + timeOutMilliseconds;
+    DiskState oldState = getDiskState();
+    DiskState newState = oldState;
+
+    while (newState == oldState && Time::currentTimeMillis() < timeout)
+    {
+        newState = getDiskState();
+        Thread::sleep (jmin (250, (int) (timeout - Time::currentTimeMillis())));
+    }
+
+    return newState;
+}
+
+Array<int> AudioCDBurner::getAvailableWriteSpeeds() const
+{
+    Array<int> results;
+    const int maxSpeed = pimpl->getIntProperty (L"MaxWriteSpeed", 1);
+    const int speeds[] = { 1, 2, 4, 8, 12, 16, 20, 24, 32, 40, 64, 80 };
+
+    for (int i = 0; i < numElementsInArray (speeds); ++i)
+        if (speeds[i] <= maxSpeed)
+            results.add (speeds[i]);
+
+    results.addIfNotAlreadyThere (maxSpeed);
+    return results;
+}
+
+bool AudioCDBurner::setBufferUnderrunProtection (const bool shouldBeEnabled)
+{
+    if (pimpl->getIntProperty (L"BufferUnderrunFreeCapable", 0) == 0)
+        return false;
+
+    pimpl->setIntProperty (L"EnableBufferUnderrunFree", shouldBeEnabled ? -1 : 0);
+    return pimpl->getIntProperty (L"EnableBufferUnderrunFree", 0) != 0;
+}
+
+int AudioCDBurner::getNumAvailableAudioBlocks() const
+{
+    long blocksFree = 0;
+    pimpl->redbook->GetAvailableAudioTrackBlocks (&blocksFree);
+    return blocksFree;
+}
+
+String AudioCDBurner::burn (AudioCDBurner::BurnProgressListener* listener, bool ejectDiscAfterwards,
+                            bool performFakeBurnForTesting, int writeSpeed)
+{
+    pimpl->setIntProperty (L"WriteSpeed", writeSpeed > 0 ? writeSpeed : -1);
+
+    pimpl->listener = listener;
+    pimpl->progress = 0;
+    pimpl->shouldCancel = false;
+
+    UINT_PTR cookie;
+    HRESULT hr = pimpl->discMaster->ProgressAdvise ((AudioCDBurner::Pimpl*) pimpl, &cookie);
+
+    hr = pimpl->discMaster->RecordDisc (performFakeBurnForTesting,
+                                        ejectDiscAfterwards);
+
+    String error;
+    if (hr != S_OK)
+    {
+        const char* e = "Couldn't open or write to the CD device";
+
+        if (hr == IMAPI_E_USERABORT)
+            e = "User cancelled the write operation";
+        else if (hr == IMAPI_E_MEDIUM_NOTPRESENT || hr == IMAPI_E_TRACKOPEN)
+            e = "No Disk present";
+
+        error = e;
+    }
+
+    pimpl->discMaster->ProgressUnadvise (cookie);
+    pimpl->listener = 0;
+
+    return error;
+}
+
+bool AudioCDBurner::addAudioTrack (AudioSource* audioSource, int numSamples)
+{
+    if (audioSource == 0)
+        return false;
+
+    ScopedPointer<AudioSource> source (audioSource);
+
+    long bytesPerBlock;
+    HRESULT hr = pimpl->redbook->GetAudioBlockSize (&bytesPerBlock);
+
+    const int samplesPerBlock = bytesPerBlock / 4;
+    bool ok = true;
+
+    hr = pimpl->redbook->CreateAudioTrack ((long) numSamples / (bytesPerBlock * 4));
+
+    HeapBlock <byte> buffer (bytesPerBlock);
+    AudioSampleBuffer sourceBuffer (2, samplesPerBlock);
+    int samplesDone = 0;
+
+    source->prepareToPlay (samplesPerBlock, 44100.0);
+
+    while (ok)
+    {
+        {
+            AudioSourceChannelInfo info (&sourceBuffer, 0, samplesPerBlock);
+            sourceBuffer.clear();
+
+            source->getNextAudioBlock (info);
+        }
+
+        buffer.clear (bytesPerBlock);
+
+        typedef AudioData::Pointer <AudioData::Int16, AudioData::LittleEndian,
+                                    AudioData::Interleaved, AudioData::NonConst> CDSampleFormat;
+
+        typedef AudioData::Pointer <AudioData::Float32, AudioData::NativeEndian,
+                                    AudioData::NonInterleaved, AudioData::Const> SourceSampleFormat;
+
+        CDSampleFormat left (buffer, 2);
+        left.convertSamples (SourceSampleFormat (sourceBuffer.getSampleData (0)), samplesPerBlock);
+        CDSampleFormat right (buffer + 2, 2);
+        right.convertSamples (SourceSampleFormat (sourceBuffer.getSampleData (1)), samplesPerBlock);
+
+        hr = pimpl->redbook->AddAudioTrackBlocks (buffer, bytesPerBlock);
+
+        if (FAILED (hr))
+            ok = false;
+
+        samplesDone += samplesPerBlock;
+
+        if (samplesDone >= numSamples)
+            break;
+    }
+
+    hr = pimpl->redbook->CloseAudioTrack();
+    return ok && hr == S_OK;
+}

source/modules/juce_audio_devices/native/juce_win32_Midi.cpp

@@ -0,0 +1,483 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+class MidiInCollector
+{
+public:
+    MidiInCollector (MidiInput* const input_,
+                     MidiInputCallback& callback_)
+        : deviceHandle (0),
+          input (input_),
+          callback (callback_),
+          concatenator (4096),
+          isStarted (false),
+          startTime (0)
+    {
+    }
+
+    ~MidiInCollector()
+    {
+        stop();
+
+        if (deviceHandle != 0)
+        {
+            for (int count = 5; --count >= 0;)
+            {
+                if (midiInClose (deviceHandle) == MMSYSERR_NOERROR)
+                    break;
+
+                Sleep (20);
+            }
+        }
+    }
+
+    //==============================================================================
+    void handleMessage (const uint8* bytes, const uint32 timeStamp)
+    {
+        if (bytes[0] >= 0x80 && isStarted)
+        {
+            concatenator.pushMidiData (bytes, MidiMessage::getMessageLengthFromFirstByte (bytes[0]),
+                                       convertTimeStamp (timeStamp), input, callback);
+            writeFinishedBlocks();
+        }
+    }
+
+    void handleSysEx (MIDIHDR* const hdr, const uint32 timeStamp)
+    {
+        if (isStarted && hdr->dwBytesRecorded > 0)
+        {
+            concatenator.pushMidiData (hdr->lpData, (int) hdr->dwBytesRecorded,
+                                       convertTimeStamp (timeStamp), input, callback);
+            writeFinishedBlocks();
+        }
+    }
+
+    void start()
+    {
+        if (deviceHandle != 0 && ! isStarted)
+        {
+            activeMidiCollectors.addIfNotAlreadyThere (this);
+
+            for (int i = 0; i < (int) numHeaders; ++i)
+            {
+                headers[i].prepare (deviceHandle);
+                headers[i].write (deviceHandle);
+            }
+
+            startTime = Time::getMillisecondCounterHiRes();
+            MMRESULT res = midiInStart (deviceHandle);
+
+            if (res == MMSYSERR_NOERROR)
+            {
+                concatenator.reset();
+                isStarted = true;
+            }
+            else
+            {
+                unprepareAllHeaders();
+            }
+        }
+    }
+
+    void stop()
+    {
+        if (isStarted)
+        {
+            isStarted = false;
+            midiInReset (deviceHandle);
+            midiInStop (deviceHandle);
+            activeMidiCollectors.removeFirstMatchingValue (this);
+            unprepareAllHeaders();
+            concatenator.reset();
+        }
+    }
+
+    static void CALLBACK midiInCallback (HMIDIIN, UINT uMsg, DWORD_PTR dwInstance,
+                                         DWORD_PTR midiMessage, DWORD_PTR timeStamp)
+    {
+        MidiInCollector* const collector = reinterpret_cast <MidiInCollector*> (dwInstance);
+
+        if (activeMidiCollectors.contains (collector))
+        {
+            if (uMsg == MIM_DATA)
+                collector->handleMessage ((const uint8*) &midiMessage, (uint32) timeStamp);
+            else if (uMsg == MIM_LONGDATA)
+                collector->handleSysEx ((MIDIHDR*) midiMessage, (uint32) timeStamp);
+        }
+    }
+
+    HMIDIIN deviceHandle;
+
+private:
+    static Array <MidiInCollector*, CriticalSection> activeMidiCollectors;
+
+    MidiInput* input;
+    MidiInputCallback& callback;
+    MidiDataConcatenator concatenator;
+    bool volatile isStarted;
+    double startTime;
+
+    class MidiHeader
+    {
+    public:
+        MidiHeader() {}
+
+        void prepare (HMIDIIN deviceHandle)
+        {
+            zerostruct (hdr);
+            hdr.lpData = data;
+            hdr.dwBufferLength = (DWORD) numElementsInArray (data);
+
+            midiInPrepareHeader (deviceHandle, &hdr, sizeof (hdr));
+        }
+
+        void unprepare (HMIDIIN deviceHandle)
+        {
+            if ((hdr.dwFlags & WHDR_DONE) != 0)
+            {
+                int c = 10;
+                while (--c >= 0 && midiInUnprepareHeader (deviceHandle, &hdr, sizeof (hdr)) == MIDIERR_STILLPLAYING)
+                    Thread::sleep (20);
+
+                jassert (c >= 0);
+            }
+        }
+
+        void write (HMIDIIN deviceHandle)
+        {
+            hdr.dwBytesRecorded = 0;
+            midiInAddBuffer (deviceHandle, &hdr, sizeof (hdr));
+        }
+
+        void writeIfFinished (HMIDIIN deviceHandle)
+        {
+            if ((hdr.dwFlags & WHDR_DONE) != 0)
+                write (deviceHandle);
+        }
+
+    private:
+        MIDIHDR hdr;
+        char data [256];
+
+        JUCE_DECLARE_NON_COPYABLE (MidiHeader)
+    };
+
+    enum { numHeaders = 32 };
+    MidiHeader headers [numHeaders];
+
+    void writeFinishedBlocks()
+    {
+        for (int i = 0; i < (int) numHeaders; ++i)
+            headers[i].writeIfFinished (deviceHandle);
+    }
+
+    void unprepareAllHeaders()
+    {
+        for (int i = 0; i < (int) numHeaders; ++i)
+            headers[i].unprepare (deviceHandle);
+    }
+
+    double convertTimeStamp (uint32 timeStamp)
+    {
+        double t = startTime + timeStamp;
+
+        const double now = Time::getMillisecondCounterHiRes();
+        if (t > now)
+        {
+            if (t > now + 2.0)
+                startTime -= 1.0;
+
+            t = now;
+        }
+
+        return t * 0.001;
+    }
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (MidiInCollector)
+};
+
+Array <MidiInCollector*, CriticalSection> MidiInCollector::activeMidiCollectors;
+
+
+//==============================================================================
+StringArray MidiInput::getDevices()
+{
+    StringArray s;
+    const UINT num = midiInGetNumDevs();
+
+    for (UINT i = 0; i < num; ++i)
+    {
+        MIDIINCAPS mc = { 0 };
+
+        if (midiInGetDevCaps (i, &mc, sizeof (mc)) == MMSYSERR_NOERROR)
+            s.add (String (mc.szPname, sizeof (mc.szPname)));
+    }
+
+    return s;
+}
+
+int MidiInput::getDefaultDeviceIndex()
+{
+    return 0;
+}
+
+MidiInput* MidiInput::openDevice (const int index, MidiInputCallback* const callback)
+{
+    if (callback == nullptr)
+        return nullptr;
+
+    UINT deviceId = MIDI_MAPPER;
+    int n = 0;
+    String name;
+
+    const UINT num = midiInGetNumDevs();
+
+    for (UINT i = 0; i < num; ++i)
+    {
+        MIDIINCAPS mc = { 0 };
+
+        if (midiInGetDevCaps (i, &mc, sizeof (mc)) == MMSYSERR_NOERROR)
+        {
+            if (index == n)
+            {
+                deviceId = i;
+                name = String (mc.szPname, (size_t) numElementsInArray (mc.szPname));
+                break;
+            }
+
+            ++n;
+        }
+    }
+
+    ScopedPointer <MidiInput> in (new MidiInput (name));
+    ScopedPointer <MidiInCollector> collector (new MidiInCollector (in, *callback));
+
+    HMIDIIN h;
+    MMRESULT err = midiInOpen (&h, deviceId,
+                               (DWORD_PTR) &MidiInCollector::midiInCallback,
+                               (DWORD_PTR) (MidiInCollector*) collector,
+                               CALLBACK_FUNCTION);
+
+    if (err == MMSYSERR_NOERROR)
+    {
+        collector->deviceHandle = h;
+        in->internal = collector.release();
+        return in.release();
+    }
+
+    return nullptr;
+}
+
+MidiInput::MidiInput (const String& name_)
+    : name (name_),
+      internal (0)
+{
+}
+
+MidiInput::~MidiInput()
+{
+    delete static_cast <MidiInCollector*> (internal);
+}
+
+void MidiInput::start()     { static_cast <MidiInCollector*> (internal)->start(); }
+void MidiInput::stop()      { static_cast <MidiInCollector*> (internal)->stop(); }
+
+
+//==============================================================================
+struct MidiOutHandle
+{
+    int refCount;
+    UINT deviceId;
+    HMIDIOUT handle;
+
+    static Array<MidiOutHandle*> activeHandles;
+
+private:
+    JUCE_LEAK_DETECTOR (MidiOutHandle)
+};
+
+Array<MidiOutHandle*> MidiOutHandle::activeHandles;
+
+//==============================================================================
+StringArray MidiOutput::getDevices()
+{
+    StringArray s;
+    const UINT num = midiOutGetNumDevs();
+
+    for (UINT i = 0; i < num; ++i)
+    {
+        MIDIOUTCAPS mc = { 0 };
+
+        if (midiOutGetDevCaps (i, &mc, sizeof (mc)) == MMSYSERR_NOERROR)
+            s.add (String (mc.szPname, sizeof (mc.szPname)));
+    }
+
+    return s;
+}
+
+int MidiOutput::getDefaultDeviceIndex()
+{
+    const UINT num = midiOutGetNumDevs();
+    int n = 0;
+
+    for (UINT i = 0; i < num; ++i)
+    {
+        MIDIOUTCAPS mc = { 0 };
+
+        if (midiOutGetDevCaps (i, &mc, sizeof (mc)) == MMSYSERR_NOERROR)
+        {
+            if ((mc.wTechnology & MOD_MAPPER) != 0)
+                return n;
+
+             ++n;
+        }
+    }
+
+    return 0;
+}
+
+MidiOutput* MidiOutput::openDevice (int index)
+{
+    UINT deviceId = MIDI_MAPPER;
+    const UINT num = midiOutGetNumDevs();
+    int n = 0;
+
+    for (UINT i = 0; i < num; ++i)
+    {
+        MIDIOUTCAPS mc = { 0 };
+
+        if (midiOutGetDevCaps (i, &mc, sizeof (mc)) == MMSYSERR_NOERROR)
+        {
+            // use the microsoft sw synth as a default - best not to allow deviceId
+            // to be MIDI_MAPPER, or else device sharing breaks
+            if (String (mc.szPname, sizeof (mc.szPname)).containsIgnoreCase ("microsoft"))
+                deviceId = i;
+
+            if (index == n)
+            {
+                deviceId = i;
+                break;
+            }
+
+            ++n;
+        }
+    }
+
+    for (int i = MidiOutHandle::activeHandles.size(); --i >= 0;)
+    {
+        MidiOutHandle* const han = MidiOutHandle::activeHandles.getUnchecked(i);
+
+        if (han->deviceId == deviceId)
+        {
+            han->refCount++;
+
+            MidiOutput* const out = new MidiOutput();
+            out->internal = han;
+            return out;
+        }
+    }
+
+    for (int i = 4; --i >= 0;)
+    {
+        HMIDIOUT h = 0;
+        MMRESULT res = midiOutOpen (&h, deviceId, 0, 0, CALLBACK_NULL);
+
+        if (res == MMSYSERR_NOERROR)
+        {
+            MidiOutHandle* const han = new MidiOutHandle();
+            han->deviceId = deviceId;
+            han->refCount = 1;
+            han->handle = h;
+            MidiOutHandle::activeHandles.add (han);
+
+            MidiOutput* const out = new MidiOutput();
+            out->internal = han;
+            return out;
+        }
+        else if (res == MMSYSERR_ALLOCATED)
+        {
+            Sleep (100);
+        }
+        else
+        {
+            break;
+        }
+    }
+
+    return nullptr;
+}
+
+MidiOutput::~MidiOutput()
+{
+    stopBackgroundThread();
+
+    MidiOutHandle* const h = static_cast <MidiOutHandle*> (internal);
+
+    if (MidiOutHandle::activeHandles.contains (h) && --(h->refCount) == 0)
+    {
+        midiOutClose (h->handle);
+        MidiOutHandle::activeHandles.removeFirstMatchingValue (h);
+        delete h;
+    }
+}
+
+void MidiOutput::sendMessageNow (const MidiMessage& message)
+{
+    const MidiOutHandle* const handle = static_cast <const MidiOutHandle*> (internal);
+
+    if (message.getRawDataSize() > 3 || message.isSysEx())
+    {
+        MIDIHDR h = { 0 };
+
+        h.lpData = (char*) message.getRawData();
+        h.dwBytesRecorded = h.dwBufferLength  = (DWORD) message.getRawDataSize();
+
+        if (midiOutPrepareHeader (handle->handle, &h, sizeof (MIDIHDR)) == MMSYSERR_NOERROR)
+        {
+            MMRESULT res = midiOutLongMsg (handle->handle, &h, sizeof (MIDIHDR));
+
+            if (res == MMSYSERR_NOERROR)
+            {
+                while ((h.dwFlags & MHDR_DONE) == 0)
+                    Sleep (1);
+
+                int count = 500; // 1 sec timeout
+
+                while (--count >= 0)
+                {
+                    res = midiOutUnprepareHeader (handle->handle, &h, sizeof (MIDIHDR));
+
+                    if (res == MIDIERR_STILLPLAYING)
+                        Sleep (2);
+                    else
+                        break;
+                }
+            }
+        }
+    }
+    else
+    {
+        midiOutShortMsg (handle->handle, *(unsigned int*) message.getRawData());
+    }
+}

source/modules/juce_audio_devices/sources/juce_AudioSourcePlayer.cpp

@@ -0,0 +1,183 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+AudioSourcePlayer::AudioSourcePlayer()
+    : source (nullptr),
+      sampleRate (0),
+      bufferSize (0),
+      tempBuffer (2, 8),
+      lastGain (1.0f),
+      gain (1.0f)
+{
+}
+
+AudioSourcePlayer::~AudioSourcePlayer()
+{
+    setSource (nullptr);
+}
+
+void AudioSourcePlayer::setSource (AudioSource* newSource)
+{
+    if (source != newSource)
+    {
+        AudioSource* const oldSource = source;
+
+        if (newSource != nullptr && bufferSize > 0 && sampleRate > 0)
+            newSource->prepareToPlay (bufferSize, sampleRate);
+
+        {
+            const ScopedLock sl (readLock);
+            source = newSource;
+        }
+
+        if (oldSource != nullptr)
+            oldSource->releaseResources();
+    }
+}
+
+void AudioSourcePlayer::setGain (const float newGain) noexcept
+{
+    gain = newGain;
+}
+
+void AudioSourcePlayer::audioDeviceIOCallback (const float** inputChannelData,
+                                               int totalNumInputChannels,
+                                               float** outputChannelData,
+                                               int totalNumOutputChannels,
+                                               int numSamples)
+{
+    // these should have been prepared by audioDeviceAboutToStart()...
+    jassert (sampleRate > 0 && bufferSize > 0);
+
+    const ScopedLock sl (readLock);
+
+    if (source != nullptr)
+    {
+        int numActiveChans = 0, numInputs = 0, numOutputs = 0;
+
+        // messy stuff needed to compact the channels down into an array
+        // of non-zero pointers..
+        for (int i = 0; i < totalNumInputChannels; ++i)
+        {
+            if (inputChannelData[i] != nullptr)
+            {
+                inputChans [numInputs++] = inputChannelData[i];
+                if (numInputs >= numElementsInArray (inputChans))
+                    break;
+            }
+        }
+
+        for (int i = 0; i < totalNumOutputChannels; ++i)
+        {
+            if (outputChannelData[i] != nullptr)
+            {
+                outputChans [numOutputs++] = outputChannelData[i];
+                if (numOutputs >= numElementsInArray (outputChans))
+                    break;
+            }
+        }
+
+        if (numInputs > numOutputs)
+        {
+            // if there aren't enough output channels for the number of
+            // inputs, we need to create some temporary extra ones (can't
+            // use the input data in case it gets written to)
+            tempBuffer.setSize (numInputs - numOutputs, numSamples,
+                                false, false, true);
+
+            for (int i = 0; i < numOutputs; ++i)
+            {
+                channels[numActiveChans] = outputChans[i];
+                memcpy (channels[numActiveChans], inputChans[i], sizeof (float) * (size_t) numSamples);
+                ++numActiveChans;
+            }
+
+            for (int i = numOutputs; i < numInputs; ++i)
+            {
+                channels[numActiveChans] = tempBuffer.getSampleData (i - numOutputs, 0);
+                memcpy (channels[numActiveChans], inputChans[i], sizeof (float) * (size_t) numSamples);
+                ++numActiveChans;
+            }
+        }
+        else
+        {
+            for (int i = 0; i < numInputs; ++i)
+            {
+                channels[numActiveChans] = outputChans[i];
+                memcpy (channels[numActiveChans], inputChans[i], sizeof (float) * (size_t) numSamples);
+                ++numActiveChans;
+            }
+
+            for (int i = numInputs; i < numOutputs; ++i)
+            {
+                channels[numActiveChans] = outputChans[i];
+                zeromem (channels[numActiveChans], sizeof (float) * (size_t) numSamples);
+                ++numActiveChans;
+            }
+        }
+
+        AudioSampleBuffer buffer (channels, numActiveChans, numSamples);
+
+        AudioSourceChannelInfo info (&buffer, 0, numSamples);
+        source->getNextAudioBlock (info);
+
+        for (int i = info.buffer->getNumChannels(); --i >= 0;)
+            buffer.applyGainRamp (i, info.startSample, info.numSamples, lastGain, gain);
+
+        lastGain = gain;
+    }
+    else
+    {
+        for (int i = 0; i < totalNumOutputChannels; ++i)
+            if (outputChannelData[i] != nullptr)
+                zeromem (outputChannelData[i], sizeof (float) * (size_t) numSamples);
+    }
+}
+
+void AudioSourcePlayer::audioDeviceAboutToStart (AudioIODevice* device)
+{
+    prepareToPlay (device->getCurrentSampleRate(),
+                   device->getCurrentBufferSizeSamples());
+}
+
+void AudioSourcePlayer::prepareToPlay (double newSampleRate, int newBufferSize)
+{
+    sampleRate = newSampleRate;
+    bufferSize = newBufferSize;
+    zeromem (channels, sizeof (channels));
+
+    if (source != nullptr)
+        source->prepareToPlay (bufferSize, sampleRate);
+}
+
+void AudioSourcePlayer::audioDeviceStopped()
+{
+    if (source != nullptr)
+        source->releaseResources();
+
+    sampleRate = 0.0;
+    bufferSize = 0;
+
+    tempBuffer.setSize (2, 8);
+}

source/modules/juce_audio_devices/sources/juce_AudioSourcePlayer.h

@@ -0,0 +1,116 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#ifndef JUCE_AUDIOSOURCEPLAYER_H_INCLUDED
+#define JUCE_AUDIOSOURCEPLAYER_H_INCLUDED
+
+
+//==============================================================================
+/**
+    Wrapper class to continuously stream audio from an audio source to an
+    AudioIODevice.
+
+    This object acts as an AudioIODeviceCallback, so can be attached to an
+    output device, and will stream audio from an AudioSource.
+*/
+class JUCE_API  AudioSourcePlayer  : public AudioIODeviceCallback
+{
+public:
+    //==============================================================================
+    /** Creates an empty AudioSourcePlayer. */
+    AudioSourcePlayer();
+
+    /** Destructor.
+
+        Make sure this object isn't still being used by an AudioIODevice before
+        deleting it!
+    */
+    virtual ~AudioSourcePlayer();
+
+    //==============================================================================
+    /** Changes the current audio source to play from.
+
+        If the source passed in is already being used, this method will do nothing.
+        If the source is not null, its prepareToPlay() method will be called
+        before it starts being used for playback.
+
+        If there's another source currently playing, its releaseResources() method
+        will be called after it has been swapped for the new one.
+
+        @param newSource                the new source to use - this will NOT be deleted
+                                        by this object when no longer needed, so it's the
+                                        caller's responsibility to manage it.
+    */
+    void setSource (AudioSource* newSource);
+
+    /** Returns the source that's playing.
+
+        May return 0 if there's no source.
+    */
+    AudioSource* getCurrentSource() const noexcept      { return source; }
+
+    /** Sets a gain to apply to the audio data.
+        @see getGain
+    */
+    void setGain (float newGain) noexcept;
+
+    /** Returns the current gain.
+        @see setGain
+    */
+    float getGain() const noexcept                      { return gain; }
+
+    //==============================================================================
+    /** Implementation of the AudioIODeviceCallback method. */
+    void audioDeviceIOCallback (const float** inputChannelData,
+                                int totalNumInputChannels,
+                                float** outputChannelData,
+                                int totalNumOutputChannels,
+                                int numSamples) override;
+
+    /** Implementation of the AudioIODeviceCallback method. */
+    void audioDeviceAboutToStart (AudioIODevice* device) override;
+
+    /** Implementation of the AudioIODeviceCallback method. */
+    void audioDeviceStopped() override;
+
+    /** An alternative method for initialising the source without an AudioIODevice. */
+    void prepareToPlay (double sampleRate, int blockSize);
+
+private:
+    //==============================================================================
+    CriticalSection readLock;
+    AudioSource* source;
+    double sampleRate;
+    int bufferSize;
+    float* channels [128];
+    float* outputChans [128];
+    const float* inputChans [128];
+    AudioSampleBuffer tempBuffer;
+    float lastGain, gain;
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AudioSourcePlayer)
+};
+
+
+#endif   // JUCE_AUDIOSOURCEPLAYER_H_INCLUDED

source/modules/juce_audio_devices/sources/juce_AudioTransportSource.cpp

@@ -0,0 +1,300 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+AudioTransportSource::AudioTransportSource()
+    : source (nullptr),
+      resamplerSource (nullptr),
+      bufferingSource (nullptr),
+      positionableSource (nullptr),
+      masterSource (nullptr),
+      gain (1.0f),
+      lastGain (1.0f),
+      playing (false),
+      stopped (true),
+      sampleRate (44100.0),
+      sourceSampleRate (0.0),
+      blockSize (128),
+      readAheadBufferSize (0),
+      isPrepared (false),
+      inputStreamEOF (false)
+{
+}
+
+AudioTransportSource::~AudioTransportSource()
+{
+    setSource (nullptr);
+
+    releaseMasterResources();
+}
+
+void AudioTransportSource::setSource (PositionableAudioSource* const newSource,
+                                      int readAheadBufferSize_,
+                                      TimeSliceThread* readAheadThread,
+                                      double sourceSampleRateToCorrectFor,
+                                      int maxNumChannels)
+{
+    if (source == newSource)
+    {
+        if (source == nullptr)
+            return;
+
+        setSource (nullptr, 0, nullptr); // deselect and reselect to avoid releasing resources wrongly
+    }
+
+    readAheadBufferSize = readAheadBufferSize_;
+    sourceSampleRate = sourceSampleRateToCorrectFor;
+
+    ResamplingAudioSource* newResamplerSource = nullptr;
+    BufferingAudioSource* newBufferingSource = nullptr;
+    PositionableAudioSource* newPositionableSource = nullptr;
+    AudioSource* newMasterSource = nullptr;
+
+    ScopedPointer <ResamplingAudioSource> oldResamplerSource (resamplerSource);
+    ScopedPointer <BufferingAudioSource> oldBufferingSource (bufferingSource);
+    AudioSource* oldMasterSource = masterSource;
+
+    if (newSource != nullptr)
+    {
+        newPositionableSource = newSource;
+
+        if (readAheadBufferSize_ > 0)
+        {
+            // If you want to use a read-ahead buffer, you must also provide a TimeSliceThread
+            // for it to use!
+            jassert (readAheadThread != nullptr);
+
+            newPositionableSource = newBufferingSource
+                = new BufferingAudioSource (newPositionableSource, *readAheadThread,
+                                            false, readAheadBufferSize_, maxNumChannels);
+        }
+
+        newPositionableSource->setNextReadPosition (0);
+
+        if (sourceSampleRateToCorrectFor > 0)
+            newMasterSource = newResamplerSource
+                = new ResamplingAudioSource (newPositionableSource, false, maxNumChannels);
+        else
+            newMasterSource = newPositionableSource;
+
+        if (isPrepared)
+        {
+            if (newResamplerSource != nullptr && sourceSampleRate > 0 && sampleRate > 0)
+                newResamplerSource->setResamplingRatio (sourceSampleRate / sampleRate);
+
+            newMasterSource->prepareToPlay (blockSize, sampleRate);
+        }
+    }
+
+    {
+        const ScopedLock sl (callbackLock);
+
+        source = newSource;
+        resamplerSource = newResamplerSource;
+        bufferingSource = newBufferingSource;
+        masterSource = newMasterSource;
+        positionableSource = newPositionableSource;
+
+        playing = false;
+    }
+
+    if (oldMasterSource != nullptr)
+        oldMasterSource->releaseResources();
+}
+
+void AudioTransportSource::start()
+{
+    if ((! playing) && masterSource != nullptr)
+    {
+        {
+            const ScopedLock sl (callbackLock);
+            playing = true;
+            stopped = false;
+            inputStreamEOF = false;
+        }
+
+        sendChangeMessage();
+    }
+}
+
+void AudioTransportSource::stop()
+{
+    if (playing)
+    {
+        {
+            const ScopedLock sl (callbackLock);
+            playing = false;
+        }
+
+        int n = 500;
+        while (--n >= 0 && ! stopped)
+            Thread::sleep (2);
+
+        sendChangeMessage();
+    }
+}
+
+void AudioTransportSource::setPosition (double newPosition)
+{
+    if (sampleRate > 0.0)
+        setNextReadPosition ((int64) (newPosition * sampleRate));
+}
+
+double AudioTransportSource::getCurrentPosition() const
+{
+    if (sampleRate > 0.0)
+        return getNextReadPosition() / sampleRate;
+
+    return 0.0;
+}
+
+double AudioTransportSource::getLengthInSeconds() const
+{
+    return getTotalLength() / sampleRate;
+}
+
+void AudioTransportSource::setNextReadPosition (int64 newPosition)
+{
+    if (positionableSource != nullptr)
+    {
+        if (sampleRate > 0 && sourceSampleRate > 0)
+            newPosition = (int64) (newPosition * sourceSampleRate / sampleRate);
+
+        positionableSource->setNextReadPosition (newPosition);
+    }
+}
+
+int64 AudioTransportSource::getNextReadPosition() const
+{
+    if (positionableSource != nullptr)
+    {
+        const double ratio = (sampleRate > 0 && sourceSampleRate > 0) ? sampleRate / sourceSampleRate : 1.0;
+
+        return (int64) (positionableSource->getNextReadPosition() * ratio);
+    }
+
+    return 0;
+}
+
+int64 AudioTransportSource::getTotalLength() const
+{
+    const ScopedLock sl (callbackLock);
+
+    if (positionableSource != nullptr)
+    {
+        const double ratio = (sampleRate > 0 && sourceSampleRate > 0) ? sampleRate / sourceSampleRate : 1.0;
+
+        return (int64) (positionableSource->getTotalLength() * ratio);
+    }
+
+    return 0;
+}
+
+bool AudioTransportSource::isLooping() const
+{
+    const ScopedLock sl (callbackLock);
+
+    return positionableSource != nullptr
+            && positionableSource->isLooping();
+}
+
+void AudioTransportSource::setGain (const float newGain) noexcept
+{
+    gain = newGain;
+}
+
+void AudioTransportSource::prepareToPlay (int samplesPerBlockExpected, double newSampleRate)
+{
+    const ScopedLock sl (callbackLock);
+
+    sampleRate = newSampleRate;
+    blockSize = samplesPerBlockExpected;
+
+    if (masterSource != nullptr)
+        masterSource->prepareToPlay (samplesPerBlockExpected, sampleRate);
+
+    if (resamplerSource != nullptr && sourceSampleRate > 0)
+        resamplerSource->setResamplingRatio (sourceSampleRate / sampleRate);
+
+    isPrepared = true;
+}
+
+void AudioTransportSource::releaseMasterResources()
+{
+    const ScopedLock sl (callbackLock);
+
+    if (masterSource != nullptr)
+        masterSource->releaseResources();
+
+    isPrepared = false;
+}
+
+void AudioTransportSource::releaseResources()
+{
+    releaseMasterResources();
+}
+
+void AudioTransportSource::getNextAudioBlock (const AudioSourceChannelInfo& info)
+{
+    const ScopedLock sl (callbackLock);
+
+    inputStreamEOF = false;
+
+    if (masterSource != nullptr && ! stopped)
+    {
+        masterSource->getNextAudioBlock (info);
+
+        if (! playing)
+        {
+            // just stopped playing, so fade out the last block..
+            for (int i = info.buffer->getNumChannels(); --i >= 0;)
+                info.buffer->applyGainRamp (i, info.startSample, jmin (256, info.numSamples), 1.0f, 0.0f);
+
+            if (info.numSamples > 256)
+                info.buffer->clear (info.startSample + 256, info.numSamples - 256);
+        }
+
+        if (positionableSource->getNextReadPosition() > positionableSource->getTotalLength() + 1
+             && ! positionableSource->isLooping())
+        {
+            playing = false;
+            inputStreamEOF = true;
+            sendChangeMessage();
+        }
+
+        stopped = ! playing;
+
+        for (int i = info.buffer->getNumChannels(); --i >= 0;)
+        {
+            info.buffer->applyGainRamp (i, info.startSample, info.numSamples,
+                                        lastGain, gain);
+        }
+    }
+    else
+    {
+        info.clearActiveBufferRegion();
+        stopped = true;
+    }
+
+    lastGain = gain;
+}

source/modules/juce_audio_devices/sources/juce_AudioTransportSource.h

@@ -0,0 +1,186 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#ifndef JUCE_AUDIOTRANSPORTSOURCE_H_INCLUDED
+#define JUCE_AUDIOTRANSPORTSOURCE_H_INCLUDED
+
+
+//==============================================================================
+/**
+    An AudioSource that takes a PositionableAudioSource and allows it to be
+    played, stopped, started, etc.
+
+    This can also be told use a buffer and background thread to read ahead, and
+    if can correct for different sample-rates.
+
+    You may want to use one of these along with an AudioSourcePlayer and AudioIODevice
+    to control playback of an audio file.
+
+    @see AudioSource, AudioSourcePlayer
+*/
+class JUCE_API  AudioTransportSource  : public PositionableAudioSource,
+                                        public ChangeBroadcaster
+{
+public:
+    //==============================================================================
+    /** Creates an AudioTransportSource.
+
+        After creating one of these, use the setSource() method to select an input source.
+    */
+    AudioTransportSource();
+
+    /** Destructor. */
+    ~AudioTransportSource();
+
+    //==============================================================================
+    /** Sets the reader that is being used as the input source.
+
+        This will stop playback, reset the position to 0 and change to the new reader.
+
+        The source passed in will not be deleted by this object, so must be managed by
+        the caller.
+
+        @param newSource                        the new input source to use. This may be zero
+        @param readAheadBufferSize              a size of buffer to use for reading ahead. If this
+                                                is zero, no reading ahead will be done; if it's
+                                                greater than zero, a BufferingAudioSource will be used
+                                                to do the reading-ahead. If you set a non-zero value here,
+                                                you'll also need to set the readAheadThread parameter.
+        @param readAheadThread                  if you set readAheadBufferSize to a non-zero value, then
+                                                you'll also need to supply this TimeSliceThread object for
+                                                the background reader to use. The thread object must not be
+                                                deleted while the AudioTransport source is still using it.
+        @param sourceSampleRateToCorrectFor     if this is non-zero, it specifies the sample
+                                                rate of the source, and playback will be sample-rate
+                                                adjusted to maintain playback at the correct pitch. If
+                                                this is 0, no sample-rate adjustment will be performed
+        @param maxNumChannels                   the maximum number of channels that may need to be played
+    */
+    void setSource (PositionableAudioSource* newSource,
+                    int readAheadBufferSize = 0,
+                    TimeSliceThread* readAheadThread = nullptr,
+                    double sourceSampleRateToCorrectFor = 0.0,
+                    int maxNumChannels = 2);
+
+    //==============================================================================
+    /** Changes the current playback position in the source stream.
+
+        The next time the getNextAudioBlock() method is called, this
+        is the time from which it'll read data.
+
+        @see getPosition
+    */
+    void setPosition (double newPosition);
+
+    /** Returns the position that the next data block will be read from
+
+        This is a time in seconds.
+    */
+    double getCurrentPosition() const;
+
+    /** Returns the stream's length in seconds. */
+    double getLengthInSeconds() const;
+
+    /** Returns true if the player has stopped because its input stream ran out of data.
+    */
+    bool hasStreamFinished() const noexcept             { return inputStreamEOF; }
+
+    //==============================================================================
+    /** Starts playing (if a source has been selected).
+
+        If it starts playing, this will send a message to any ChangeListeners
+        that are registered with this object.
+    */
+    void start();
+
+    /** Stops playing.
+
+        If it's actually playing, this will send a message to any ChangeListeners
+        that are registered with this object.
+    */
+    void stop();
+
+    /** Returns true if it's currently playing. */
+    bool isPlaying() const noexcept     { return playing; }
+
+    //==============================================================================
+    /** Changes the gain to apply to the output.
+
+        @param newGain  a factor by which to multiply the outgoing samples,
+                        so 1.0 = 0dB, 0.5 = -6dB, 2.0 = 6dB, etc.
+    */
+    void setGain (float newGain) noexcept;
+
+    /** Returns the current gain setting.
+
+        @see setGain
+    */
+    float getGain() const noexcept      { return gain; }
+
+
+    //==============================================================================
+    /** Implementation of the AudioSource method. */
+    void prepareToPlay (int samplesPerBlockExpected, double sampleRate) override;
+
+    /** Implementation of the AudioSource method. */
+    void releaseResources() override;
+
+    /** Implementation of the AudioSource method. */
+    void getNextAudioBlock (const AudioSourceChannelInfo&) override;
+
+    //==============================================================================
+    /** Implements the PositionableAudioSource method. */
+    void setNextReadPosition (int64 newPosition) override;
+
+    /** Implements the PositionableAudioSource method. */
+    int64 getNextReadPosition() const override;
+
+    /** Implements the PositionableAudioSource method. */
+    int64 getTotalLength() const override;
+
+    /** Implements the PositionableAudioSource method. */
+    bool isLooping() const override;
+
+private:
+    //==============================================================================
+    PositionableAudioSource* source;
+    ResamplingAudioSource* resamplerSource;
+    BufferingAudioSource* bufferingSource;
+    PositionableAudioSource* positionableSource;
+    AudioSource* masterSource;
+
+    CriticalSection callbackLock;
+    float volatile gain, lastGain;
+    bool volatile playing, stopped;
+    double sampleRate, sourceSampleRate;
+    int blockSize, readAheadBufferSize;
+    bool isPrepared, inputStreamEOF;
+
+    void releaseMasterResources();
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AudioTransportSource)
+};
+
+
+#endif   // JUCE_AUDIOTRANSPORTSOURCE_H_INCLUDED

source/modules/juce_audio_formats.h

@@ -0,0 +1,26 @@
+/*
+ * Carla Juce setup
+ * Copyright (C) 2013 Filipe Coelho <falktx@falktx.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as
+ * published by the Free Software Foundation; either version 2 of
+ * the License, or any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * For a full copy of the GNU General Public License see the doc/GPL.txt file.
+ */
+
+#ifndef CARLA_JUCE_AUDIO_FORMATS_H_INCLUDED
+#define CARLA_JUCE_AUDIO_FORMATS_H_INCLUDED
+
+#include "juce_audio_basics.h"
+
+#include "juce_audio_formats/AppConfig.h"
+#include "juce_audio_formats/juce_audio_formats.h"
+
+#endif // CARLA_JUCE_AUDIO_FORMATS_H_INCLUDED

source/modules/juce_audio_formats/AppConfig.h

@@ -0,0 +1,53 @@
+/*
+  ==============================================================================
+
+   Build options for juce_audio_formats static library
+
+  ==============================================================================
+*/
+
+#ifndef CARLA_JUCE_AUDIO_FORMATS_APPCONFIG_H_INCLUDED
+#define CARLA_JUCE_AUDIO_FORMATS_APPCONFIG_H_INCLUDED
+
+#include "../juce_audio_basics/AppConfig.h"
+
+//=============================================================================
+/** Config: JUCE_USE_FLAC
+    Enables the FLAC audio codec classes (available on all platforms).
+    If your app doesn't need to read FLAC files, you might want to disable this to
+    reduce the size of your codebase and build time.
+*/
+#define JUCE_USE_FLAC 0
+
+/** Config: JUCE_USE_OGGVORBIS
+    Enables the Ogg-Vorbis audio codec classes (available on all platforms).
+    If your app doesn't need to read Ogg-Vorbis files, you might want to disable this to
+    reduce the size of your codebase and build time.
+*/
+#define JUCE_USE_OGGVORBIS 0
+
+/** Config: JUCE_USE_MP3AUDIOFORMAT
+    Enables the software-based MP3AudioFormat class.
+    IMPORTANT DISCLAIMER: By choosing to enable the JUCE_USE_MP3AUDIOFORMAT flag and to compile
+    this MP3 code into your software, you do so AT YOUR OWN RISK! By doing so, you are agreeing
+    that Raw Material Software is in no way responsible for any patent, copyright, or other
+    legal issues that you may suffer as a result.
+
+    The code in juce_MP3AudioFormat.cpp is NOT guaranteed to be free from infringements of 3rd-party
+    intellectual property. If you wish to use it, please seek your own independent advice about the
+    legality of doing so. If you are not willing to accept full responsibility for the consequences
+    of using this code, then do not enable this setting.
+*/
+#define JUCE_USE_MP3AUDIOFORMAT 0
+
+/** Config: JUCE_USE_LAME_AUDIO_FORMAT
+    Enables the LameEncoderAudioFormat class.
+*/
+#define JUCE_USE_LAME_AUDIO_FORMAT 0
+
+/** Config: JUCE_USE_WINDOWS_MEDIA_FORMAT
+    Enables the Windows Media SDK codecs.
+*/
+#define JUCE_USE_WINDOWS_MEDIA_FORMAT 0
+
+#endif // CARLA_JUCE_AUDIO_FORMATS_APPCONFIG_H_INCLUDED

source/modules/juce_audio_formats/Makefile

@@ -0,0 +1,88 @@
+#!/usr/bin/make -f
+# Makefile for juce_audio_formats #
+# ------------------------------- #
+# Created by falkTX
+#
+
+include ../../Makefile.mk
+
+# --------------------------------------------------------------
+
+BUILD_CXX_FLAGS += -I.
+
+ifeq ($(MACOS),true)
+BUILD_CXX_FLAGS += -objc++
+OBJS         = juce_audio_formats.mm.o
+OBJS_posix32 = juce_audio_formats.mm.posix32.o
+OBJS_posix64 = juce_audio_formats.mm.posix64.o
+else
+OBJS         = juce_audio_formats.cpp.o
+OBJS_posix32 = juce_audio_formats.cpp.posix32.o
+OBJS_posix64 = juce_audio_formats.cpp.posix64.o
+endif
+
+OBJS_win32 = juce_audio_formats.cpp.win32.o
+OBJS_win64 = juce_audio_formats.cpp.win64.o
+
+# --------------------------------------------------------------
+
+all: ../juce_audio_formats.a
+
+posix32: ../juce_audio_formats.posix32.a
+posix64: ../juce_audio_formats.posix64.a
+win32:   ../juce_audio_formats.win32.a
+win64:   ../juce_audio_formats.win64.a
+
+# --------------------------------------------------------------
+
+../juce_audio_formats.a: $(OBJS)
+	$(AR) rs $@ $^
+
+../juce_audio_formats.posix32.a: $(OBJS_posix32)
+	$(AR) rs $@ $^
+
+../juce_audio_formats.posix64.a: $(OBJS_posix64)
+	$(AR) rs $@ $^
+
+../juce_audio_formats.win32.a: $(OBJS_win32)
+	$(AR) rs $@ $^
+
+../juce_audio_formats.win64.a: $(OBJS_win64)
+	$(AR) rs $@ $^
+
+../juce_audio_formats.dll: $(OBJS)
+	$(CXX) $^ -shared $(LINK_FLAGS) -o $@
+
+../juce_audio_formats.dylib: $(OBJS)
+	$(CXX) $^ -dynamiclib $(LINK_FLAGS) -o $@
+
+../juce_audio_formats.so: $(OBJS)
+	$(CXX) $^ -shared $(LINK_FLAGS) -o $@
+
+# --------------------------------------------------------------
+
+%.cpp.o: %.cpp
+	$(CXX) $< $(BUILD_CXX_FLAGS) -c -o $@
+
+%.mm.o: %.mm
+	$(CXX) $< $(BUILD_CXX_FLAGS) -c -o $@
+
+%.posix32.o: %
+	$(CXX) $< $(BUILD_CXX_FLAGS) $(32BIT_FLAGS) -c -o $@
+
+%.posix64.o: %
+	$(CXX) $< $(BUILD_CXX_FLAGS) $(64BIT_FLAGS) -c -o $@
+
+%.win32.o: %
+	$(CXX) $< $(BUILD_CXX_FLAGS) $(32BIT_FLAGS) -c -o $@
+
+%.win64.o: %
+	$(CXX) $< $(BUILD_CXX_FLAGS) $(64BIT_FLAGS) -c -o $@
+
+# --------------------------------------------------------------
+
+clean:
+	rm -f *.o ../juce_audio_formats*.a
+
+debug:
+	$(MAKE) DEBUG=true

source/modules/juce_audio_formats/codecs/flac/Flac Licence.txt

@@ -0,0 +1,49 @@
+
+=====================================================================
+
+I've incorporated FLAC directly into the Juce codebase because it makes
+things much easier than having to make all your builds link correctly to
+the appropriate libraries on every different platform.
+
+I've made minimal changes to the FLAC code - just tweaked a few include paths 
+to make it build smoothly, added some headers to allow you to turn off FLAC 
+compilation, and commented-out a couple of unused bits of code.
+
+=====================================================================
+
+
+The following license is the BSD-style license that comes with the
+Flac distribution, and which applies just to the files I've
+included in this directory. For more info, and to get the rest of the
+distribution, visit the Flac homepage: flac.sourceforge.net
+
+=====================================================================
+
+Copyright (C) 2000,2001,2002,2003,2004,2005,2006 Josh Coalson
+
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+
+- Redistributions of source code must retain the above copyright
+notice, this list of conditions and the following disclaimer.
+
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+
+- Neither the name of the Xiph.org Foundation nor the names of its
+contributors may be used to endorse or promote products derived from
+this software without specific prior written permission.
+
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.

source/modules/juce_audio_formats/codecs/flac/all.h

@@ -0,0 +1,402 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__ALL_H
+#define FLAC__ALL_H
+
+#include "export.h"
+
+#include "assert.h"
+#include "callback.h"
+#include "format.h"
+#include "metadata.h"
+#include "ordinals.h"
+#include "stream_decoder.h"
+#include "stream_encoder.h"
+
+
+#ifdef _MSC_VER
+/* OPT: an MSVC built-in would be better */
+static _inline FLAC__uint32 local_swap32_(FLAC__uint32 x)
+{
+	x = ((x<<8)&0xFF00FF00) | ((x>>8)&0x00FF00FF);
+	return (x>>16) | (x<<16);
+}
+#endif
+
+#if defined(_MSC_VER) && defined(_X86_)
+/* OPT: an MSVC built-in would be better */
+static void local_swap32_block_(FLAC__uint32 *start, FLAC__uint32 len)
+{
+	__asm {
+		mov edx, start
+		mov ecx, len
+		test ecx, ecx
+loop1:
+		jz done1
+		mov eax, [edx]
+		bswap eax
+		mov [edx], eax
+		add edx, 4
+		dec ecx
+		jmp short loop1
+done1:
+	}
+}
+#endif
+
+
+/** \mainpage
+ *
+ * \section intro Introduction
+ *
+ * This is the documentation for the FLAC C and C++ APIs.  It is
+ * highly interconnected; this introduction should give you a top
+ * level idea of the structure and how to find the information you
+ * need.  As a prerequisite you should have at least a basic
+ * knowledge of the FLAC format, documented
+ * <A HREF="../format.html">here</A>.
+ *
+ * \section c_api FLAC C API
+ *
+ * The FLAC C API is the interface to libFLAC, a set of structures
+ * describing the components of FLAC streams, and functions for
+ * encoding and decoding streams, as well as manipulating FLAC
+ * metadata in files.  The public include files will be installed
+ * in your include area (for example /usr/include/FLAC/...).
+ *
+ * By writing a little code and linking against libFLAC, it is
+ * relatively easy to add FLAC support to another program.  The
+ * library is licensed under <A HREF="../license.html">Xiph's BSD license</A>.
+ * Complete source code of libFLAC as well as the command-line
+ * encoder and plugins is available and is a useful source of
+ * examples.
+ *
+ * Aside from encoders and decoders, libFLAC provides a powerful
+ * metadata interface for manipulating metadata in FLAC files.  It
+ * allows the user to add, delete, and modify FLAC metadata blocks
+ * and it can automatically take advantage of PADDING blocks to avoid
+ * rewriting the entire FLAC file when changing the size of the
+ * metadata.
+ *
+ * libFLAC usually only requires the standard C library and C math
+ * library. In particular, threading is not used so there is no
+ * dependency on a thread library. However, libFLAC does not use
+ * global variables and should be thread-safe.
+ *
+ * libFLAC also supports encoding to and decoding from Ogg FLAC.
+ * However the metadata editing interfaces currently have limited
+ * read-only support for Ogg FLAC files.
+ *
+ * \section cpp_api FLAC C++ API
+ *
+ * The FLAC C++ API is a set of classes that encapsulate the
+ * structures and functions in libFLAC.  They provide slightly more
+ * functionality with respect to metadata but are otherwise
+ * equivalent.  For the most part, they share the same usage as
+ * their counterparts in libFLAC, and the FLAC C API documentation
+ * can be used as a supplement.  The public include files
+ * for the C++ API will be installed in your include area (for
+ * example /usr/include/FLAC++/...).
+ *
+ * libFLAC++ is also licensed under
+ * <A HREF="../license.html">Xiph's BSD license</A>.
+ *
+ * \section getting_started Getting Started
+ *
+ * A good starting point for learning the API is to browse through
+ * the <A HREF="modules.html">modules</A>.  Modules are logical
+ * groupings of related functions or classes, which correspond roughly
+ * to header files or sections of header files.  Each module includes a
+ * detailed description of the general usage of its functions or
+ * classes.
+ *
+ * From there you can go on to look at the documentation of
+ * individual functions.  You can see different views of the individual
+ * functions through the links in top bar across this page.
+ *
+ * If you prefer a more hands-on approach, you can jump right to some
+ * <A HREF="../documentation_example_code.html">example code</A>.
+ *
+ * \section porting_guide Porting Guide
+ *
+ * Starting with FLAC 1.1.3 a \link porting Porting Guide \endlink
+ * has been introduced which gives detailed instructions on how to
+ * port your code to newer versions of FLAC.
+ *
+ * \section embedded_developers Embedded Developers
+ *
+ * libFLAC has grown larger over time as more functionality has been
+ * included, but much of it may be unnecessary for a particular embedded
+ * implementation.  Unused parts may be pruned by some simple editing of
+ * src/libFLAC/Makefile.am.  In general, the decoders, encoders, and
+ * metadata interface are all independent from each other.
+ *
+ * It is easiest to just describe the dependencies:
+ *
+ * - All modules depend on the \link flac_format Format \endlink module.
+ * - The decoders and encoders depend on the bitbuffer.
+ * - The decoder is independent of the encoder.  The encoder uses the
+ *   decoder because of the verify feature, but this can be removed if
+ *   not needed.
+ * - Parts of the metadata interface require the stream decoder (but not
+ *   the encoder).
+ * - Ogg support is selectable through the compile time macro
+ *   \c FLAC__HAS_OGG.
+ *
+ * For example, if your application only requires the stream decoder, no
+ * encoder, and no metadata interface, you can remove the stream encoder
+ * and the metadata interface, which will greatly reduce the size of the
+ * library.
+ *
+ * Also, there are several places in the libFLAC code with comments marked
+ * with "OPT:" where a #define can be changed to enable code that might be
+ * faster on a specific platform.  Experimenting with these can yield faster
+ * binaries.
+ */
+
+/** \defgroup porting Porting Guide for New Versions
+ *
+ * This module describes differences in the library interfaces from
+ * version to version.  It assists in the porting of code that uses
+ * the libraries to newer versions of FLAC.
+ *
+ * One simple facility for making porting easier that has been added
+ * in FLAC 1.1.3 is a set of \c #defines in \c export.h of each
+ * library's includes (e.g. \c include/FLAC/export.h).  The
+ * \c #defines mirror the libraries'
+ * <A HREF="http://www.gnu.org/software/libtool/manual.html#Libtool-versioning">libtool version numbers</A>,
+ * e.g. in libFLAC there are \c FLAC_API_VERSION_CURRENT,
+ * \c FLAC_API_VERSION_REVISION, and \c FLAC_API_VERSION_AGE.
+ * These can be used to support multiple versions of an API during the
+ * transition phase, e.g.
+ *
+ * \code
+ * #if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7
+ *   legacy code
+ * #else
+ *   new code
+ * #endif
+ * \endcode
+ *
+ * The the source will work for multiple versions and the legacy code can
+ * easily be removed when the transition is complete.
+ *
+ * Another available symbol is FLAC_API_SUPPORTS_OGG_FLAC (defined in
+ * include/FLAC/export.h), which can be used to determine whether or not
+ * the library has been compiled with support for Ogg FLAC.  This is
+ * simpler than trying to call an Ogg init function and catching the
+ * error.
+ */
+
+/** \defgroup porting_1_1_2_to_1_1_3 Porting from FLAC 1.1.2 to 1.1.3
+ *  \ingroup porting
+ *
+ *  \brief
+ *  This module describes porting from FLAC 1.1.2 to FLAC 1.1.3.
+ *
+ * The main change between the APIs in 1.1.2 and 1.1.3 is that they have
+ * been simplified.  First, libOggFLAC has been merged into libFLAC and
+ * libOggFLAC++ has been merged into libFLAC++.  Second, both the three
+ * decoding layers and three encoding layers have been merged into a
+ * single stream decoder and stream encoder.  That is, the functionality
+ * of FLAC__SeekableStreamDecoder and FLAC__FileDecoder has been merged
+ * into FLAC__StreamDecoder, and FLAC__SeekableStreamEncoder and
+ * FLAC__FileEncoder into FLAC__StreamEncoder.  Only the
+ * FLAC__StreamDecoder and FLAC__StreamEncoder remain.  What this means
+ * is there is now a single API that can be used to encode or decode
+ * streams to/from native FLAC or Ogg FLAC and the single API can work
+ * on both seekable and non-seekable streams.
+ *
+ * Instead of creating an encoder or decoder of a certain layer, now the
+ * client will always create a FLAC__StreamEncoder or
+ * FLAC__StreamDecoder.  The old layers are now differentiated by the
+ * initialization function.  For example, for the decoder,
+ * FLAC__stream_decoder_init() has been replaced by
+ * FLAC__stream_decoder_init_stream().  This init function takes
+ * callbacks for the I/O, and the seeking callbacks are optional.  This
+ * allows the client to use the same object for seekable and
+ * non-seekable streams.  For decoding a FLAC file directly, the client
+ * can use FLAC__stream_decoder_init_file() and pass just a filename
+ * and fewer callbacks; most of the other callbacks are supplied
+ * internally.  For situations where fopen()ing by filename is not
+ * possible (e.g. Unicode filenames on Windows) the client can instead
+ * open the file itself and supply the FILE* to
+ * FLAC__stream_decoder_init_FILE().  The init functions now returns a
+ * FLAC__StreamDecoderInitStatus instead of FLAC__StreamDecoderState.
+ * Since the callbacks and client data are now passed to the init
+ * function, the FLAC__stream_decoder_set_*_callback() functions and
+ * FLAC__stream_decoder_set_client_data() are no longer needed.  The
+ * rest of the calls to the decoder are the same as before.
+ *
+ * There are counterpart init functions for Ogg FLAC, e.g.
+ * FLAC__stream_decoder_init_ogg_stream().  All the rest of the calls
+ * and callbacks are the same as for native FLAC.
+ *
+ * As an example, in FLAC 1.1.2 a seekable stream decoder would have
+ * been set up like so:
+ *
+ * \code
+ * FLAC__SeekableStreamDecoder *decoder = FLAC__seekable_stream_decoder_new();
+ * if(decoder == NULL) do_something;
+ * FLAC__seekable_stream_decoder_set_md5_checking(decoder, true);
+ * [... other settings ...]
+ * FLAC__seekable_stream_decoder_set_read_callback(decoder, my_read_callback);
+ * FLAC__seekable_stream_decoder_set_seek_callback(decoder, my_seek_callback);
+ * FLAC__seekable_stream_decoder_set_tell_callback(decoder, my_tell_callback);
+ * FLAC__seekable_stream_decoder_set_length_callback(decoder, my_length_callback);
+ * FLAC__seekable_stream_decoder_set_eof_callback(decoder, my_eof_callback);
+ * FLAC__seekable_stream_decoder_set_write_callback(decoder, my_write_callback);
+ * FLAC__seekable_stream_decoder_set_metadata_callback(decoder, my_metadata_callback);
+ * FLAC__seekable_stream_decoder_set_error_callback(decoder, my_error_callback);
+ * FLAC__seekable_stream_decoder_set_client_data(decoder, my_client_data);
+ * if(FLAC__seekable_stream_decoder_init(decoder) != FLAC__SEEKABLE_STREAM_DECODER_OK) do_something;
+ * \endcode
+ *
+ * In FLAC 1.1.3 it is like this:
+ *
+ * \code
+ * FLAC__StreamDecoder *decoder = FLAC__stream_decoder_new();
+ * if(decoder == NULL) do_something;
+ * FLAC__stream_decoder_set_md5_checking(decoder, true);
+ * [... other settings ...]
+ * if(FLAC__stream_decoder_init_stream(
+ *   decoder,
+ *   my_read_callback,
+ *   my_seek_callback,      // or NULL
+ *   my_tell_callback,      // or NULL
+ *   my_length_callback,    // or NULL
+ *   my_eof_callback,       // or NULL
+ *   my_write_callback,
+ *   my_metadata_callback,  // or NULL
+ *   my_error_callback,
+ *   my_client_data
+ * ) != FLAC__STREAM_DECODER_INIT_STATUS_OK) do_something;
+ * \endcode
+ *
+ * or you could do;
+ *
+ * \code
+ * [...]
+ * FILE *file = fopen("somefile.flac","rb");
+ * if(file == NULL) do_somthing;
+ * if(FLAC__stream_decoder_init_FILE(
+ *   decoder,
+ *   file,
+ *   my_write_callback,
+ *   my_metadata_callback,  // or NULL
+ *   my_error_callback,
+ *   my_client_data
+ * ) != FLAC__STREAM_DECODER_INIT_STATUS_OK) do_something;
+ * \endcode
+ *
+ * or just:
+ *
+ * \code
+ * [...]
+ * if(FLAC__stream_decoder_init_file(
+ *   decoder,
+ *   "somefile.flac",
+ *   my_write_callback,
+ *   my_metadata_callback,  // or NULL
+ *   my_error_callback,
+ *   my_client_data
+ * ) != FLAC__STREAM_DECODER_INIT_STATUS_OK) do_something;
+ * \endcode
+ *
+ * Another small change to the decoder is in how it handles unparseable
+ * streams.  Before, when the decoder found an unparseable stream
+ * (reserved for when the decoder encounters a stream from a future
+ * encoder that it can't parse), it changed the state to
+ * \c FLAC__STREAM_DECODER_UNPARSEABLE_STREAM.  Now the decoder instead
+ * drops sync and calls the error callback with a new error code
+ * \c FLAC__STREAM_DECODER_ERROR_STATUS_UNPARSEABLE_STREAM.  This is
+ * more robust.  If your error callback does not discriminate on the the
+ * error state, your code does not need to be changed.
+ *
+ * The encoder now has a new setting:
+ * FLAC__stream_encoder_set_apodization().  This is for setting the
+ * method used to window the data before LPC analysis.  You only need to
+ * add a call to this function if the default is not suitable.   There
+ * are also two new convenience functions that may be useful:
+ * FLAC__metadata_object_cuesheet_calculate_cddb_id() and
+ * FLAC__metadata_get_cuesheet().
+ *
+ * The \a bytes parameter to FLAC__StreamDecoderReadCallback,
+ * FLAC__StreamEncoderReadCallback, and FLAC__StreamEncoderWriteCallback
+ * is now \c size_t instead of \c unsigned.
+ */
+
+/** \defgroup porting_1_1_3_to_1_1_4 Porting from FLAC 1.1.3 to 1.1.4
+ *  \ingroup porting
+ *
+ *  \brief
+ *  This module describes porting from FLAC 1.1.3 to FLAC 1.1.4.
+ *
+ * There were no changes to any of the interfaces from 1.1.3 to 1.1.4.
+ * There was a slight change in the implementation of
+ * FLAC__stream_encoder_set_metadata(); the function now makes a copy
+ * of the \a metadata array of pointers so the client no longer needs
+ * to maintain it after the call.  The objects themselves that are
+ * pointed to by the array are still not copied though and must be
+ * maintained until the call to FLAC__stream_encoder_finish().
+ */
+
+/** \defgroup porting_1_1_4_to_1_2_0 Porting from FLAC 1.1.4 to 1.2.0
+ *  \ingroup porting
+ *
+ *  \brief
+ *  This module describes porting from FLAC 1.1.4 to FLAC 1.2.0.
+ *
+ * There were only very minor changes to the interfaces from 1.1.4 to 1.2.0.
+ * In libFLAC, \c FLAC__format_sample_rate_is_subset() was added.
+ * In libFLAC++, \c FLAC::Decoder::Stream::get_decode_position() was added.
+ *
+ * Finally, value of the constant \c FLAC__FRAME_HEADER_RESERVED_LEN
+ * has changed to reflect the conversion of one of the reserved bits
+ * into active use.  It used to be \c 2 and now is \c 1.  However the
+ * FLAC frame header length has not changed, so to skip the proper
+ * number of bits, use \c FLAC__FRAME_HEADER_RESERVED_LEN +
+ * \c FLAC__FRAME_HEADER_BLOCKING_STRATEGY_LEN
+ */
+
+/** \defgroup flac FLAC C API
+ *
+ * The FLAC C API is the interface to libFLAC, a set of structures
+ * describing the components of FLAC streams, and functions for
+ * encoding and decoding streams, as well as manipulating FLAC
+ * metadata in files.
+ *
+ * You should start with the format components as all other modules
+ * are dependent on it.
+ */
+
+#endif

source/modules/juce_audio_formats/codecs/flac/alloc.h

@@ -0,0 +1,212 @@
+/* alloc - Convenience routines for safely allocating memory
+ * Copyright (C) 2007  Josh Coalson
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301  USA
+ */
+
+#ifndef FLAC__SHARE__ALLOC_H
+#define FLAC__SHARE__ALLOC_H
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+/* WATCHOUT: for c++ you may have to #define __STDC_LIMIT_MACROS 1 real early
+ * before #including this file,  otherwise SIZE_MAX might not be defined
+ */
+
+#include <limits.h> /* for SIZE_MAX */
+#if !defined _MSC_VER && !defined __MINGW32__ && !defined __EMX__
+#include <stdint.h> /* for SIZE_MAX in case limits.h didn't get it */
+#endif
+#include <stdlib.h> /* for size_t, malloc(), etc */
+
+#ifndef SIZE_MAX
+# ifndef SIZE_T_MAX
+#  ifdef _MSC_VER
+#   define SIZE_T_MAX UINT_MAX
+#  else
+#   error
+#  endif
+# endif
+# define SIZE_MAX SIZE_T_MAX
+#endif
+
+#ifndef FLaC__INLINE
+#define FLaC__INLINE
+#endif
+
+/* avoid malloc()ing 0 bytes, see:
+ * https://www.securecoding.cert.org/confluence/display/seccode/MEM04-A.+Do+not+make+assumptions+about+the+result+of+allocating+0+bytes?focusedCommentId=5407003
+*/
+static FLaC__INLINE void *safe_malloc_(size_t size)
+{
+	/* malloc(0) is undefined; FLAC src convention is to always allocate */
+	if(!size)
+		size++;
+	return malloc(size);
+}
+
+static FLaC__INLINE void *safe_calloc_(size_t nmemb, size_t size)
+{
+	if(!nmemb || !size)
+		return malloc(1); /* malloc(0) is undefined; FLAC src convention is to always allocate */
+	return calloc(nmemb, size);
+}
+
+/*@@@@ there's probably a better way to prevent overflows when allocating untrusted sums but this works for now */
+
+static FLaC__INLINE void *safe_malloc_add_2op_(size_t size1, size_t size2)
+{
+	size2 += size1;
+	if(size2 < size1)
+		return 0;
+	return safe_malloc_(size2);
+}
+
+static FLaC__INLINE void *safe_malloc_add_3op_(size_t size1, size_t size2, size_t size3)
+{
+	size2 += size1;
+	if(size2 < size1)
+		return 0;
+	size3 += size2;
+	if(size3 < size2)
+		return 0;
+	return safe_malloc_(size3);
+}
+
+static FLaC__INLINE void *safe_malloc_add_4op_(size_t size1, size_t size2, size_t size3, size_t size4)
+{
+	size2 += size1;
+	if(size2 < size1)
+		return 0;
+	size3 += size2;
+	if(size3 < size2)
+		return 0;
+	size4 += size3;
+	if(size4 < size3)
+		return 0;
+	return safe_malloc_(size4);
+}
+
+static FLaC__INLINE void *safe_malloc_mul_2op_(size_t size1, size_t size2)
+#if 0
+needs support for cases where sizeof(size_t) != 4
+{
+	/* could be faster #ifdef'ing off SIZEOF_SIZE_T */
+	if(sizeof(size_t) == 4) {
+		if ((double)size1 * (double)size2 < 4294967296.0)
+			return malloc(size1*size2);
+	}
+	return 0;
+}
+#else
+/* better? */
+{
+	if(!size1 || !size2)
+		return malloc(1); /* malloc(0) is undefined; FLAC src convention is to always allocate */
+	if(size1 > SIZE_MAX / size2)
+		return 0;
+	return malloc(size1*size2);
+}
+#endif
+
+static FLaC__INLINE void *safe_malloc_mul_3op_(size_t size1, size_t size2, size_t size3)
+{
+	if(!size1 || !size2 || !size3)
+		return malloc(1); /* malloc(0) is undefined; FLAC src convention is to always allocate */
+	if(size1 > SIZE_MAX / size2)
+		return 0;
+	size1 *= size2;
+	if(size1 > SIZE_MAX / size3)
+		return 0;
+	return malloc(size1*size3);
+}
+
+/* size1*size2 + size3 */
+static FLaC__INLINE void *safe_malloc_mul2add_(size_t size1, size_t size2, size_t size3)
+{
+	if(!size1 || !size2)
+		return safe_malloc_(size3);
+	if(size1 > SIZE_MAX / size2)
+		return 0;
+	return safe_malloc_add_2op_(size1*size2, size3);
+}
+
+/* size1 * (size2 + size3) */
+static FLaC__INLINE void *safe_malloc_muladd2_(size_t size1, size_t size2, size_t size3)
+{
+	if(!size1 || (!size2 && !size3))
+		return malloc(1); /* malloc(0) is undefined; FLAC src convention is to always allocate */
+	size2 += size3;
+	if(size2 < size3)
+		return 0;
+	return safe_malloc_mul_2op_(size1, size2);
+}
+
+static FLaC__INLINE void *safe_realloc_add_2op_(void *ptr, size_t size1, size_t size2)
+{
+	size2 += size1;
+	if(size2 < size1)
+		return 0;
+	return realloc(ptr, size2);
+}
+
+static FLaC__INLINE void *safe_realloc_add_3op_(void *ptr, size_t size1, size_t size2, size_t size3)
+{
+	size2 += size1;
+	if(size2 < size1)
+		return 0;
+	size3 += size2;
+	if(size3 < size2)
+		return 0;
+	return realloc(ptr, size3);
+}
+
+static FLaC__INLINE void *safe_realloc_add_4op_(void *ptr, size_t size1, size_t size2, size_t size3, size_t size4)
+{
+	size2 += size1;
+	if(size2 < size1)
+		return 0;
+	size3 += size2;
+	if(size3 < size2)
+		return 0;
+	size4 += size3;
+	if(size4 < size3)
+		return 0;
+	return realloc(ptr, size4);
+}
+
+static FLaC__INLINE void *safe_realloc_mul_2op_(void *ptr, size_t size1, size_t size2)
+{
+	if(!size1 || !size2)
+		return realloc(ptr, 0); /* preserve POSIX realloc(ptr, 0) semantics */
+	if(size1 > SIZE_MAX / size2)
+		return 0;
+	return realloc(ptr, size1*size2);
+}
+
+/* size1 * (size2 + size3) */
+static FLaC__INLINE void *safe_realloc_muladd2_(void *ptr, size_t size1, size_t size2, size_t size3)
+{
+	if(!size1 || (!size2 && !size3))
+		return realloc(ptr, 0); /* preserve POSIX realloc(ptr, 0) semantics */
+	size2 += size3;
+	if(size2 < size3)
+		return 0;
+	return safe_realloc_mul_2op_(ptr, size1, size2);
+}
+
+#endif

source/modules/juce_audio_formats/codecs/flac/assert.h

@@ -0,0 +1,45 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__ASSERT_H
+#define FLAC__ASSERT_H
+
+/* we need this since some compilers (like MSVC) leave assert()s on release code (and we don't want to use their ASSERT) */
+#ifdef DEBUG
+#include <assert.h>
+#define FLAC__ASSERT(x) assert(x)
+#define FLAC__ASSERT_DECLARATION(x) x
+#else
+#define FLAC__ASSERT(x)
+#define FLAC__ASSERT_DECLARATION(x)
+#endif
+
+#endif

source/modules/juce_audio_formats/codecs/flac/callback.h

@@ -0,0 +1,184 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__CALLBACK_H
+#define FLAC__CALLBACK_H
+
+#include "ordinals.h"
+#include <stdlib.h> /* for size_t */
+
+/** \file include/FLAC/callback.h
+ *
+ *  \brief
+ *  This module defines the structures for describing I/O callbacks
+ *  to the other FLAC interfaces.
+ *
+ *  See the detailed documentation for callbacks in the
+ *  \link flac_callbacks callbacks \endlink module.
+ */
+
+/** \defgroup flac_callbacks FLAC/callback.h: I/O callback structures
+ *  \ingroup flac
+ *
+ *  \brief
+ *  This module defines the structures for describing I/O callbacks
+ *  to the other FLAC interfaces.
+ *
+ *  The purpose of the I/O callback functions is to create a common way
+ *  for the metadata interfaces to handle I/O.
+ *
+ *  Originally the metadata interfaces required filenames as the way of
+ *  specifying FLAC files to operate on.  This is problematic in some
+ *  environments so there is an additional option to specify a set of
+ *  callbacks for doing I/O on the FLAC file, instead of the filename.
+ *
+ *  In addition to the callbacks, a FLAC__IOHandle type is defined as an
+ *  opaque structure for a data source.
+ *
+ *  The callback function prototypes are similar (but not identical) to the
+ *  stdio functions fread, fwrite, fseek, ftell, feof, and fclose.  If you use
+ *  stdio streams to implement the callbacks, you can pass fread, fwrite, and
+ *  fclose anywhere a FLAC__IOCallback_Read, FLAC__IOCallback_Write, or
+ *  FLAC__IOCallback_Close is required, and a FILE* anywhere a FLAC__IOHandle
+ *  is required.  \warning You generally CANNOT directly use fseek or ftell
+ *  for FLAC__IOCallback_Seek or FLAC__IOCallback_Tell since on most systems
+ *  these use 32-bit offsets and FLAC requires 64-bit offsets to deal with
+ *  large files.  You will have to find an equivalent function (e.g. ftello),
+ *  or write a wrapper.  The same is true for feof() since this is usually
+ *  implemented as a macro, not as a function whose address can be taken.
+ *
+ * \{
+ */
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** This is the opaque handle type used by the callbacks.  Typically
+ *  this is a \c FILE* or address of a file descriptor.
+ */
+typedef void* FLAC__IOHandle;
+
+/** Signature for the read callback.
+ *  The signature and semantics match POSIX fread() implementations
+ *  and can generally be used interchangeably.
+ *
+ * \param  ptr      The address of the read buffer.
+ * \param  size     The size of the records to be read.
+ * \param  nmemb    The number of records to be read.
+ * \param  handle   The handle to the data source.
+ * \retval size_t
+ *    The number of records read.
+ */
+typedef size_t (*FLAC__IOCallback_Read) (void *ptr, size_t size, size_t nmemb, FLAC__IOHandle handle);
+
+/** Signature for the write callback.
+ *  The signature and semantics match POSIX fwrite() implementations
+ *  and can generally be used interchangeably.
+ *
+ * \param  ptr      The address of the write buffer.
+ * \param  size     The size of the records to be written.
+ * \param  nmemb    The number of records to be written.
+ * \param  handle   The handle to the data source.
+ * \retval size_t
+ *    The number of records written.
+ */
+typedef size_t (*FLAC__IOCallback_Write) (const void *ptr, size_t size, size_t nmemb, FLAC__IOHandle handle);
+
+/** Signature for the seek callback.
+ *  The signature and semantics mostly match POSIX fseek() WITH ONE IMPORTANT
+ *  EXCEPTION: the offset is a 64-bit type whereas fseek() is generally 'long'
+ *  and 32-bits wide.
+ *
+ * \param  handle   The handle to the data source.
+ * \param  offset   The new position, relative to \a whence
+ * \param  whence   \c SEEK_SET, \c SEEK_CUR, or \c SEEK_END
+ * \retval int
+ *    \c 0 on success, \c -1 on error.
+ */
+typedef int (*FLAC__IOCallback_Seek) (FLAC__IOHandle handle, FLAC__int64 offset, int whence);
+
+/** Signature for the tell callback.
+ *  The signature and semantics mostly match POSIX ftell() WITH ONE IMPORTANT
+ *  EXCEPTION: the offset is a 64-bit type whereas ftell() is generally 'long'
+ *  and 32-bits wide.
+ *
+ * \param  handle   The handle to the data source.
+ * \retval FLAC__int64
+ *    The current position on success, \c -1 on error.
+ */
+typedef FLAC__int64 (*FLAC__IOCallback_Tell) (FLAC__IOHandle handle);
+
+/** Signature for the EOF callback.
+ *  The signature and semantics mostly match POSIX feof() but WATCHOUT:
+ *  on many systems, feof() is a macro, so in this case a wrapper function
+ *  must be provided instead.
+ *
+ * \param  handle   The handle to the data source.
+ * \retval int
+ *    \c 0 if not at end of file, nonzero if at end of file.
+ */
+typedef int (*FLAC__IOCallback_Eof) (FLAC__IOHandle handle);
+
+/** Signature for the close callback.
+ *  The signature and semantics match POSIX fclose() implementations
+ *  and can generally be used interchangeably.
+ *
+ * \param  handle   The handle to the data source.
+ * \retval int
+ *    \c 0 on success, \c EOF on error.
+ */
+typedef int (*FLAC__IOCallback_Close) (FLAC__IOHandle handle);
+
+/** A structure for holding a set of callbacks.
+ *  Each FLAC interface that requires a FLAC__IOCallbacks structure will
+ *  describe which of the callbacks are required.  The ones that are not
+ *  required may be set to NULL.
+ *
+ *  If the seek requirement for an interface is optional, you can signify that
+ *  a data sorce is not seekable by setting the \a seek field to \c NULL.
+ */
+typedef struct {
+	FLAC__IOCallback_Read read;
+	FLAC__IOCallback_Write write;
+	FLAC__IOCallback_Seek seek;
+	FLAC__IOCallback_Tell tell;
+	FLAC__IOCallback_Eof eof;
+	FLAC__IOCallback_Close close;
+} FLAC__IOCallbacks;
+
+/* \} */
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

source/modules/juce_audio_formats/codecs/flac/export.h

@@ -0,0 +1,91 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__EXPORT_H
+#define FLAC__EXPORT_H
+
+/** \file include/FLAC/export.h
+ *
+ *  \brief
+ *  This module contains #defines and symbols for exporting function
+ *  calls, and providing version information and compiled-in features.
+ *
+ *  See the \link flac_export export \endlink module.
+ */
+
+/** \defgroup flac_export FLAC/export.h: export symbols
+ *  \ingroup flac
+ *
+ *  \brief
+ *  This module contains #defines and symbols for exporting function
+ *  calls, and providing version information and compiled-in features.
+ *
+ *  If you are compiling with MSVC and will link to the static library
+ *  (libFLAC.lib) you should define FLAC__NO_DLL in your project to
+ *  make sure the symbols are exported properly.
+ *
+ * \{
+ */
+
+#if defined(FLAC__NO_DLL) || !defined(_MSC_VER)
+#define FLAC_API
+
+#else
+
+#ifdef FLAC_API_EXPORTS
+#define	FLAC_API	_declspec(dllexport)
+#else
+#define FLAC_API	_declspec(dllimport)
+
+#endif
+#endif
+
+/** These #defines will mirror the libtool-based library version number, see
+ * http://www.gnu.org/software/libtool/manual.html#Libtool-versioning
+ */
+#define FLAC_API_VERSION_CURRENT 10
+#define FLAC_API_VERSION_REVISION 0 /**< see above */
+#define FLAC_API_VERSION_AGE 2 /**< see above */
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/** \c 1 if the library has been compiled with support for Ogg FLAC, else \c 0. */
+extern FLAC_API int FLAC_API_SUPPORTS_OGG_FLAC;
+
+#ifdef __cplusplus
+}
+#endif
+
+/* \} */
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/bitmath.c

@@ -0,0 +1,149 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include "include/private/bitmath.h"
+#include "../assert.h"
+
+/* An example of what FLAC__bitmath_ilog2() computes:
+ *
+ * ilog2( 0) = assertion failure
+ * ilog2( 1) = 0
+ * ilog2( 2) = 1
+ * ilog2( 3) = 1
+ * ilog2( 4) = 2
+ * ilog2( 5) = 2
+ * ilog2( 6) = 2
+ * ilog2( 7) = 2
+ * ilog2( 8) = 3
+ * ilog2( 9) = 3
+ * ilog2(10) = 3
+ * ilog2(11) = 3
+ * ilog2(12) = 3
+ * ilog2(13) = 3
+ * ilog2(14) = 3
+ * ilog2(15) = 3
+ * ilog2(16) = 4
+ * ilog2(17) = 4
+ * ilog2(18) = 4
+ */
+unsigned FLAC__bitmath_ilog2(FLAC__uint32 v)
+{
+	unsigned l = 0;
+	FLAC__ASSERT(v > 0);
+	while(v >>= 1)
+		l++;
+	return l;
+}
+
+unsigned FLAC__bitmath_ilog2_wide(FLAC__uint64 v)
+{
+	unsigned l = 0;
+	FLAC__ASSERT(v > 0);
+	while(v >>= 1)
+		l++;
+	return l;
+}
+
+/* An example of what FLAC__bitmath_silog2() computes:
+ *
+ * silog2(-10) = 5
+ * silog2(- 9) = 5
+ * silog2(- 8) = 4
+ * silog2(- 7) = 4
+ * silog2(- 6) = 4
+ * silog2(- 5) = 4
+ * silog2(- 4) = 3
+ * silog2(- 3) = 3
+ * silog2(- 2) = 2
+ * silog2(- 1) = 2
+ * silog2(  0) = 0
+ * silog2(  1) = 2
+ * silog2(  2) = 3
+ * silog2(  3) = 3
+ * silog2(  4) = 4
+ * silog2(  5) = 4
+ * silog2(  6) = 4
+ * silog2(  7) = 4
+ * silog2(  8) = 5
+ * silog2(  9) = 5
+ * silog2( 10) = 5
+ */
+unsigned FLAC__bitmath_silog2(int v)
+{
+	while(1) {
+		if(v == 0) {
+			return 0;
+		}
+		else if(v > 0) {
+			unsigned l = 0;
+			while(v) {
+				l++;
+				v >>= 1;
+			}
+			return l+1;
+		}
+		else if(v == -1) {
+			return 2;
+		}
+		else {
+			v++;
+			v = -v;
+		}
+	}
+}
+
+unsigned FLAC__bitmath_silog2_wide(FLAC__int64 v)
+{
+	while(1) {
+		if(v == 0) {
+			return 0;
+		}
+		else if(v > 0) {
+			unsigned l = 0;
+			while(v) {
+				l++;
+				v >>= 1;
+			}
+			return l+1;
+		}
+		else if(v == -1) {
+			return 2;
+		}
+		else {
+			v++;
+			v = -v;
+		}
+	}
+}

source/modules/juce_audio_formats/codecs/flac/libFLAC/bitwriter.c

@@ -0,0 +1,880 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include <stdlib.h> /* for malloc() */
+#include <string.h> /* for memcpy(), memset() */
+#ifdef _MSC_VER
+#include <winsock.h> /* for ntohl() */
+#elif defined FLAC__SYS_DARWIN
+#include <machine/endian.h> /* for ntohl() */
+#elif defined __MINGW32__
+#include <winsock.h> /* for ntohl() */
+#else
+#include <netinet/in.h> /* for ntohl() */
+#endif
+#if 0 /* UNUSED */
+#include "include/private/bitmath.h"
+#endif
+#include "include/private/bitwriter.h"
+#include "include/private/crc.h"
+#include "../assert.h"
+#include "../alloc.h"
+
+/* Things should be fastest when this matches the machine word size */
+/* WATCHOUT: if you change this you must also change the following #defines down to SWAP_BE_WORD_TO_HOST below to match */
+/* WATCHOUT: there are a few places where the code will not work unless bwword is >= 32 bits wide */
+typedef FLAC__uint32 bwword;
+#define FLAC__BYTES_PER_WORD 4
+#define FLAC__BITS_PER_WORD 32
+#define FLAC__WORD_ALL_ONES ((FLAC__uint32)0xffffffff)
+/* SWAP_BE_WORD_TO_HOST swaps bytes in a bwword (which is always big-endian) if necessary to match host byte order */
+#if WORDS_BIGENDIAN
+#define SWAP_BE_WORD_TO_HOST(x) (x)
+#else
+#ifdef _MSC_VER
+#define SWAP_BE_WORD_TO_HOST(x) local_swap32_(x)
+#else
+#define SWAP_BE_WORD_TO_HOST(x) ntohl(x)
+#endif
+#endif
+
+/*
+ * The default capacity here doesn't matter too much.  The buffer always grows
+ * to hold whatever is written to it.  Usually the encoder will stop adding at
+ * a frame or metadata block, then write that out and clear the buffer for the
+ * next one.
+ */
+static const unsigned FLAC__BITWRITER_DEFAULT_CAPACITY = 32768u / sizeof(bwword); /* size in words */
+/* When growing, increment 4K at a time */
+static const unsigned FLAC__BITWRITER_DEFAULT_INCREMENT = 4096u / sizeof(bwword); /* size in words */
+
+#define FLAC__WORDS_TO_BITS(words) ((words) * FLAC__BITS_PER_WORD)
+#define FLAC__TOTAL_BITS(bw) (FLAC__WORDS_TO_BITS((bw)->words) + (bw)->bits)
+
+#ifdef min
+#undef min
+#endif
+#define min(x,y) ((x)<(y)?(x):(y))
+
+/* adjust for compilers that can't understand using LLU suffix for uint64_t literals */
+#ifdef _MSC_VER
+#define FLAC__U64L(x) x
+#else
+#define FLAC__U64L(x) x##LLU
+#endif
+
+#ifndef FLaC__INLINE
+#define FLaC__INLINE
+#endif
+
+struct FLAC__BitWriter {
+	bwword *buffer;
+	bwword accum; /* accumulator; bits are right-justified; when full, accum is appended to buffer */
+	unsigned capacity; /* capacity of buffer in words */
+	unsigned words; /* # of complete words in buffer */
+	unsigned bits; /* # of used bits in accum */
+};
+
+/* * WATCHOUT: The current implementation only grows the buffer. */
+static FLAC__bool bitwriter_grow_(FLAC__BitWriter *bw, unsigned bits_to_add)
+{
+	unsigned new_capacity;
+	bwword *new_buffer;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+
+	/* calculate total words needed to store 'bits_to_add' additional bits */
+	new_capacity = bw->words + ((bw->bits + bits_to_add + FLAC__BITS_PER_WORD - 1) / FLAC__BITS_PER_WORD);
+
+	/* it's possible (due to pessimism in the growth estimation that
+	 * leads to this call) that we don't actually need to grow
+	 */
+	if(bw->capacity >= new_capacity)
+		return true;
+
+	/* round up capacity increase to the nearest FLAC__BITWRITER_DEFAULT_INCREMENT */
+	if((new_capacity - bw->capacity) % FLAC__BITWRITER_DEFAULT_INCREMENT)
+		new_capacity += FLAC__BITWRITER_DEFAULT_INCREMENT - ((new_capacity - bw->capacity) % FLAC__BITWRITER_DEFAULT_INCREMENT);
+	/* make sure we got everything right */
+	FLAC__ASSERT(0 == (new_capacity - bw->capacity) % FLAC__BITWRITER_DEFAULT_INCREMENT);
+	FLAC__ASSERT(new_capacity > bw->capacity);
+	FLAC__ASSERT(new_capacity >= bw->words + ((bw->bits + bits_to_add + FLAC__BITS_PER_WORD - 1) / FLAC__BITS_PER_WORD));
+
+	new_buffer = (bwword*)safe_realloc_mul_2op_(bw->buffer, sizeof(bwword), /*times*/new_capacity);
+	if(new_buffer == 0)
+		return false;
+	bw->buffer = new_buffer;
+	bw->capacity = new_capacity;
+	return true;
+}
+
+
+/***********************************************************************
+ *
+ * Class constructor/destructor
+ *
+ ***********************************************************************/
+
+FLAC__BitWriter *FLAC__bitwriter_new(void)
+{
+	FLAC__BitWriter *bw = (FLAC__BitWriter*)calloc(1, sizeof(FLAC__BitWriter));
+	/* note that calloc() sets all members to 0 for us */
+	return bw;
+}
+
+void FLAC__bitwriter_delete(FLAC__BitWriter *bw)
+{
+	FLAC__ASSERT(0 != bw);
+
+	FLAC__bitwriter_free(bw);
+	free(bw);
+}
+
+/***********************************************************************
+ *
+ * Public class methods
+ *
+ ***********************************************************************/
+
+FLAC__bool FLAC__bitwriter_init(FLAC__BitWriter *bw)
+{
+	FLAC__ASSERT(0 != bw);
+
+	bw->words = bw->bits = 0;
+	bw->capacity = FLAC__BITWRITER_DEFAULT_CAPACITY;
+	bw->buffer = (bwword*)malloc(sizeof(bwword) * bw->capacity);
+	if(bw->buffer == 0)
+		return false;
+
+	return true;
+}
+
+void FLAC__bitwriter_free(FLAC__BitWriter *bw)
+{
+	FLAC__ASSERT(0 != bw);
+
+	if(0 != bw->buffer)
+		free(bw->buffer);
+	bw->buffer = 0;
+	bw->capacity = 0;
+	bw->words = bw->bits = 0;
+}
+
+void FLAC__bitwriter_clear(FLAC__BitWriter *bw)
+{
+	bw->words = bw->bits = 0;
+}
+
+void FLAC__bitwriter_dump(const FLAC__BitWriter *bw, FILE *out)
+{
+	unsigned i, j;
+	if(bw == 0) {
+		fprintf(out, "bitwriter is NULL\n");
+	}
+	else {
+		fprintf(out, "bitwriter: capacity=%u words=%u bits=%u total_bits=%u\n", bw->capacity, bw->words, bw->bits, FLAC__TOTAL_BITS(bw));
+
+		for(i = 0; i < bw->words; i++) {
+			fprintf(out, "%08X: ", i);
+			for(j = 0; j < FLAC__BITS_PER_WORD; j++)
+				fprintf(out, "%01u", bw->buffer[i] & (1 << (FLAC__BITS_PER_WORD-j-1)) ? 1:0);
+			fprintf(out, "\n");
+		}
+		if(bw->bits > 0) {
+			fprintf(out, "%08X: ", i);
+			for(j = 0; j < bw->bits; j++)
+				fprintf(out, "%01u", bw->accum & (1 << (bw->bits-j-1)) ? 1:0);
+			fprintf(out, "\n");
+		}
+	}
+}
+
+FLAC__bool FLAC__bitwriter_get_write_crc16(FLAC__BitWriter *bw, FLAC__uint16 *crc)
+{
+	const FLAC__byte *buffer;
+	size_t bytes;
+
+	FLAC__ASSERT((bw->bits & 7) == 0); /* assert that we're byte-aligned */
+
+	if(!FLAC__bitwriter_get_buffer(bw, &buffer, &bytes))
+		return false;
+
+	*crc = (FLAC__uint16)FLAC__crc16(buffer, bytes);
+	FLAC__bitwriter_release_buffer(bw);
+	return true;
+}
+
+FLAC__bool FLAC__bitwriter_get_write_crc8(FLAC__BitWriter *bw, FLAC__byte *crc)
+{
+	const FLAC__byte *buffer;
+	size_t bytes;
+
+	FLAC__ASSERT((bw->bits & 7) == 0); /* assert that we're byte-aligned */
+
+	if(!FLAC__bitwriter_get_buffer(bw, &buffer, &bytes))
+		return false;
+
+	*crc = FLAC__crc8(buffer, bytes);
+	FLAC__bitwriter_release_buffer(bw);
+	return true;
+}
+
+FLAC__bool FLAC__bitwriter_is_byte_aligned(const FLAC__BitWriter *bw)
+{
+	return ((bw->bits & 7) == 0);
+}
+
+unsigned FLAC__bitwriter_get_input_bits_unconsumed(const FLAC__BitWriter *bw)
+{
+	return FLAC__TOTAL_BITS(bw);
+}
+
+FLAC__bool FLAC__bitwriter_get_buffer(FLAC__BitWriter *bw, const FLAC__byte **buffer, size_t *bytes)
+{
+	FLAC__ASSERT((bw->bits & 7) == 0);
+	/* double protection */
+	if(bw->bits & 7)
+		return false;
+	/* if we have bits in the accumulator we have to flush those to the buffer first */
+	if(bw->bits) {
+		FLAC__ASSERT(bw->words <= bw->capacity);
+		if(bw->words == bw->capacity && !bitwriter_grow_(bw, FLAC__BITS_PER_WORD))
+			return false;
+		/* append bits as complete word to buffer, but don't change bw->accum or bw->bits */
+		bw->buffer[bw->words] = SWAP_BE_WORD_TO_HOST(bw->accum << (FLAC__BITS_PER_WORD-bw->bits));
+	}
+	/* now we can just return what we have */
+	*buffer = (FLAC__byte*)bw->buffer;
+	*bytes = (FLAC__BYTES_PER_WORD * bw->words) + (bw->bits >> 3);
+	return true;
+}
+
+void FLAC__bitwriter_release_buffer(FLAC__BitWriter *bw)
+{
+	/* nothing to do.  in the future, strict checking of a 'writer-is-in-
+	 * get-mode' flag could be added everywhere and then cleared here
+	 */
+	(void)bw;
+}
+
+FLaC__INLINE FLAC__bool FLAC__bitwriter_write_zeroes(FLAC__BitWriter *bw, unsigned bits)
+{
+	unsigned n;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+
+	if(bits == 0)
+		return true;
+	/* slightly pessimistic size check but faster than "<= bw->words + (bw->bits+bits+FLAC__BITS_PER_WORD-1)/FLAC__BITS_PER_WORD" */
+	if(bw->capacity <= bw->words + bits && !bitwriter_grow_(bw, bits))
+		return false;
+	/* first part gets to word alignment */
+	if(bw->bits) {
+		n = min(FLAC__BITS_PER_WORD - bw->bits, bits);
+		bw->accum <<= n;
+		bits -= n;
+		bw->bits += n;
+		if(bw->bits == FLAC__BITS_PER_WORD) {
+			bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
+			bw->bits = 0;
+		}
+		else
+			return true;
+	}
+	/* do whole words */
+	while(bits >= FLAC__BITS_PER_WORD) {
+		bw->buffer[bw->words++] = 0;
+		bits -= FLAC__BITS_PER_WORD;
+	}
+	/* do any leftovers */
+	if(bits > 0) {
+		bw->accum = 0;
+		bw->bits = bits;
+	}
+	return true;
+}
+
+FLaC__INLINE FLAC__bool FLAC__bitwriter_write_raw_uint32(FLAC__BitWriter *bw, FLAC__uint32 val, unsigned bits)
+{
+	register unsigned left;
+
+	/* WATCHOUT: code does not work with <32bit words; we can make things much faster with this assertion */
+	FLAC__ASSERT(FLAC__BITS_PER_WORD >= 32);
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+
+	FLAC__ASSERT(bits <= 32);
+	if(bits == 0)
+		return true;
+
+	/* slightly pessimistic size check but faster than "<= bw->words + (bw->bits+bits+FLAC__BITS_PER_WORD-1)/FLAC__BITS_PER_WORD" */
+	if(bw->capacity <= bw->words + bits && !bitwriter_grow_(bw, bits))
+		return false;
+
+	left = FLAC__BITS_PER_WORD - bw->bits;
+	if(bits < left) {
+		bw->accum <<= bits;
+		bw->accum |= val;
+		bw->bits += bits;
+	}
+	else if(bw->bits) { /* WATCHOUT: if bw->bits == 0, left==FLAC__BITS_PER_WORD and bw->accum<<=left is a NOP instead of setting to 0 */
+		bw->accum <<= left;
+		bw->accum |= val >> (bw->bits = bits - left);
+		bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
+		bw->accum = val;
+	}
+	else {
+		bw->accum = val;
+		bw->bits = 0;
+		bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(val);
+	}
+
+	return true;
+}
+
+FLaC__INLINE FLAC__bool FLAC__bitwriter_write_raw_int32(FLAC__BitWriter *bw, FLAC__int32 val, unsigned bits)
+{
+	/* zero-out unused bits */
+	if(bits < 32)
+		val &= (~(0xffffffff << bits));
+
+	return FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)val, bits);
+}
+
+FLaC__INLINE FLAC__bool FLAC__bitwriter_write_raw_uint64(FLAC__BitWriter *bw, FLAC__uint64 val, unsigned bits)
+{
+	/* this could be a little faster but it's not used for much */
+	if(bits > 32) {
+		return
+			FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)(val>>32), bits-32) &&
+			FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)val, 32);
+	}
+	else
+		return FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)val, bits);
+}
+
+FLaC__INLINE FLAC__bool FLAC__bitwriter_write_raw_uint32_little_endian(FLAC__BitWriter *bw, FLAC__uint32 val)
+{
+	/* this doesn't need to be that fast as currently it is only used for vorbis comments */
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, val & 0xff, 8))
+		return false;
+	if(!FLAC__bitwriter_write_raw_uint32(bw, (val>>8) & 0xff, 8))
+		return false;
+	if(!FLAC__bitwriter_write_raw_uint32(bw, (val>>16) & 0xff, 8))
+		return false;
+	if(!FLAC__bitwriter_write_raw_uint32(bw, val>>24, 8))
+		return false;
+
+	return true;
+}
+
+FLaC__INLINE FLAC__bool FLAC__bitwriter_write_byte_block(FLAC__BitWriter *bw, const FLAC__byte vals[], unsigned nvals)
+{
+	unsigned i;
+
+	/* this could be faster but currently we don't need it to be since it's only used for writing metadata */
+	for(i = 0; i < nvals; i++) {
+		if(!FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)(vals[i]), 8))
+			return false;
+	}
+
+	return true;
+}
+
+FLAC__bool FLAC__bitwriter_write_unary_unsigned(FLAC__BitWriter *bw, unsigned val)
+{
+	if(val < 32)
+		return FLAC__bitwriter_write_raw_uint32(bw, 1, ++val);
+	else
+		return
+			FLAC__bitwriter_write_zeroes(bw, val) &&
+			FLAC__bitwriter_write_raw_uint32(bw, 1, 1);
+}
+
+unsigned FLAC__bitwriter_rice_bits(FLAC__int32 val, unsigned parameter)
+{
+	FLAC__uint32 uval;
+
+	FLAC__ASSERT(parameter < sizeof(unsigned)*8);
+
+	/* fold signed to unsigned; actual formula is: negative(v)? -2v-1 : 2v */
+	uval = (val<<1) ^ (val>>31);
+
+	return 1 + parameter + (uval >> parameter);
+}
+
+#if 0 /* UNUSED */
+unsigned FLAC__bitwriter_golomb_bits_signed(int val, unsigned parameter)
+{
+	unsigned bits, msbs, uval;
+	unsigned k;
+
+	FLAC__ASSERT(parameter > 0);
+
+	/* fold signed to unsigned */
+	if(val < 0)
+		uval = (unsigned)(((-(++val)) << 1) + 1);
+	else
+		uval = (unsigned)(val << 1);
+
+	k = FLAC__bitmath_ilog2(parameter);
+	if(parameter == 1u<<k) {
+		FLAC__ASSERT(k <= 30);
+
+		msbs = uval >> k;
+		bits = 1 + k + msbs;
+	}
+	else {
+		unsigned q, r, d;
+
+		d = (1 << (k+1)) - parameter;
+		q = uval / parameter;
+		r = uval - (q * parameter);
+
+		bits = 1 + q + k;
+		if(r >= d)
+			bits++;
+	}
+	return bits;
+}
+
+unsigned FLAC__bitwriter_golomb_bits_unsigned(unsigned uval, unsigned parameter)
+{
+	unsigned bits, msbs;
+	unsigned k;
+
+	FLAC__ASSERT(parameter > 0);
+
+	k = FLAC__bitmath_ilog2(parameter);
+	if(parameter == 1u<<k) {
+		FLAC__ASSERT(k <= 30);
+
+		msbs = uval >> k;
+		bits = 1 + k + msbs;
+	}
+	else {
+		unsigned q, r, d;
+
+		d = (1 << (k+1)) - parameter;
+		q = uval / parameter;
+		r = uval - (q * parameter);
+
+		bits = 1 + q + k;
+		if(r >= d)
+			bits++;
+	}
+	return bits;
+}
+#endif /* UNUSED */
+
+FLAC__bool FLAC__bitwriter_write_rice_signed(FLAC__BitWriter *bw, FLAC__int32 val, unsigned parameter)
+{
+	unsigned total_bits, interesting_bits, msbs;
+	FLAC__uint32 uval, pattern;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+	FLAC__ASSERT(parameter < 8*sizeof(uval));
+
+	/* fold signed to unsigned; actual formula is: negative(v)? -2v-1 : 2v */
+	uval = (val<<1) ^ (val>>31);
+
+	msbs = uval >> parameter;
+	interesting_bits = 1 + parameter;
+	total_bits = interesting_bits + msbs;
+	pattern = 1 << parameter; /* the unary end bit */
+	pattern |= (uval & ((1<<parameter)-1)); /* the binary LSBs */
+
+	if(total_bits <= 32)
+		return FLAC__bitwriter_write_raw_uint32(bw, pattern, total_bits);
+	else
+		return
+			FLAC__bitwriter_write_zeroes(bw, msbs) && /* write the unary MSBs */
+			FLAC__bitwriter_write_raw_uint32(bw, pattern, interesting_bits); /* write the unary end bit and binary LSBs */
+}
+
+FLAC__bool FLAC__bitwriter_write_rice_signed_block(FLAC__BitWriter *bw, const FLAC__int32 *vals, unsigned nvals, unsigned parameter)
+{
+	const FLAC__uint32 mask1 = FLAC__WORD_ALL_ONES << parameter; /* we val|=mask1 to set the stop bit above it... */
+	const FLAC__uint32 mask2 = FLAC__WORD_ALL_ONES >> (31-parameter); /* ...then mask off the bits above the stop bit with val&=mask2*/
+	FLAC__uint32 uval;
+	unsigned left;
+	const unsigned lsbits = 1 + parameter;
+	unsigned msbits;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+	FLAC__ASSERT(parameter < 8*sizeof(bwword)-1);
+	/* WATCHOUT: code does not work with <32bit words; we can make things much faster with this assertion */
+	FLAC__ASSERT(FLAC__BITS_PER_WORD >= 32);
+
+	while(nvals) {
+		/* fold signed to unsigned; actual formula is: negative(v)? -2v-1 : 2v */
+		uval = (*vals<<1) ^ (*vals>>31);
+
+		msbits = uval >> parameter;
+
+#if 0 /* OPT: can remove this special case if it doesn't make up for the extra compare (doesn't make a statistically significant difference with msvc or gcc/x86) */
+		if(bw->bits && bw->bits + msbits + lsbits <= FLAC__BITS_PER_WORD) { /* i.e. if the whole thing fits in the current bwword */
+			/* ^^^ if bw->bits is 0 then we may have filled the buffer and have no free bwword to work in */
+			bw->bits = bw->bits + msbits + lsbits;
+			uval |= mask1; /* set stop bit */
+			uval &= mask2; /* mask off unused top bits */
+			/* NOT: bw->accum <<= msbits + lsbits because msbits+lsbits could be 32, then the shift would be a NOP */
+			bw->accum <<= msbits;
+			bw->accum <<= lsbits;
+			bw->accum |= uval;
+			if(bw->bits == FLAC__BITS_PER_WORD) {
+				bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
+				bw->bits = 0;
+				/* burying the capacity check down here means we have to grow the buffer a little if there are more vals to do */
+				if(bw->capacity <= bw->words && nvals > 1 && !bitwriter_grow_(bw, 1)) {
+					FLAC__ASSERT(bw->capacity == bw->words);
+					return false;
+				}
+			}
+		}
+		else {
+#elif 1 /*@@@@@@ OPT: try this version with MSVC6 to see if better, not much difference for gcc-4 */
+		if(bw->bits && bw->bits + msbits + lsbits < FLAC__BITS_PER_WORD) { /* i.e. if the whole thing fits in the current bwword */
+			/* ^^^ if bw->bits is 0 then we may have filled the buffer and have no free bwword to work in */
+			bw->bits = bw->bits + msbits + lsbits;
+			uval |= mask1; /* set stop bit */
+			uval &= mask2; /* mask off unused top bits */
+			bw->accum <<= msbits + lsbits;
+			bw->accum |= uval;
+		}
+		else {
+#endif
+			/* slightly pessimistic size check but faster than "<= bw->words + (bw->bits+msbits+lsbits+FLAC__BITS_PER_WORD-1)/FLAC__BITS_PER_WORD" */
+			/* OPT: pessimism may cause flurry of false calls to grow_ which eat up all savings before it */
+			if(bw->capacity <= bw->words + bw->bits + msbits + 1/*lsbits always fit in 1 bwword*/ && !bitwriter_grow_(bw, msbits+lsbits))
+				return false;
+
+			if(msbits) {
+				/* first part gets to word alignment */
+				if(bw->bits) {
+					left = FLAC__BITS_PER_WORD - bw->bits;
+					if(msbits < left) {
+						bw->accum <<= msbits;
+						bw->bits += msbits;
+						goto break1;
+					}
+					else {
+						bw->accum <<= left;
+						msbits -= left;
+						bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
+						bw->bits = 0;
+					}
+				}
+				/* do whole words */
+				while(msbits >= FLAC__BITS_PER_WORD) {
+					bw->buffer[bw->words++] = 0;
+					msbits -= FLAC__BITS_PER_WORD;
+				}
+				/* do any leftovers */
+				if(msbits > 0) {
+					bw->accum = 0;
+					bw->bits = msbits;
+				}
+			}
+break1:
+			uval |= mask1; /* set stop bit */
+			uval &= mask2; /* mask off unused top bits */
+
+			left = FLAC__BITS_PER_WORD - bw->bits;
+			if(lsbits < left) {
+				bw->accum <<= lsbits;
+				bw->accum |= uval;
+				bw->bits += lsbits;
+			}
+			else {
+				/* if bw->bits == 0, left==FLAC__BITS_PER_WORD which will always
+				 * be > lsbits (because of previous assertions) so it would have
+				 * triggered the (lsbits<left) case above.
+				 */
+				FLAC__ASSERT(bw->bits);
+				FLAC__ASSERT(left < FLAC__BITS_PER_WORD);
+				bw->accum <<= left;
+				bw->accum |= uval >> (bw->bits = lsbits - left);
+				bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
+				bw->accum = uval;
+			}
+#if 1
+		}
+#endif
+		vals++;
+		nvals--;
+	}
+	return true;
+}
+
+#if 0 /* UNUSED */
+FLAC__bool FLAC__bitwriter_write_golomb_signed(FLAC__BitWriter *bw, int val, unsigned parameter)
+{
+	unsigned total_bits, msbs, uval;
+	unsigned k;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+	FLAC__ASSERT(parameter > 0);
+
+	/* fold signed to unsigned */
+	if(val < 0)
+		uval = (unsigned)(((-(++val)) << 1) + 1);
+	else
+		uval = (unsigned)(val << 1);
+
+	k = FLAC__bitmath_ilog2(parameter);
+	if(parameter == 1u<<k) {
+		unsigned pattern;
+
+		FLAC__ASSERT(k <= 30);
+
+		msbs = uval >> k;
+		total_bits = 1 + k + msbs;
+		pattern = 1 << k; /* the unary end bit */
+		pattern |= (uval & ((1u<<k)-1)); /* the binary LSBs */
+
+		if(total_bits <= 32) {
+			if(!FLAC__bitwriter_write_raw_uint32(bw, pattern, total_bits))
+				return false;
+		}
+		else {
+			/* write the unary MSBs */
+			if(!FLAC__bitwriter_write_zeroes(bw, msbs))
+				return false;
+			/* write the unary end bit and binary LSBs */
+			if(!FLAC__bitwriter_write_raw_uint32(bw, pattern, k+1))
+				return false;
+		}
+	}
+	else {
+		unsigned q, r, d;
+
+		d = (1 << (k+1)) - parameter;
+		q = uval / parameter;
+		r = uval - (q * parameter);
+		/* write the unary MSBs */
+		if(!FLAC__bitwriter_write_zeroes(bw, q))
+			return false;
+		/* write the unary end bit */
+		if(!FLAC__bitwriter_write_raw_uint32(bw, 1, 1))
+			return false;
+		/* write the binary LSBs */
+		if(r >= d) {
+			if(!FLAC__bitwriter_write_raw_uint32(bw, r+d, k+1))
+				return false;
+		}
+		else {
+			if(!FLAC__bitwriter_write_raw_uint32(bw, r, k))
+				return false;
+		}
+	}
+	return true;
+}
+
+FLAC__bool FLAC__bitwriter_write_golomb_unsigned(FLAC__BitWriter *bw, unsigned uval, unsigned parameter)
+{
+	unsigned total_bits, msbs;
+	unsigned k;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+	FLAC__ASSERT(parameter > 0);
+
+	k = FLAC__bitmath_ilog2(parameter);
+	if(parameter == 1u<<k) {
+		unsigned pattern;
+
+		FLAC__ASSERT(k <= 30);
+
+		msbs = uval >> k;
+		total_bits = 1 + k + msbs;
+		pattern = 1 << k; /* the unary end bit */
+		pattern |= (uval & ((1u<<k)-1)); /* the binary LSBs */
+
+		if(total_bits <= 32) {
+			if(!FLAC__bitwriter_write_raw_uint32(bw, pattern, total_bits))
+				return false;
+		}
+		else {
+			/* write the unary MSBs */
+			if(!FLAC__bitwriter_write_zeroes(bw, msbs))
+				return false;
+			/* write the unary end bit and binary LSBs */
+			if(!FLAC__bitwriter_write_raw_uint32(bw, pattern, k+1))
+				return false;
+		}
+	}
+	else {
+		unsigned q, r, d;
+
+		d = (1 << (k+1)) - parameter;
+		q = uval / parameter;
+		r = uval - (q * parameter);
+		/* write the unary MSBs */
+		if(!FLAC__bitwriter_write_zeroes(bw, q))
+			return false;
+		/* write the unary end bit */
+		if(!FLAC__bitwriter_write_raw_uint32(bw, 1, 1))
+			return false;
+		/* write the binary LSBs */
+		if(r >= d) {
+			if(!FLAC__bitwriter_write_raw_uint32(bw, r+d, k+1))
+				return false;
+		}
+		else {
+			if(!FLAC__bitwriter_write_raw_uint32(bw, r, k))
+				return false;
+		}
+	}
+	return true;
+}
+#endif /* UNUSED */
+
+FLAC__bool FLAC__bitwriter_write_utf8_uint32(FLAC__BitWriter *bw, FLAC__uint32 val)
+{
+	FLAC__bool ok = 1;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+
+	FLAC__ASSERT(!(val & 0x80000000)); /* this version only handles 31 bits */
+
+	if(val < 0x80) {
+		return FLAC__bitwriter_write_raw_uint32(bw, val, 8);
+	}
+	else if(val < 0x800) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xC0 | (val>>6), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
+	}
+	else if(val < 0x10000) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xE0 | (val>>12), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
+	}
+	else if(val < 0x200000) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xF0 | (val>>18), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>12)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
+	}
+	else if(val < 0x4000000) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xF8 | (val>>24), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>18)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>12)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
+	}
+	else {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xFC | (val>>30), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>24)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>18)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>12)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
+	}
+
+	return ok;
+}
+
+FLAC__bool FLAC__bitwriter_write_utf8_uint64(FLAC__BitWriter *bw, FLAC__uint64 val)
+{
+	FLAC__bool ok = 1;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+
+	FLAC__ASSERT(!(val & FLAC__U64L(0xFFFFFFF000000000))); /* this version only handles 36 bits */
+
+	if(val < 0x80) {
+		return FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)val, 8);
+	}
+	else if(val < 0x800) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xC0 | (FLAC__uint32)(val>>6), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
+	}
+	else if(val < 0x10000) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xE0 | (FLAC__uint32)(val>>12), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
+	}
+	else if(val < 0x200000) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xF0 | (FLAC__uint32)(val>>18), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>12)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
+	}
+	else if(val < 0x4000000) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xF8 | (FLAC__uint32)(val>>24), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>18)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>12)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
+	}
+	else if(val < 0x80000000) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xFC | (FLAC__uint32)(val>>30), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>24)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>18)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>12)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
+	}
+	else {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xFE, 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>30)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>24)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>18)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>12)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
+	}
+
+	return ok;
+}
+
+FLAC__bool FLAC__bitwriter_zero_pad_to_byte_boundary(FLAC__BitWriter *bw)
+{
+	/* 0-pad to byte boundary */
+	if(bw->bits & 7u)
+		return FLAC__bitwriter_write_zeroes(bw, 8 - (bw->bits & 7u));
+	else
+		return true;
+}

source/modules/juce_audio_formats/codecs/flac/libFLAC/cpu.c

@@ -0,0 +1,418 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include "include/private/cpu.h"
+#include <stdlib.h>
+#include <stdio.h>
+
+#if defined FLAC__CPU_IA32
+# include <signal.h>
+#elif defined FLAC__CPU_PPC
+# if !defined FLAC__NO_ASM
+#  if defined FLAC__SYS_DARWIN
+#   include <sys/sysctl.h>
+#   include <mach/mach.h>
+#   include <mach/mach_host.h>
+#   include <mach/host_info.h>
+#   include <mach/machine.h>
+#   ifndef CPU_SUBTYPE_POWERPC_970
+#    define CPU_SUBTYPE_POWERPC_970 ((cpu_subtype_t) 100)
+#   endif
+#  else /* FLAC__SYS_DARWIN */
+
+#   include <signal.h>
+#   include <setjmp.h>
+
+static sigjmp_buf jmpbuf;
+static volatile sig_atomic_t canjump = 0;
+
+static void sigill_handler (int sig)
+{
+	if (!canjump) {
+		signal (sig, SIG_DFL);
+		raise (sig);
+	}
+	canjump = 0;
+	siglongjmp (jmpbuf, 1);
+}
+#  endif /* FLAC__SYS_DARWIN */
+# endif /* FLAC__NO_ASM */
+#endif /* FLAC__CPU_PPC */
+
+#if defined (__NetBSD__) || defined(__OpenBSD__)
+#include <sys/param.h>
+#include <sys/sysctl.h>
+#include <machine/cpu.h>
+#endif
+
+#if defined(__FreeBSD__) || defined(__FreeBSD_kernel__) || defined(__DragonFly__)
+#include <sys/types.h>
+#include <sys/sysctl.h>
+#endif
+
+#if defined(__APPLE__)
+/* how to get sysctlbyname()? */
+#endif
+
+/* these are flags in EDX of CPUID AX=00000001 */
+static const unsigned FLAC__CPUINFO_IA32_CPUID_CMOV = 0x00008000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_MMX = 0x00800000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_FXSR = 0x01000000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE = 0x02000000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE2 = 0x04000000;
+/* these are flags in ECX of CPUID AX=00000001 */
+static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE3 = 0x00000001;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_SSSE3 = 0x00000200;
+/* these are flags in EDX of CPUID AX=80000001 */
+static const unsigned FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_3DNOW = 0x80000000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_EXT3DNOW = 0x40000000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_EXTMMX = 0x00400000;
+
+
+/*
+ * Extra stuff needed for detection of OS support for SSE on IA-32
+ */
+#if defined(FLAC__CPU_IA32) && !defined FLAC__NO_ASM && defined FLAC__HAS_NASM && !defined FLAC__NO_SSE_OS && !defined FLAC__SSE_OS
+# if defined(__linux__)
+/*
+ * If the OS doesn't support SSE, we will get here with a SIGILL.  We
+ * modify the return address to jump over the offending SSE instruction
+ * and also the operation following it that indicates the instruction
+ * executed successfully.  In this way we use no global variables and
+ * stay thread-safe.
+ *
+ * 3 + 3 + 6:
+ *   3 bytes for "xorps xmm0,xmm0"
+ *   3 bytes for estimate of how long the follwing "inc var" instruction is
+ *   6 bytes extra in case our estimate is wrong
+ * 12 bytes puts us in the NOP "landing zone"
+ */
+#  undef USE_OBSOLETE_SIGCONTEXT_FLAVOR /* #define this to use the older signal handler method */
+#  ifdef USE_OBSOLETE_SIGCONTEXT_FLAVOR
+	static void sigill_handler_sse_os(int signal, struct sigcontext sc)
+	{
+		(void)signal;
+		sc.eip += 3 + 3 + 6;
+	}
+#  else
+#   include <sys/ucontext.h>
+	static void sigill_handler_sse_os(int signal, siginfo_t *si, void *uc)
+	{
+		(void)signal, (void)si;
+		((ucontext_t*)uc)->uc_mcontext.gregs[14/*REG_EIP*/] += 3 + 3 + 6;
+	}
+#  endif
+# elif defined(_MSC_VER)
+#  include <windows.h>
+#  undef USE_TRY_CATCH_FLAVOR /* #define this to use the try/catch method for catching illegal opcode exception */
+#  ifdef USE_TRY_CATCH_FLAVOR
+#  else
+	LONG CALLBACK sigill_handler_sse_os(EXCEPTION_POINTERS *ep)
+	{
+		if(ep->ExceptionRecord->ExceptionCode == EXCEPTION_ILLEGAL_INSTRUCTION) {
+			ep->ContextRecord->Eip += 3 + 3 + 6;
+			return EXCEPTION_CONTINUE_EXECUTION;
+		}
+		return EXCEPTION_CONTINUE_SEARCH;
+	}
+#  endif
+# endif
+#endif
+
+
+void FLAC__cpu_info(FLAC__CPUInfo *info)
+{
+/*
+ * IA32-specific
+ */
+#ifdef FLAC__CPU_IA32
+	info->type = FLAC__CPUINFO_TYPE_IA32;
+#if !defined FLAC__NO_ASM && defined FLAC__HAS_NASM
+	info->use_asm = true; /* we assume a minimum of 80386 with FLAC__CPU_IA32 */
+	info->data.ia32.cpuid = FLAC__cpu_have_cpuid_asm_ia32()? true : false;
+	info->data.ia32.bswap = info->data.ia32.cpuid; /* CPUID => BSWAP since it came after */
+	info->data.ia32.cmov = false;
+	info->data.ia32.mmx = false;
+	info->data.ia32.fxsr = false;
+	info->data.ia32.sse = false;
+	info->data.ia32.sse2 = false;
+	info->data.ia32.sse3 = false;
+	info->data.ia32.ssse3 = false;
+	info->data.ia32._3dnow = false;
+	info->data.ia32.ext3dnow = false;
+	info->data.ia32.extmmx = false;
+	if(info->data.ia32.cpuid) {
+		/* http://www.sandpile.org/ia32/cpuid.htm */
+		FLAC__uint32 flags_edx, flags_ecx;
+		FLAC__cpu_info_asm_ia32(&flags_edx, &flags_ecx);
+		info->data.ia32.cmov  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_CMOV )? true : false;
+		info->data.ia32.mmx   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_MMX  )? true : false;
+		info->data.ia32.fxsr  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_FXSR )? true : false;
+		info->data.ia32.sse   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_SSE  )? true : false;
+		info->data.ia32.sse2  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_SSE2 )? true : false;
+		info->data.ia32.sse3  = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE3 )? true : false;
+		info->data.ia32.ssse3 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSSE3)? true : false;
+
+#ifdef FLAC__USE_3DNOW
+		flags_edx = FLAC__cpu_info_extended_amd_asm_ia32();
+		info->data.ia32._3dnow   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_3DNOW   )? true : false;
+		info->data.ia32.ext3dnow = (flags_edx & FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_EXT3DNOW)? true : false;
+		info->data.ia32.extmmx   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_EXTMMX  )? true : false;
+#else
+		info->data.ia32._3dnow = info->data.ia32.ext3dnow = info->data.ia32.extmmx = false;
+#endif
+
+#ifdef DEBUG
+		fprintf(stderr, "CPU info (IA-32):\n");
+		fprintf(stderr, "  CPUID ...... %c\n", info->data.ia32.cpuid   ? 'Y' : 'n');
+		fprintf(stderr, "  BSWAP ...... %c\n", info->data.ia32.bswap   ? 'Y' : 'n');
+		fprintf(stderr, "  CMOV ....... %c\n", info->data.ia32.cmov    ? 'Y' : 'n');
+		fprintf(stderr, "  MMX ........ %c\n", info->data.ia32.mmx     ? 'Y' : 'n');
+		fprintf(stderr, "  FXSR ....... %c\n", info->data.ia32.fxsr    ? 'Y' : 'n');
+		fprintf(stderr, "  SSE ........ %c\n", info->data.ia32.sse     ? 'Y' : 'n');
+		fprintf(stderr, "  SSE2 ....... %c\n", info->data.ia32.sse2    ? 'Y' : 'n');
+		fprintf(stderr, "  SSE3 ....... %c\n", info->data.ia32.sse3    ? 'Y' : 'n');
+		fprintf(stderr, "  SSSE3 ...... %c\n", info->data.ia32.ssse3   ? 'Y' : 'n');
+		fprintf(stderr, "  3DNow! ..... %c\n", info->data.ia32._3dnow  ? 'Y' : 'n');
+		fprintf(stderr, "  3DNow!-ext . %c\n", info->data.ia32.ext3dnow? 'Y' : 'n');
+		fprintf(stderr, "  3DNow!-MMX . %c\n", info->data.ia32.extmmx  ? 'Y' : 'n');
+#endif
+
+		/*
+		 * now have to check for OS support of SSE/SSE2
+		 */
+		if(info->data.ia32.fxsr || info->data.ia32.sse || info->data.ia32.sse2) {
+#if defined FLAC__NO_SSE_OS
+			/* assume user knows better than us; turn it off */
+			info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+#elif defined FLAC__SSE_OS
+			/* assume user knows better than us; leave as detected above */
+#elif defined(__FreeBSD__) || defined(__FreeBSD_kernel__) || defined(__DragonFly__) || defined(__APPLE__)
+			int sse = 0;
+			size_t len;
+			/* at least one of these must work: */
+			len = sizeof(sse); sse = sse || (sysctlbyname("hw.instruction_sse", &sse, &len, NULL, 0) == 0 && sse);
+			len = sizeof(sse); sse = sse || (sysctlbyname("hw.optional.sse"   , &sse, &len, NULL, 0) == 0 && sse); /* __APPLE__ ? */
+			if(!sse)
+				info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+#elif defined(__NetBSD__) || defined (__OpenBSD__)
+# if __NetBSD_Version__ >= 105250000 || (defined __OpenBSD__)
+			int val = 0, mib[2] = { CTL_MACHDEP, CPU_SSE };
+			size_t len = sizeof(val);
+			if(sysctl(mib, 2, &val, &len, NULL, 0) < 0 || !val)
+				info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+			else { /* double-check SSE2 */
+				mib[1] = CPU_SSE2;
+				len = sizeof(val);
+				if(sysctl(mib, 2, &val, &len, NULL, 0) < 0 || !val)
+					info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+			}
+# else
+			info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+# endif
+#elif defined(__linux__)
+			int sse = 0;
+			struct sigaction sigill_save;
+#ifdef USE_OBSOLETE_SIGCONTEXT_FLAVOR
+			if(0 == sigaction(SIGILL, NULL, &sigill_save) && signal(SIGILL, (void (*)(int))sigill_handler_sse_os) != SIG_ERR)
+#else
+			struct sigaction sigill_sse;
+			sigill_sse.sa_sigaction = sigill_handler_sse_os;
+			__sigemptyset(&sigill_sse.sa_mask);
+			sigill_sse.sa_flags = SA_SIGINFO | SA_RESETHAND; /* SA_RESETHAND just in case our SIGILL return jump breaks, so we don't get stuck in a loop */
+			if(0 == sigaction(SIGILL, &sigill_sse, &sigill_save))
+#endif
+			{
+				/* http://www.ibiblio.org/gferg/ldp/GCC-Inline-Assembly-HOWTO.html */
+				/* see sigill_handler_sse_os() for an explanation of the following: */
+				asm volatile (
+					"xorl %0,%0\n\t"          /* for some reason, still need to do this to clear 'sse' var */
+					"xorps %%xmm0,%%xmm0\n\t" /* will cause SIGILL if unsupported by OS */
+					"incl %0\n\t"             /* SIGILL handler will jump over this */
+					/* landing zone */
+					"nop\n\t" /* SIGILL jump lands here if "inc" is 9 bytes */
+					"nop\n\t"
+					"nop\n\t"
+					"nop\n\t"
+					"nop\n\t"
+					"nop\n\t"
+					"nop\n\t" /* SIGILL jump lands here if "inc" is 3 bytes (expected) */
+					"nop\n\t"
+					"nop"     /* SIGILL jump lands here if "inc" is 1 byte */
+					: "=r"(sse)
+					: "r"(sse)
+				);
+
+				sigaction(SIGILL, &sigill_save, NULL);
+			}
+
+			if(!sse)
+				info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+#elif defined(_MSC_VER)
+# ifdef USE_TRY_CATCH_FLAVOR
+			_try {
+				__asm {
+#  if _MSC_VER <= 1200
+					/* VC6 assembler doesn't know SSE, have to emit bytecode instead */
+					_emit 0x0F
+					_emit 0x57
+					_emit 0xC0
+#  else
+					xorps xmm0,xmm0
+#  endif
+				}
+			}
+			_except(EXCEPTION_EXECUTE_HANDLER) {
+				if (_exception_code() == STATUS_ILLEGAL_INSTRUCTION)
+					info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+			}
+# else
+			int sse = 0;
+			LPTOP_LEVEL_EXCEPTION_FILTER save = SetUnhandledExceptionFilter(sigill_handler_sse_os);
+			/* see GCC version above for explanation */
+			/*  http://msdn2.microsoft.com/en-us/library/4ks26t93.aspx */
+			/*  http://www.codeproject.com/cpp/gccasm.asp */
+			/*  http://www.hick.org/~mmiller/msvc_inline_asm.html */
+			__asm {
+#  if _MSC_VER <= 1200
+				/* VC6 assembler doesn't know SSE, have to emit bytecode instead */
+				_emit 0x0F
+				_emit 0x57
+				_emit 0xC0
+#  else
+				xorps xmm0,xmm0
+#  endif
+				inc sse
+				nop
+				nop
+				nop
+				nop
+				nop
+				nop
+				nop
+				nop
+				nop
+			}
+			SetUnhandledExceptionFilter(save);
+			if(!sse)
+				info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+# endif
+#else
+			/* no way to test, disable to be safe */
+			info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+#endif
+#ifdef DEBUG
+		fprintf(stderr, "  SSE OS sup . %c\n", info->data.ia32.sse     ? 'Y' : 'n');
+#endif
+
+		}
+	}
+#else
+	info->use_asm = false;
+#endif
+
+/*
+ * PPC-specific
+ */
+#elif defined FLAC__CPU_PPC
+	info->type = FLAC__CPUINFO_TYPE_PPC;
+# if !defined FLAC__NO_ASM
+	info->use_asm = true;
+#  ifdef FLAC__USE_ALTIVEC
+#   if defined FLAC__SYS_DARWIN
+	{
+		int val = 0, mib[2] = { CTL_HW, HW_VECTORUNIT };
+		size_t len = sizeof(val);
+		info->data.ppc.altivec = !(sysctl(mib, 2, &val, &len, NULL, 0) || !val);
+	}
+	{
+		host_basic_info_data_t hostInfo;
+		mach_msg_type_number_t infoCount;
+
+		infoCount = HOST_BASIC_INFO_COUNT;
+		host_info(mach_host_self(), HOST_BASIC_INFO, (host_info_t)&hostInfo, &infoCount);
+
+		info->data.ppc.ppc64 = (hostInfo.cpu_type == CPU_TYPE_POWERPC) && (hostInfo.cpu_subtype == CPU_SUBTYPE_POWERPC_970);
+	}
+#   else /* FLAC__USE_ALTIVEC && !FLAC__SYS_DARWIN */
+	{
+		/* no Darwin, do it the brute-force way */
+		/* @@@@@@ this is not thread-safe; replace with SSE OS method above or remove */
+		info->data.ppc.altivec = 0;
+		info->data.ppc.ppc64 = 0;
+
+		signal (SIGILL, sigill_handler);
+		canjump = 0;
+		if (!sigsetjmp (jmpbuf, 1)) {
+			canjump = 1;
+
+			asm volatile (
+				"mtspr 256, %0\n\t"
+				"vand %%v0, %%v0, %%v0"
+				:
+				: "r" (-1)
+			);
+
+			info->data.ppc.altivec = 1;
+		}
+		canjump = 0;
+		if (!sigsetjmp (jmpbuf, 1)) {
+			int x = 0;
+			canjump = 1;
+
+			/* PPC64 hardware implements the cntlzd instruction */
+			asm volatile ("cntlzd %0, %1" : "=r" (x) : "r" (x) );
+
+			info->data.ppc.ppc64 = 1;
+		}
+		signal (SIGILL, SIG_DFL); /*@@@@@@ should save and restore old signal */
+	}
+#   endif
+#  else /* !FLAC__USE_ALTIVEC */
+	info->data.ppc.altivec = 0;
+	info->data.ppc.ppc64 = 0;
+#  endif
+# else
+	info->use_asm = false;
+# endif
+
+/*
+ * unknown CPI
+ */
+#else
+	info->type = FLAC__CPUINFO_TYPE_UNKNOWN;
+	info->use_asm = false;
+#endif
+}

source/modules/juce_audio_formats/codecs/flac/libFLAC/crc.c

@@ -0,0 +1,142 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include "include/private/crc.h"
+
+/* CRC-8, poly = x^8 + x^2 + x^1 + x^0, init = 0 */
+
+FLAC__byte const FLAC__crc8_table[256] = {
+	0x00, 0x07, 0x0E, 0x09, 0x1C, 0x1B, 0x12, 0x15,
+	0x38, 0x3F, 0x36, 0x31, 0x24, 0x23, 0x2A, 0x2D,
+	0x70, 0x77, 0x7E, 0x79, 0x6C, 0x6B, 0x62, 0x65,
+	0x48, 0x4F, 0x46, 0x41, 0x54, 0x53, 0x5A, 0x5D,
+	0xE0, 0xE7, 0xEE, 0xE9, 0xFC, 0xFB, 0xF2, 0xF5,
+	0xD8, 0xDF, 0xD6, 0xD1, 0xC4, 0xC3, 0xCA, 0xCD,
+	0x90, 0x97, 0x9E, 0x99, 0x8C, 0x8B, 0x82, 0x85,
+	0xA8, 0xAF, 0xA6, 0xA1, 0xB4, 0xB3, 0xBA, 0xBD,
+	0xC7, 0xC0, 0xC9, 0xCE, 0xDB, 0xDC, 0xD5, 0xD2,
+	0xFF, 0xF8, 0xF1, 0xF6, 0xE3, 0xE4, 0xED, 0xEA,
+	0xB7, 0xB0, 0xB9, 0xBE, 0xAB, 0xAC, 0xA5, 0xA2,
+	0x8F, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9D, 0x9A,
+	0x27, 0x20, 0x29, 0x2E, 0x3B, 0x3C, 0x35, 0x32,
+	0x1F, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0D, 0x0A,
+	0x57, 0x50, 0x59, 0x5E, 0x4B, 0x4C, 0x45, 0x42,
+	0x6F, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7D, 0x7A,
+	0x89, 0x8E, 0x87, 0x80, 0x95, 0x92, 0x9B, 0x9C,
+	0xB1, 0xB6, 0xBF, 0xB8, 0xAD, 0xAA, 0xA3, 0xA4,
+	0xF9, 0xFE, 0xF7, 0xF0, 0xE5, 0xE2, 0xEB, 0xEC,
+	0xC1, 0xC6, 0xCF, 0xC8, 0xDD, 0xDA, 0xD3, 0xD4,
+	0x69, 0x6E, 0x67, 0x60, 0x75, 0x72, 0x7B, 0x7C,
+	0x51, 0x56, 0x5F, 0x58, 0x4D, 0x4A, 0x43, 0x44,
+	0x19, 0x1E, 0x17, 0x10, 0x05, 0x02, 0x0B, 0x0C,
+	0x21, 0x26, 0x2F, 0x28, 0x3D, 0x3A, 0x33, 0x34,
+	0x4E, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5C, 0x5B,
+	0x76, 0x71, 0x78, 0x7F, 0x6A, 0x6D, 0x64, 0x63,
+	0x3E, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2C, 0x2B,
+	0x06, 0x01, 0x08, 0x0F, 0x1A, 0x1D, 0x14, 0x13,
+	0xAE, 0xA9, 0xA0, 0xA7, 0xB2, 0xB5, 0xBC, 0xBB,
+	0x96, 0x91, 0x98, 0x9F, 0x8A, 0x8D, 0x84, 0x83,
+	0xDE, 0xD9, 0xD0, 0xD7, 0xC2, 0xC5, 0xCC, 0xCB,
+	0xE6, 0xE1, 0xE8, 0xEF, 0xFA, 0xFD, 0xF4, 0xF3
+};
+
+/* CRC-16, poly = x^16 + x^15 + x^2 + x^0, init = 0 */
+
+unsigned FLAC__crc16_table[256] = {
+	0x0000,  0x8005,  0x800f,  0x000a,  0x801b,  0x001e,  0x0014,  0x8011,
+	0x8033,  0x0036,  0x003c,  0x8039,  0x0028,  0x802d,  0x8027,  0x0022,
+	0x8063,  0x0066,  0x006c,  0x8069,  0x0078,  0x807d,  0x8077,  0x0072,
+	0x0050,  0x8055,  0x805f,  0x005a,  0x804b,  0x004e,  0x0044,  0x8041,
+	0x80c3,  0x00c6,  0x00cc,  0x80c9,  0x00d8,  0x80dd,  0x80d7,  0x00d2,
+	0x00f0,  0x80f5,  0x80ff,  0x00fa,  0x80eb,  0x00ee,  0x00e4,  0x80e1,
+	0x00a0,  0x80a5,  0x80af,  0x00aa,  0x80bb,  0x00be,  0x00b4,  0x80b1,
+	0x8093,  0x0096,  0x009c,  0x8099,  0x0088,  0x808d,  0x8087,  0x0082,
+	0x8183,  0x0186,  0x018c,  0x8189,  0x0198,  0x819d,  0x8197,  0x0192,
+	0x01b0,  0x81b5,  0x81bf,  0x01ba,  0x81ab,  0x01ae,  0x01a4,  0x81a1,
+	0x01e0,  0x81e5,  0x81ef,  0x01ea,  0x81fb,  0x01fe,  0x01f4,  0x81f1,
+	0x81d3,  0x01d6,  0x01dc,  0x81d9,  0x01c8,  0x81cd,  0x81c7,  0x01c2,
+	0x0140,  0x8145,  0x814f,  0x014a,  0x815b,  0x015e,  0x0154,  0x8151,
+	0x8173,  0x0176,  0x017c,  0x8179,  0x0168,  0x816d,  0x8167,  0x0162,
+	0x8123,  0x0126,  0x012c,  0x8129,  0x0138,  0x813d,  0x8137,  0x0132,
+	0x0110,  0x8115,  0x811f,  0x011a,  0x810b,  0x010e,  0x0104,  0x8101,
+	0x8303,  0x0306,  0x030c,  0x8309,  0x0318,  0x831d,  0x8317,  0x0312,
+	0x0330,  0x8335,  0x833f,  0x033a,  0x832b,  0x032e,  0x0324,  0x8321,
+	0x0360,  0x8365,  0x836f,  0x036a,  0x837b,  0x037e,  0x0374,  0x8371,
+	0x8353,  0x0356,  0x035c,  0x8359,  0x0348,  0x834d,  0x8347,  0x0342,
+	0x03c0,  0x83c5,  0x83cf,  0x03ca,  0x83db,  0x03de,  0x03d4,  0x83d1,
+	0x83f3,  0x03f6,  0x03fc,  0x83f9,  0x03e8,  0x83ed,  0x83e7,  0x03e2,
+	0x83a3,  0x03a6,  0x03ac,  0x83a9,  0x03b8,  0x83bd,  0x83b7,  0x03b2,
+	0x0390,  0x8395,  0x839f,  0x039a,  0x838b,  0x038e,  0x0384,  0x8381,
+	0x0280,  0x8285,  0x828f,  0x028a,  0x829b,  0x029e,  0x0294,  0x8291,
+	0x82b3,  0x02b6,  0x02bc,  0x82b9,  0x02a8,  0x82ad,  0x82a7,  0x02a2,
+	0x82e3,  0x02e6,  0x02ec,  0x82e9,  0x02f8,  0x82fd,  0x82f7,  0x02f2,
+	0x02d0,  0x82d5,  0x82df,  0x02da,  0x82cb,  0x02ce,  0x02c4,  0x82c1,
+	0x8243,  0x0246,  0x024c,  0x8249,  0x0258,  0x825d,  0x8257,  0x0252,
+	0x0270,  0x8275,  0x827f,  0x027a,  0x826b,  0x026e,  0x0264,  0x8261,
+	0x0220,  0x8225,  0x822f,  0x022a,  0x823b,  0x023e,  0x0234,  0x8231,
+	0x8213,  0x0216,  0x021c,  0x8219,  0x0208,  0x820d,  0x8207,  0x0202
+};
+
+
+void FLAC__crc8_update(const FLAC__byte data, FLAC__uint8 *crc)
+{
+	*crc = FLAC__crc8_table[*crc ^ data];
+}
+
+void FLAC__crc8_update_block(const FLAC__byte *data, unsigned len, FLAC__uint8 *crc)
+{
+	while(len--)
+		*crc = FLAC__crc8_table[*crc ^ *data++];
+}
+
+FLAC__uint8 FLAC__crc8(const FLAC__byte *data, unsigned len)
+{
+	FLAC__uint8 crc = 0;
+
+	while(len--)
+		crc = FLAC__crc8_table[crc ^ *data++];
+
+	return crc;
+}
+
+unsigned FLAC__crc16(const FLAC__byte *data, unsigned len)
+{
+	unsigned crc = 0;
+
+	while(len--)
+		crc = ((crc<<8) ^ FLAC__crc16_table[(crc>>8) ^ *data++]) & 0xffff;
+
+	return crc;
+}

source/modules/juce_audio_formats/codecs/flac/libFLAC/fixed.c

@@ -0,0 +1,435 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include <math.h>
+#include <string.h>
+#include "include/private/bitmath.h"
+#include "include/private/fixed.h"
+#include "../assert.h"
+
+#ifndef M_LN2
+/* math.h in VC++ doesn't seem to have this (how Microsoft is that?) */
+#define M_LN2 0.69314718055994530942
+#endif
+
+#ifdef min
+#undef min
+#endif
+#define min(x,y) ((x) < (y)? (x) : (y))
+
+#ifdef local_abs
+#undef local_abs
+#endif
+#define local_abs(x) ((unsigned)((x)<0? -(x) : (x)))
+
+#ifdef FLAC__INTEGER_ONLY_LIBRARY
+/* rbps stands for residual bits per sample
+ *
+ *             (ln(2) * err)
+ * rbps = log  (-----------)
+ *           2 (     n     )
+ */
+static FLAC__fixedpoint local__compute_rbps_integerized(FLAC__uint32 err, FLAC__uint32 n)
+{
+	FLAC__uint32 rbps;
+	unsigned bits; /* the number of bits required to represent a number */
+	int fracbits; /* the number of bits of rbps that comprise the fractional part */
+
+	FLAC__ASSERT(sizeof(rbps) == sizeof(FLAC__fixedpoint));
+	FLAC__ASSERT(err > 0);
+	FLAC__ASSERT(n > 0);
+
+	FLAC__ASSERT(n <= FLAC__MAX_BLOCK_SIZE);
+	if(err <= n)
+		return 0;
+	/*
+	 * The above two things tell us 1) n fits in 16 bits; 2) err/n > 1.
+	 * These allow us later to know we won't lose too much precision in the
+	 * fixed-point division (err<<fracbits)/n.
+	 */
+
+	fracbits = (8*sizeof(err)) - (FLAC__bitmath_ilog2(err)+1);
+
+	err <<= fracbits;
+	err /= n;
+	/* err now holds err/n with fracbits fractional bits */
+
+	/*
+	 * Whittle err down to 16 bits max.  16 significant bits is enough for
+	 * our purposes.
+	 */
+	FLAC__ASSERT(err > 0);
+	bits = FLAC__bitmath_ilog2(err)+1;
+	if(bits > 16) {
+		err >>= (bits-16);
+		fracbits -= (bits-16);
+	}
+	rbps = (FLAC__uint32)err;
+
+	/* Multiply by fixed-point version of ln(2), with 16 fractional bits */
+	rbps *= FLAC__FP_LN2;
+	fracbits += 16;
+	FLAC__ASSERT(fracbits >= 0);
+
+	/* FLAC__fixedpoint_log2 requires fracbits%4 to be 0 */
+	{
+		const int f = fracbits & 3;
+		if(f) {
+			rbps >>= f;
+			fracbits -= f;
+		}
+	}
+
+	rbps = FLAC__fixedpoint_log2(rbps, fracbits, (unsigned)(-1));
+
+	if(rbps == 0)
+		return 0;
+
+	/*
+	 * The return value must have 16 fractional bits.  Since the whole part
+	 * of the base-2 log of a 32 bit number must fit in 5 bits, and fracbits
+	 * must be >= -3, these assertion allows us to be able to shift rbps
+	 * left if necessary to get 16 fracbits without losing any bits of the
+	 * whole part of rbps.
+	 *
+	 * There is a slight chance due to accumulated error that the whole part
+	 * will require 6 bits, so we use 6 in the assertion.  Really though as
+	 * long as it fits in 13 bits (32 - (16 - (-3))) we are fine.
+	 */
+	FLAC__ASSERT((int)FLAC__bitmath_ilog2(rbps)+1 <= fracbits + 6);
+	FLAC__ASSERT(fracbits >= -3);
+
+	/* now shift the decimal point into place */
+	if(fracbits < 16)
+		return rbps << (16-fracbits);
+	else if(fracbits > 16)
+		return rbps >> (fracbits-16);
+	else
+		return rbps;
+}
+
+static FLAC__fixedpoint local__compute_rbps_wide_integerized(FLAC__uint64 err, FLAC__uint32 n)
+{
+	FLAC__uint32 rbps;
+	unsigned bits; /* the number of bits required to represent a number */
+	int fracbits; /* the number of bits of rbps that comprise the fractional part */
+
+	FLAC__ASSERT(sizeof(rbps) == sizeof(FLAC__fixedpoint));
+	FLAC__ASSERT(err > 0);
+	FLAC__ASSERT(n > 0);
+
+	FLAC__ASSERT(n <= FLAC__MAX_BLOCK_SIZE);
+	if(err <= n)
+		return 0;
+	/*
+	 * The above two things tell us 1) n fits in 16 bits; 2) err/n > 1.
+	 * These allow us later to know we won't lose too much precision in the
+	 * fixed-point division (err<<fracbits)/n.
+	 */
+
+	fracbits = (8*sizeof(err)) - (FLAC__bitmath_ilog2_wide(err)+1);
+
+	err <<= fracbits;
+	err /= n;
+	/* err now holds err/n with fracbits fractional bits */
+
+	/*
+	 * Whittle err down to 16 bits max.  16 significant bits is enough for
+	 * our purposes.
+	 */
+	FLAC__ASSERT(err > 0);
+	bits = FLAC__bitmath_ilog2_wide(err)+1;
+	if(bits > 16) {
+		err >>= (bits-16);
+		fracbits -= (bits-16);
+	}
+	rbps = (FLAC__uint32)err;
+
+	/* Multiply by fixed-point version of ln(2), with 16 fractional bits */
+	rbps *= FLAC__FP_LN2;
+	fracbits += 16;
+	FLAC__ASSERT(fracbits >= 0);
+
+	/* FLAC__fixedpoint_log2 requires fracbits%4 to be 0 */
+	{
+		const int f = fracbits & 3;
+		if(f) {
+			rbps >>= f;
+			fracbits -= f;
+		}
+	}
+
+	rbps = FLAC__fixedpoint_log2(rbps, fracbits, (unsigned)(-1));
+
+	if(rbps == 0)
+		return 0;
+
+	/*
+	 * The return value must have 16 fractional bits.  Since the whole part
+	 * of the base-2 log of a 32 bit number must fit in 5 bits, and fracbits
+	 * must be >= -3, these assertion allows us to be able to shift rbps
+	 * left if necessary to get 16 fracbits without losing any bits of the
+	 * whole part of rbps.
+	 *
+	 * There is a slight chance due to accumulated error that the whole part
+	 * will require 6 bits, so we use 6 in the assertion.  Really though as
+	 * long as it fits in 13 bits (32 - (16 - (-3))) we are fine.
+	 */
+	FLAC__ASSERT((int)FLAC__bitmath_ilog2(rbps)+1 <= fracbits + 6);
+	FLAC__ASSERT(fracbits >= -3);
+
+	/* now shift the decimal point into place */
+	if(fracbits < 16)
+		return rbps << (16-fracbits);
+	else if(fracbits > 16)
+		return rbps >> (fracbits-16);
+	else
+		return rbps;
+}
+#endif
+
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+unsigned FLAC__fixed_compute_best_predictor(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
+#else
+unsigned FLAC__fixed_compute_best_predictor(const FLAC__int32 data[], unsigned data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
+#endif
+{
+	FLAC__int32 last_error_0 = data[-1];
+	FLAC__int32 last_error_1 = data[-1] - data[-2];
+	FLAC__int32 last_error_2 = last_error_1 - (data[-2] - data[-3]);
+	FLAC__int32 last_error_3 = last_error_2 - (data[-2] - 2*data[-3] + data[-4]);
+	FLAC__int32 error, save;
+	FLAC__uint32 total_error_0 = 0, total_error_1 = 0, total_error_2 = 0, total_error_3 = 0, total_error_4 = 0;
+	unsigned i, order;
+
+	for(i = 0; i < data_len; i++) {
+		error  = data[i]     ; total_error_0 += local_abs(error);                      save = error;
+		error -= last_error_0; total_error_1 += local_abs(error); last_error_0 = save; save = error;
+		error -= last_error_1; total_error_2 += local_abs(error); last_error_1 = save; save = error;
+		error -= last_error_2; total_error_3 += local_abs(error); last_error_2 = save; save = error;
+		error -= last_error_3; total_error_4 += local_abs(error); last_error_3 = save;
+	}
+
+	if(total_error_0 < min(min(min(total_error_1, total_error_2), total_error_3), total_error_4))
+		order = 0;
+	else if(total_error_1 < min(min(total_error_2, total_error_3), total_error_4))
+		order = 1;
+	else if(total_error_2 < min(total_error_3, total_error_4))
+		order = 2;
+	else if(total_error_3 < total_error_4)
+		order = 3;
+	else
+		order = 4;
+
+	/* Estimate the expected number of bits per residual signal sample. */
+	/* 'total_error*' is linearly related to the variance of the residual */
+	/* signal, so we use it directly to compute E(|x|) */
+	FLAC__ASSERT(data_len > 0 || total_error_0 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_1 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_2 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_3 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_4 == 0);
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+	residual_bits_per_sample[0] = (FLAC__float)((total_error_0 > 0) ? log(M_LN2 * (FLAC__double)total_error_0 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[1] = (FLAC__float)((total_error_1 > 0) ? log(M_LN2 * (FLAC__double)total_error_1 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[2] = (FLAC__float)((total_error_2 > 0) ? log(M_LN2 * (FLAC__double)total_error_2 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[3] = (FLAC__float)((total_error_3 > 0) ? log(M_LN2 * (FLAC__double)total_error_3 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[4] = (FLAC__float)((total_error_4 > 0) ? log(M_LN2 * (FLAC__double)total_error_4 / (FLAC__double)data_len) / M_LN2 : 0.0);
+#else
+	residual_bits_per_sample[0] = (total_error_0 > 0) ? local__compute_rbps_integerized(total_error_0, data_len) : 0;
+	residual_bits_per_sample[1] = (total_error_1 > 0) ? local__compute_rbps_integerized(total_error_1, data_len) : 0;
+	residual_bits_per_sample[2] = (total_error_2 > 0) ? local__compute_rbps_integerized(total_error_2, data_len) : 0;
+	residual_bits_per_sample[3] = (total_error_3 > 0) ? local__compute_rbps_integerized(total_error_3, data_len) : 0;
+	residual_bits_per_sample[4] = (total_error_4 > 0) ? local__compute_rbps_integerized(total_error_4, data_len) : 0;
+#endif
+
+	return order;
+}
+
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+unsigned FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
+#else
+unsigned FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], unsigned data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
+#endif
+{
+	FLAC__int32 last_error_0 = data[-1];
+	FLAC__int32 last_error_1 = data[-1] - data[-2];
+	FLAC__int32 last_error_2 = last_error_1 - (data[-2] - data[-3]);
+	FLAC__int32 last_error_3 = last_error_2 - (data[-2] - 2*data[-3] + data[-4]);
+	FLAC__int32 error, save;
+	/* total_error_* are 64-bits to avoid overflow when encoding
+	 * erratic signals when the bits-per-sample and blocksize are
+	 * large.
+	 */
+	FLAC__uint64 total_error_0 = 0, total_error_1 = 0, total_error_2 = 0, total_error_3 = 0, total_error_4 = 0;
+	unsigned i, order;
+
+	for(i = 0; i < data_len; i++) {
+		error  = data[i]     ; total_error_0 += local_abs(error);                      save = error;
+		error -= last_error_0; total_error_1 += local_abs(error); last_error_0 = save; save = error;
+		error -= last_error_1; total_error_2 += local_abs(error); last_error_1 = save; save = error;
+		error -= last_error_2; total_error_3 += local_abs(error); last_error_2 = save; save = error;
+		error -= last_error_3; total_error_4 += local_abs(error); last_error_3 = save;
+	}
+
+	if(total_error_0 < min(min(min(total_error_1, total_error_2), total_error_3), total_error_4))
+		order = 0;
+	else if(total_error_1 < min(min(total_error_2, total_error_3), total_error_4))
+		order = 1;
+	else if(total_error_2 < min(total_error_3, total_error_4))
+		order = 2;
+	else if(total_error_3 < total_error_4)
+		order = 3;
+	else
+		order = 4;
+
+	/* Estimate the expected number of bits per residual signal sample. */
+	/* 'total_error*' is linearly related to the variance of the residual */
+	/* signal, so we use it directly to compute E(|x|) */
+	FLAC__ASSERT(data_len > 0 || total_error_0 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_1 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_2 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_3 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_4 == 0);
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+#if defined _MSC_VER || defined __MINGW32__
+	/* with MSVC you have to spoon feed it the casting */
+	residual_bits_per_sample[0] = (FLAC__float)((total_error_0 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_0 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[1] = (FLAC__float)((total_error_1 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_1 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[2] = (FLAC__float)((total_error_2 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_2 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[3] = (FLAC__float)((total_error_3 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_3 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[4] = (FLAC__float)((total_error_4 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_4 / (FLAC__double)data_len) / M_LN2 : 0.0);
+#else
+	residual_bits_per_sample[0] = (FLAC__float)((total_error_0 > 0) ? log(M_LN2 * (FLAC__double)total_error_0 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[1] = (FLAC__float)((total_error_1 > 0) ? log(M_LN2 * (FLAC__double)total_error_1 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[2] = (FLAC__float)((total_error_2 > 0) ? log(M_LN2 * (FLAC__double)total_error_2 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[3] = (FLAC__float)((total_error_3 > 0) ? log(M_LN2 * (FLAC__double)total_error_3 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[4] = (FLAC__float)((total_error_4 > 0) ? log(M_LN2 * (FLAC__double)total_error_4 / (FLAC__double)data_len) / M_LN2 : 0.0);
+#endif
+#else
+	residual_bits_per_sample[0] = (total_error_0 > 0) ? local__compute_rbps_wide_integerized(total_error_0, data_len) : 0;
+	residual_bits_per_sample[1] = (total_error_1 > 0) ? local__compute_rbps_wide_integerized(total_error_1, data_len) : 0;
+	residual_bits_per_sample[2] = (total_error_2 > 0) ? local__compute_rbps_wide_integerized(total_error_2, data_len) : 0;
+	residual_bits_per_sample[3] = (total_error_3 > 0) ? local__compute_rbps_wide_integerized(total_error_3, data_len) : 0;
+	residual_bits_per_sample[4] = (total_error_4 > 0) ? local__compute_rbps_wide_integerized(total_error_4, data_len) : 0;
+#endif
+
+	return order;
+}
+
+void FLAC__fixed_compute_residual(const FLAC__int32 data[], unsigned data_len, unsigned order, FLAC__int32 residual[])
+{
+	const int idata_len = (int)data_len;
+	int i;
+
+	switch(order) {
+		case 0:
+			FLAC__ASSERT(sizeof(residual[0]) == sizeof(data[0]));
+			memcpy(residual, data, sizeof(residual[0])*data_len);
+			break;
+		case 1:
+			for(i = 0; i < idata_len; i++)
+				residual[i] = data[i] - data[i-1];
+			break;
+		case 2:
+			for(i = 0; i < idata_len; i++)
+#if 1 /* OPT: may be faster with some compilers on some systems */
+				residual[i] = data[i] - (data[i-1] << 1) + data[i-2];
+#else
+				residual[i] = data[i] - 2*data[i-1] + data[i-2];
+#endif
+			break;
+		case 3:
+			for(i = 0; i < idata_len; i++)
+#if 1 /* OPT: may be faster with some compilers on some systems */
+				residual[i] = data[i] - (((data[i-1]-data[i-2])<<1) + (data[i-1]-data[i-2])) - data[i-3];
+#else
+				residual[i] = data[i] - 3*data[i-1] + 3*data[i-2] - data[i-3];
+#endif
+			break;
+		case 4:
+			for(i = 0; i < idata_len; i++)
+#if 1 /* OPT: may be faster with some compilers on some systems */
+				residual[i] = data[i] - ((data[i-1]+data[i-3])<<2) + ((data[i-2]<<2) + (data[i-2]<<1)) + data[i-4];
+#else
+				residual[i] = data[i] - 4*data[i-1] + 6*data[i-2] - 4*data[i-3] + data[i-4];
+#endif
+			break;
+		default:
+			FLAC__ASSERT(0);
+	}
+}
+
+void FLAC__fixed_restore_signal(const FLAC__int32 residual[], unsigned data_len, unsigned order, FLAC__int32 data[])
+{
+	int i, idata_len = (int)data_len;
+
+	switch(order) {
+		case 0:
+			FLAC__ASSERT(sizeof(residual[0]) == sizeof(data[0]));
+			memcpy(data, residual, sizeof(residual[0])*data_len);
+			break;
+		case 1:
+			for(i = 0; i < idata_len; i++)
+				data[i] = residual[i] + data[i-1];
+			break;
+		case 2:
+			for(i = 0; i < idata_len; i++)
+#if 1 /* OPT: may be faster with some compilers on some systems */
+				data[i] = residual[i] + (data[i-1]<<1) - data[i-2];
+#else
+				data[i] = residual[i] + 2*data[i-1] - data[i-2];
+#endif
+			break;
+		case 3:
+			for(i = 0; i < idata_len; i++)
+#if 1 /* OPT: may be faster with some compilers on some systems */
+				data[i] = residual[i] + (((data[i-1]-data[i-2])<<1) + (data[i-1]-data[i-2])) + data[i-3];
+#else
+				data[i] = residual[i] + 3*data[i-1] - 3*data[i-2] + data[i-3];
+#endif
+			break;
+		case 4:
+			for(i = 0; i < idata_len; i++)
+#if 1 /* OPT: may be faster with some compilers on some systems */
+				data[i] = residual[i] + ((data[i-1]+data[i-3])<<2) - ((data[i-2]<<2) + (data[i-2]<<1)) - data[i-4];
+#else
+				data[i] = residual[i] + 4*data[i-1] - 6*data[i-2] + 4*data[i-3] - data[i-4];
+#endif
+			break;
+		default:
+			FLAC__ASSERT(0);
+	}
+}

source/modules/juce_audio_formats/codecs/flac/libFLAC/float.c

@@ -0,0 +1,308 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include "../assert.h"
+
+#include "include/private/float.h"
+
+#ifdef FLAC__INTEGER_ONLY_LIBRARY
+
+/* adjust for compilers that can't understand using LLU suffix for uint64_t literals */
+#ifdef _MSC_VER
+#define FLAC__U64L(x) x
+#else
+#define FLAC__U64L(x) x##LLU
+#endif
+
+const FLAC__fixedpoint FLAC__FP_ZERO = 0;
+const FLAC__fixedpoint FLAC__FP_ONE_HALF = 0x00008000;
+const FLAC__fixedpoint FLAC__FP_ONE = 0x00010000;
+const FLAC__fixedpoint FLAC__FP_LN2 = 45426;
+const FLAC__fixedpoint FLAC__FP_E = 178145;
+
+/* Lookup tables for Knuth's logarithm algorithm */
+#define LOG2_LOOKUP_PRECISION 16
+static const FLAC__uint32 log2_lookup[][LOG2_LOOKUP_PRECISION] = {
+	{
+		/*
+		 * 0 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00000001,
+		/* lg(4/3) = */ 0x00000000,
+		/* lg(8/7) = */ 0x00000000,
+		/* lg(16/15) = */ 0x00000000,
+		/* lg(32/31) = */ 0x00000000,
+		/* lg(64/63) = */ 0x00000000,
+		/* lg(128/127) = */ 0x00000000,
+		/* lg(256/255) = */ 0x00000000,
+		/* lg(512/511) = */ 0x00000000,
+		/* lg(1024/1023) = */ 0x00000000,
+		/* lg(2048/2047) = */ 0x00000000,
+		/* lg(4096/4095) = */ 0x00000000,
+		/* lg(8192/8191) = */ 0x00000000,
+		/* lg(16384/16383) = */ 0x00000000,
+		/* lg(32768/32767) = */ 0x00000000
+	},
+	{
+		/*
+		 * 4 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00000010,
+		/* lg(4/3) = */ 0x00000007,
+		/* lg(8/7) = */ 0x00000003,
+		/* lg(16/15) = */ 0x00000001,
+		/* lg(32/31) = */ 0x00000001,
+		/* lg(64/63) = */ 0x00000000,
+		/* lg(128/127) = */ 0x00000000,
+		/* lg(256/255) = */ 0x00000000,
+		/* lg(512/511) = */ 0x00000000,
+		/* lg(1024/1023) = */ 0x00000000,
+		/* lg(2048/2047) = */ 0x00000000,
+		/* lg(4096/4095) = */ 0x00000000,
+		/* lg(8192/8191) = */ 0x00000000,
+		/* lg(16384/16383) = */ 0x00000000,
+		/* lg(32768/32767) = */ 0x00000000
+	},
+	{
+		/*
+		 * 8 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00000100,
+		/* lg(4/3) = */ 0x0000006a,
+		/* lg(8/7) = */ 0x00000031,
+		/* lg(16/15) = */ 0x00000018,
+		/* lg(32/31) = */ 0x0000000c,
+		/* lg(64/63) = */ 0x00000006,
+		/* lg(128/127) = */ 0x00000003,
+		/* lg(256/255) = */ 0x00000001,
+		/* lg(512/511) = */ 0x00000001,
+		/* lg(1024/1023) = */ 0x00000000,
+		/* lg(2048/2047) = */ 0x00000000,
+		/* lg(4096/4095) = */ 0x00000000,
+		/* lg(8192/8191) = */ 0x00000000,
+		/* lg(16384/16383) = */ 0x00000000,
+		/* lg(32768/32767) = */ 0x00000000
+	},
+	{
+		/*
+		 * 12 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00001000,
+		/* lg(4/3) = */ 0x000006a4,
+		/* lg(8/7) = */ 0x00000315,
+		/* lg(16/15) = */ 0x0000017d,
+		/* lg(32/31) = */ 0x000000bc,
+		/* lg(64/63) = */ 0x0000005d,
+		/* lg(128/127) = */ 0x0000002e,
+		/* lg(256/255) = */ 0x00000017,
+		/* lg(512/511) = */ 0x0000000c,
+		/* lg(1024/1023) = */ 0x00000006,
+		/* lg(2048/2047) = */ 0x00000003,
+		/* lg(4096/4095) = */ 0x00000001,
+		/* lg(8192/8191) = */ 0x00000001,
+		/* lg(16384/16383) = */ 0x00000000,
+		/* lg(32768/32767) = */ 0x00000000
+	},
+	{
+		/*
+		 * 16 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00010000,
+		/* lg(4/3) = */ 0x00006a40,
+		/* lg(8/7) = */ 0x00003151,
+		/* lg(16/15) = */ 0x000017d6,
+		/* lg(32/31) = */ 0x00000bba,
+		/* lg(64/63) = */ 0x000005d1,
+		/* lg(128/127) = */ 0x000002e6,
+		/* lg(256/255) = */ 0x00000172,
+		/* lg(512/511) = */ 0x000000b9,
+		/* lg(1024/1023) = */ 0x0000005c,
+		/* lg(2048/2047) = */ 0x0000002e,
+		/* lg(4096/4095) = */ 0x00000017,
+		/* lg(8192/8191) = */ 0x0000000c,
+		/* lg(16384/16383) = */ 0x00000006,
+		/* lg(32768/32767) = */ 0x00000003
+	},
+	{
+		/*
+		 * 20 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00100000,
+		/* lg(4/3) = */ 0x0006a3fe,
+		/* lg(8/7) = */ 0x00031513,
+		/* lg(16/15) = */ 0x00017d60,
+		/* lg(32/31) = */ 0x0000bb9d,
+		/* lg(64/63) = */ 0x00005d10,
+		/* lg(128/127) = */ 0x00002e59,
+		/* lg(256/255) = */ 0x00001721,
+		/* lg(512/511) = */ 0x00000b8e,
+		/* lg(1024/1023) = */ 0x000005c6,
+		/* lg(2048/2047) = */ 0x000002e3,
+		/* lg(4096/4095) = */ 0x00000171,
+		/* lg(8192/8191) = */ 0x000000b9,
+		/* lg(16384/16383) = */ 0x0000005c,
+		/* lg(32768/32767) = */ 0x0000002e
+	},
+	{
+		/*
+		 * 24 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x01000000,
+		/* lg(4/3) = */ 0x006a3fe6,
+		/* lg(8/7) = */ 0x00315130,
+		/* lg(16/15) = */ 0x0017d605,
+		/* lg(32/31) = */ 0x000bb9ca,
+		/* lg(64/63) = */ 0x0005d0fc,
+		/* lg(128/127) = */ 0x0002e58f,
+		/* lg(256/255) = */ 0x0001720e,
+		/* lg(512/511) = */ 0x0000b8d8,
+		/* lg(1024/1023) = */ 0x00005c61,
+		/* lg(2048/2047) = */ 0x00002e2d,
+		/* lg(4096/4095) = */ 0x00001716,
+		/* lg(8192/8191) = */ 0x00000b8b,
+		/* lg(16384/16383) = */ 0x000005c5,
+		/* lg(32768/32767) = */ 0x000002e3
+	},
+	{
+		/*
+		 * 28 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x10000000,
+		/* lg(4/3) = */ 0x06a3fe5c,
+		/* lg(8/7) = */ 0x03151301,
+		/* lg(16/15) = */ 0x017d6049,
+		/* lg(32/31) = */ 0x00bb9ca6,
+		/* lg(64/63) = */ 0x005d0fba,
+		/* lg(128/127) = */ 0x002e58f7,
+		/* lg(256/255) = */ 0x001720da,
+		/* lg(512/511) = */ 0x000b8d87,
+		/* lg(1024/1023) = */ 0x0005c60b,
+		/* lg(2048/2047) = */ 0x0002e2d7,
+		/* lg(4096/4095) = */ 0x00017160,
+		/* lg(8192/8191) = */ 0x0000b8ad,
+		/* lg(16384/16383) = */ 0x00005c56,
+		/* lg(32768/32767) = */ 0x00002e2b
+	}
+};
+
+#if 0
+static const FLAC__uint64 log2_lookup_wide[] = {
+	{
+		/*
+		 * 32 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ FLAC__U64L(0x100000000),
+		/* lg(4/3) = */ FLAC__U64L(0x6a3fe5c6),
+		/* lg(8/7) = */ FLAC__U64L(0x31513015),
+		/* lg(16/15) = */ FLAC__U64L(0x17d60497),
+		/* lg(32/31) = */ FLAC__U64L(0x0bb9ca65),
+		/* lg(64/63) = */ FLAC__U64L(0x05d0fba2),
+		/* lg(128/127) = */ FLAC__U64L(0x02e58f74),
+		/* lg(256/255) = */ FLAC__U64L(0x01720d9c),
+		/* lg(512/511) = */ FLAC__U64L(0x00b8d875),
+		/* lg(1024/1023) = */ FLAC__U64L(0x005c60aa),
+		/* lg(2048/2047) = */ FLAC__U64L(0x002e2d72),
+		/* lg(4096/4095) = */ FLAC__U64L(0x00171600),
+		/* lg(8192/8191) = */ FLAC__U64L(0x000b8ad2),
+		/* lg(16384/16383) = */ FLAC__U64L(0x0005c55d),
+		/* lg(32768/32767) = */ FLAC__U64L(0x0002e2ac)
+	},
+	{
+		/*
+		 * 48 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ FLAC__U64L(0x1000000000000),
+		/* lg(4/3) = */ FLAC__U64L(0x6a3fe5c60429),
+		/* lg(8/7) = */ FLAC__U64L(0x315130157f7a),
+		/* lg(16/15) = */ FLAC__U64L(0x17d60496cfbb),
+		/* lg(32/31) = */ FLAC__U64L(0xbb9ca64ecac),
+		/* lg(64/63) = */ FLAC__U64L(0x5d0fba187cd),
+		/* lg(128/127) = */ FLAC__U64L(0x2e58f7441ee),
+		/* lg(256/255) = */ FLAC__U64L(0x1720d9c06a8),
+		/* lg(512/511) = */ FLAC__U64L(0xb8d8752173),
+		/* lg(1024/1023) = */ FLAC__U64L(0x5c60aa252e),
+		/* lg(2048/2047) = */ FLAC__U64L(0x2e2d71b0d8),
+		/* lg(4096/4095) = */ FLAC__U64L(0x1716001719),
+		/* lg(8192/8191) = */ FLAC__U64L(0xb8ad1de1b),
+		/* lg(16384/16383) = */ FLAC__U64L(0x5c55d640d),
+		/* lg(32768/32767) = */ FLAC__U64L(0x2e2abcf52)
+	}
+};
+#endif
+
+FLAC__uint32 FLAC__fixedpoint_log2(FLAC__uint32 x, unsigned fracbits, unsigned precision)
+{
+	const FLAC__uint32 ONE = (1u << fracbits);
+	const FLAC__uint32 *table = log2_lookup[fracbits >> 2];
+
+	FLAC__ASSERT(fracbits < 32);
+	FLAC__ASSERT((fracbits & 0x3) == 0);
+
+	if(x < ONE)
+		return 0;
+
+	if(precision > LOG2_LOOKUP_PRECISION)
+		precision = LOG2_LOOKUP_PRECISION;
+
+	/* Knuth's algorithm for computing logarithms, optimized for base-2 with lookup tables */
+	{
+		FLAC__uint32 y = 0;
+		FLAC__uint32 z = x >> 1, k = 1;
+		while (x > ONE && k < precision) {
+			if (x - z >= ONE) {
+				x -= z;
+				z = x >> k;
+				y += table[k];
+			}
+			else {
+				z >>= 1;
+				k++;
+			}
+		}
+		return y;
+	}
+}
+
+#endif /* defined FLAC__INTEGER_ONLY_LIBRARY */

source/modules/juce_audio_formats/codecs/flac/libFLAC/format.c

@@ -0,0 +1,598 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include <stdio.h>
+#include <stdlib.h> /* for qsort() */
+#include <string.h> /* for memset() */
+#include "../assert.h"
+#include "../format.h"
+#include "include/private/format.h"
+
+#ifndef FLaC__INLINE
+#define FLaC__INLINE
+#endif
+
+#ifdef min
+#undef min
+#endif
+#define min(a,b) ((a)<(b)?(a):(b))
+
+/* adjust for compilers that can't understand using LLU suffix for uint64_t literals */
+#ifdef _MSC_VER
+#define FLAC__U64L(x) x
+#else
+#define FLAC__U64L(x) x##LLU
+#endif
+
+/* VERSION should come from configure */
+FLAC_API const char *FLAC__VERSION_STRING = VERSION
+
+
+
+
+;
+
+#if defined _MSC_VER || defined __BORLANDC__ || defined __MINW32__
+/* yet one more hack because of MSVC6: */
+FLAC_API const char *FLAC__VENDOR_STRING = "reference libFLAC 1.2.1 20070917";
+#else
+FLAC_API const char *FLAC__VENDOR_STRING = "reference libFLAC " VERSION " 20070917";
+#endif
+
+FLAC_API const FLAC__byte FLAC__STREAM_SYNC_STRING[4] = { 'f','L','a','C' };
+FLAC_API const unsigned FLAC__STREAM_SYNC = 0x664C6143;
+FLAC_API const unsigned FLAC__STREAM_SYNC_LEN = 32; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MIN_BLOCK_SIZE_LEN = 16; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MAX_BLOCK_SIZE_LEN = 16; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MIN_FRAME_SIZE_LEN = 24; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MAX_FRAME_SIZE_LEN = 24; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_SAMPLE_RATE_LEN = 20; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_CHANNELS_LEN = 3; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN = 5; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_TOTAL_SAMPLES_LEN = 36; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MD5SUM_LEN = 128; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_APPLICATION_ID_LEN = 32; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_SEEKPOINT_SAMPLE_NUMBER_LEN = 64; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_SEEKPOINT_STREAM_OFFSET_LEN = 64; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_SEEKPOINT_FRAME_SAMPLES_LEN = 16; /* bits */
+
+FLAC_API const FLAC__uint64 FLAC__STREAM_METADATA_SEEKPOINT_PLACEHOLDER = FLAC__U64L(0xffffffffffffffff);
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_VORBIS_COMMENT_ENTRY_LENGTH_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_VORBIS_COMMENT_NUM_COMMENTS_LEN = 32; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_INDEX_OFFSET_LEN = 64; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_INDEX_NUMBER_LEN = 8; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_INDEX_RESERVED_LEN = 3*8; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_OFFSET_LEN = 64; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_NUMBER_LEN = 8; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_ISRC_LEN = 12*8; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_TYPE_LEN = 1; /* bit */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_PRE_EMPHASIS_LEN = 1; /* bit */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_RESERVED_LEN = 6+13*8; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_NUM_INDICES_LEN = 8; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_MEDIA_CATALOG_NUMBER_LEN = 128*8; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_LEAD_IN_LEN = 64; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_IS_CD_LEN = 1; /* bit */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_RESERVED_LEN = 7+258*8; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_NUM_TRACKS_LEN = 8; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_TYPE_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_MIME_TYPE_LENGTH_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_DESCRIPTION_LENGTH_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_WIDTH_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_HEIGHT_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_DEPTH_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_COLORS_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_DATA_LENGTH_LEN = 32; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_IS_LAST_LEN = 1; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_TYPE_LEN = 7; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_LENGTH_LEN = 24; /* bits */
+
+FLAC_API const unsigned FLAC__FRAME_HEADER_SYNC = 0x3ffe;
+FLAC_API const unsigned FLAC__FRAME_HEADER_SYNC_LEN = 14; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_RESERVED_LEN = 1; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_BLOCKING_STRATEGY_LEN = 1; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_BLOCK_SIZE_LEN = 4; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_SAMPLE_RATE_LEN = 4; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_CHANNEL_ASSIGNMENT_LEN = 4; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_BITS_PER_SAMPLE_LEN = 3; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_ZERO_PAD_LEN = 1; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_CRC_LEN = 8; /* bits */
+
+FLAC_API const unsigned FLAC__FRAME_FOOTER_CRC_LEN = 16; /* bits */
+
+FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_TYPE_LEN = 2; /* bits */
+FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_ORDER_LEN = 4; /* bits */
+FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_PARAMETER_LEN = 4; /* bits */
+FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_PARAMETER_LEN = 5; /* bits */
+FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_RAW_LEN = 5; /* bits */
+
+FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_ESCAPE_PARAMETER = 15; /* == (1<<FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_PARAMETER_LEN)-1 */
+FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_ESCAPE_PARAMETER = 31; /* == (1<<FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_PARAMETER_LEN)-1 */
+
+FLAC_API const char * const FLAC__EntropyCodingMethodTypeString[] = {
+	"PARTITIONED_RICE",
+	"PARTITIONED_RICE2"
+};
+
+FLAC_API const unsigned FLAC__SUBFRAME_LPC_QLP_COEFF_PRECISION_LEN = 4; /* bits */
+FLAC_API const unsigned FLAC__SUBFRAME_LPC_QLP_SHIFT_LEN = 5; /* bits */
+
+FLAC_API const unsigned FLAC__SUBFRAME_ZERO_PAD_LEN = 1; /* bits */
+FLAC_API const unsigned FLAC__SUBFRAME_TYPE_LEN = 6; /* bits */
+FLAC_API const unsigned FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN = 1; /* bits */
+
+FLAC_API const unsigned FLAC__SUBFRAME_TYPE_CONSTANT_BYTE_ALIGNED_MASK = 0x00;
+FLAC_API const unsigned FLAC__SUBFRAME_TYPE_VERBATIM_BYTE_ALIGNED_MASK = 0x02;
+FLAC_API const unsigned FLAC__SUBFRAME_TYPE_FIXED_BYTE_ALIGNED_MASK = 0x10;
+FLAC_API const unsigned FLAC__SUBFRAME_TYPE_LPC_BYTE_ALIGNED_MASK = 0x40;
+
+FLAC_API const char * const FLAC__SubframeTypeString[] = {
+	"CONSTANT",
+	"VERBATIM",
+	"FIXED",
+	"LPC"
+};
+
+FLAC_API const char * const FLAC__ChannelAssignmentString[] = {
+	"INDEPENDENT",
+	"LEFT_SIDE",
+	"RIGHT_SIDE",
+	"MID_SIDE"
+};
+
+FLAC_API const char * const FLAC__FrameNumberTypeString[] = {
+	"FRAME_NUMBER_TYPE_FRAME_NUMBER",
+	"FRAME_NUMBER_TYPE_SAMPLE_NUMBER"
+};
+
+FLAC_API const char * const FLAC__MetadataTypeString[] = {
+	"STREAMINFO",
+	"PADDING",
+	"APPLICATION",
+	"SEEKTABLE",
+	"VORBIS_COMMENT",
+	"CUESHEET",
+	"PICTURE"
+};
+
+FLAC_API const char * const FLAC__StreamMetadata_Picture_TypeString[] = {
+	"Other",
+	"32x32 pixels 'file icon' (PNG only)",
+	"Other file icon",
+	"Cover (front)",
+	"Cover (back)",
+	"Leaflet page",
+	"Media (e.g. label side of CD)",
+	"Lead artist/lead performer/soloist",
+	"Artist/performer",
+	"Conductor",
+	"Band/Orchestra",
+	"Composer",
+	"Lyricist/text writer",
+	"Recording Location",
+	"During recording",
+	"During performance",
+	"Movie/video screen capture",
+	"A bright coloured fish",
+	"Illustration",
+	"Band/artist logotype",
+	"Publisher/Studio logotype"
+};
+
+FLAC_API FLAC__bool FLAC__format_sample_rate_is_valid(unsigned sample_rate)
+{
+	if(sample_rate == 0 || sample_rate > FLAC__MAX_SAMPLE_RATE) {
+		return false;
+	}
+	else
+		return true;
+}
+
+FLAC_API FLAC__bool FLAC__format_sample_rate_is_subset(unsigned sample_rate)
+{
+	if(
+		!FLAC__format_sample_rate_is_valid(sample_rate) ||
+		(
+			sample_rate >= (1u << 16) &&
+			!(sample_rate % 1000 == 0 || sample_rate % 10 == 0)
+		)
+	) {
+		return false;
+	}
+	else
+		return true;
+}
+
+/* @@@@ add to unit tests; it is already indirectly tested by the metadata_object tests */
+FLAC_API FLAC__bool FLAC__format_seektable_is_legal(const FLAC__StreamMetadata_SeekTable *seek_table)
+{
+	unsigned i;
+	FLAC__uint64 prev_sample_number = 0;
+	FLAC__bool got_prev = false;
+
+	FLAC__ASSERT(0 != seek_table);
+
+	for(i = 0; i < seek_table->num_points; i++) {
+		if(got_prev) {
+			if(
+				seek_table->points[i].sample_number != FLAC__STREAM_METADATA_SEEKPOINT_PLACEHOLDER &&
+				seek_table->points[i].sample_number <= prev_sample_number
+			)
+				return false;
+		}
+		prev_sample_number = seek_table->points[i].sample_number;
+		got_prev = true;
+	}
+
+	return true;
+}
+
+/* used as the sort predicate for qsort() */
+static int JUCE_CDECL seekpoint_compare_(const FLAC__StreamMetadata_SeekPoint *l, const FLAC__StreamMetadata_SeekPoint *r)
+{
+	/* we don't just 'return l->sample_number - r->sample_number' since the result (FLAC__int64) might overflow an 'int' */
+	if(l->sample_number == r->sample_number)
+		return 0;
+	else if(l->sample_number < r->sample_number)
+		return -1;
+	else
+		return 1;
+}
+
+/* @@@@ add to unit tests; it is already indirectly tested by the metadata_object tests */
+FLAC_API unsigned FLAC__format_seektable_sort(FLAC__StreamMetadata_SeekTable *seek_table)
+{
+	unsigned i, j;
+	FLAC__bool first;
+
+	FLAC__ASSERT(0 != seek_table);
+
+	/* sort the seekpoints */
+	qsort(seek_table->points, seek_table->num_points, sizeof(FLAC__StreamMetadata_SeekPoint), (int (JUCE_CDECL *)(const void *, const void *))seekpoint_compare_);
+
+	/* uniquify the seekpoints */
+	first = true;
+	for(i = j = 0; i < seek_table->num_points; i++) {
+		if(seek_table->points[i].sample_number != FLAC__STREAM_METADATA_SEEKPOINT_PLACEHOLDER) {
+			if(!first) {
+				if(seek_table->points[i].sample_number == seek_table->points[j-1].sample_number)
+					continue;
+			}
+		}
+		first = false;
+		seek_table->points[j++] = seek_table->points[i];
+	}
+
+	for(i = j; i < seek_table->num_points; i++) {
+		seek_table->points[i].sample_number = FLAC__STREAM_METADATA_SEEKPOINT_PLACEHOLDER;
+		seek_table->points[i].stream_offset = 0;
+		seek_table->points[i].frame_samples = 0;
+	}
+
+	return j;
+}
+
+/*
+ * also disallows non-shortest-form encodings, c.f.
+ *   http://www.unicode.org/versions/corrigendum1.html
+ * and a more clear explanation at the end of this section:
+ *   http://www.cl.cam.ac.uk/~mgk25/unicode.html#utf-8
+ */
+static FLaC__INLINE unsigned utf8len_(const FLAC__byte *utf8)
+{
+	FLAC__ASSERT(0 != utf8);
+	if ((utf8[0] & 0x80) == 0) {
+		return 1;
+	}
+	else if ((utf8[0] & 0xE0) == 0xC0 && (utf8[1] & 0xC0) == 0x80) {
+		if ((utf8[0] & 0xFE) == 0xC0) /* overlong sequence check */
+			return 0;
+		return 2;
+	}
+	else if ((utf8[0] & 0xF0) == 0xE0 && (utf8[1] & 0xC0) == 0x80 && (utf8[2] & 0xC0) == 0x80) {
+		if (utf8[0] == 0xE0 && (utf8[1] & 0xE0) == 0x80) /* overlong sequence check */
+			return 0;
+		/* illegal surrogates check (U+D800...U+DFFF and U+FFFE...U+FFFF) */
+		if (utf8[0] == 0xED && (utf8[1] & 0xE0) == 0xA0) /* D800-DFFF */
+			return 0;
+		if (utf8[0] == 0xEF && utf8[1] == 0xBF && (utf8[2] & 0xFE) == 0xBE) /* FFFE-FFFF */
+			return 0;
+		return 3;
+	}
+	else if ((utf8[0] & 0xF8) == 0xF0 && (utf8[1] & 0xC0) == 0x80 && (utf8[2] & 0xC0) == 0x80 && (utf8[3] & 0xC0) == 0x80) {
+		if (utf8[0] == 0xF0 && (utf8[1] & 0xF0) == 0x80) /* overlong sequence check */
+			return 0;
+		return 4;
+	}
+	else if ((utf8[0] & 0xFC) == 0xF8 && (utf8[1] & 0xC0) == 0x80 && (utf8[2] & 0xC0) == 0x80 && (utf8[3] & 0xC0) == 0x80 && (utf8[4] & 0xC0) == 0x80) {
+		if (utf8[0] == 0xF8 && (utf8[1] & 0xF8) == 0x80) /* overlong sequence check */
+			return 0;
+		return 5;
+	}
+	else if ((utf8[0] & 0xFE) == 0xFC && (utf8[1] & 0xC0) == 0x80 && (utf8[2] & 0xC0) == 0x80 && (utf8[3] & 0xC0) == 0x80 && (utf8[4] & 0xC0) == 0x80 && (utf8[5] & 0xC0) == 0x80) {
+		if (utf8[0] == 0xFC && (utf8[1] & 0xFC) == 0x80) /* overlong sequence check */
+			return 0;
+		return 6;
+	}
+	else {
+		return 0;
+	}
+}
+
+FLAC_API FLAC__bool FLAC__format_vorbiscomment_entry_name_is_legal(const char *name)
+{
+	char c;
+	for(c = *name; c; c = *(++name))
+		if(c < 0x20 || c == 0x3d || c > 0x7d)
+			return false;
+	return true;
+}
+
+FLAC_API FLAC__bool FLAC__format_vorbiscomment_entry_value_is_legal(const FLAC__byte *value, unsigned length)
+{
+	if(length == (unsigned)(-1)) {
+		while(*value) {
+			unsigned n = utf8len_(value);
+			if(n == 0)
+				return false;
+			value += n;
+		}
+	}
+	else {
+		const FLAC__byte *end = value + length;
+		while(value < end) {
+			unsigned n = utf8len_(value);
+			if(n == 0)
+				return false;
+			value += n;
+		}
+		if(value != end)
+			return false;
+	}
+	return true;
+}
+
+FLAC_API FLAC__bool FLAC__format_vorbiscomment_entry_is_legal(const FLAC__byte *entry, unsigned length)
+{
+	const FLAC__byte *s, *end;
+
+	for(s = entry, end = s + length; s < end && *s != '='; s++) {
+		if(*s < 0x20 || *s > 0x7D)
+			return false;
+	}
+	if(s == end)
+		return false;
+
+	s++; /* skip '=' */
+
+	while(s < end) {
+		unsigned n = utf8len_(s);
+		if(n == 0)
+			return false;
+		s += n;
+	}
+	if(s != end)
+		return false;
+
+	return true;
+}
+
+/* @@@@ add to unit tests; it is already indirectly tested by the metadata_object tests */
+FLAC_API FLAC__bool FLAC__format_cuesheet_is_legal(const FLAC__StreamMetadata_CueSheet *cue_sheet, FLAC__bool check_cd_da_subset, const char **violation)
+{
+	unsigned i, j;
+
+	if(check_cd_da_subset) {
+		if(cue_sheet->lead_in < 2 * 44100) {
+			if(violation) *violation = "CD-DA cue sheet must have a lead-in length of at least 2 seconds";
+			return false;
+		}
+		if(cue_sheet->lead_in % 588 != 0) {
+			if(violation) *violation = "CD-DA cue sheet lead-in length must be evenly divisible by 588 samples";
+			return false;
+		}
+	}
+
+	if(cue_sheet->num_tracks == 0) {
+		if(violation) *violation = "cue sheet must have at least one track (the lead-out)";
+		return false;
+	}
+
+	if(check_cd_da_subset && cue_sheet->tracks[cue_sheet->num_tracks-1].number != 170) {
+		if(violation) *violation = "CD-DA cue sheet must have a lead-out track number 170 (0xAA)";
+		return false;
+	}
+
+	for(i = 0; i < cue_sheet->num_tracks; i++) {
+		if(cue_sheet->tracks[i].number == 0) {
+			if(violation) *violation = "cue sheet may not have a track number 0";
+			return false;
+		}
+
+		if(check_cd_da_subset) {
+			if(!((cue_sheet->tracks[i].number >= 1 && cue_sheet->tracks[i].number <= 99) || cue_sheet->tracks[i].number == 170)) {
+				if(violation) *violation = "CD-DA cue sheet track number must be 1-99 or 170";
+				return false;
+			}
+		}
+
+		if(check_cd_da_subset && cue_sheet->tracks[i].offset % 588 != 0) {
+			if(violation) {
+				if(i == cue_sheet->num_tracks-1) /* the lead-out track... */
+					*violation = "CD-DA cue sheet lead-out offset must be evenly divisible by 588 samples";
+				else
+					*violation = "CD-DA cue sheet track offset must be evenly divisible by 588 samples";
+			}
+			return false;
+		}
+
+		if(i < cue_sheet->num_tracks - 1) {
+			if(cue_sheet->tracks[i].num_indices == 0) {
+				if(violation) *violation = "cue sheet track must have at least one index point";
+				return false;
+			}
+
+			if(cue_sheet->tracks[i].indices[0].number > 1) {
+				if(violation) *violation = "cue sheet track's first index number must be 0 or 1";
+				return false;
+			}
+		}
+
+		for(j = 0; j < cue_sheet->tracks[i].num_indices; j++) {
+			if(check_cd_da_subset && cue_sheet->tracks[i].indices[j].offset % 588 != 0) {
+				if(violation) *violation = "CD-DA cue sheet track index offset must be evenly divisible by 588 samples";
+				return false;
+			}
+
+			if(j > 0) {
+				if(cue_sheet->tracks[i].indices[j].number != cue_sheet->tracks[i].indices[j-1].number + 1) {
+					if(violation) *violation = "cue sheet track index numbers must increase by 1";
+					return false;
+				}
+			}
+		}
+	}
+
+	return true;
+}
+
+/* @@@@ add to unit tests; it is already indirectly tested by the metadata_object tests */
+FLAC_API FLAC__bool FLAC__format_picture_is_legal(const FLAC__StreamMetadata_Picture *picture, const char **violation)
+{
+	char *p;
+	FLAC__byte *b;
+
+	for(p = picture->mime_type; *p; p++) {
+		if(*p < 0x20 || *p > 0x7e) {
+			if(violation) *violation = "MIME type string must contain only printable ASCII characters (0x20-0x7e)";
+			return false;
+		}
+	}
+
+	for(b = picture->description; *b; ) {
+		unsigned n = utf8len_(b);
+		if(n == 0) {
+			if(violation) *violation = "description string must be valid UTF-8";
+			return false;
+		}
+		b += n;
+	}
+
+	return true;
+}
+
+/*
+ * These routines are private to libFLAC
+ */
+unsigned FLAC__format_get_max_rice_partition_order(unsigned blocksize, unsigned predictor_order)
+{
+	return
+		FLAC__format_get_max_rice_partition_order_from_blocksize_limited_max_and_predictor_order(
+			FLAC__format_get_max_rice_partition_order_from_blocksize(blocksize),
+			blocksize,
+			predictor_order
+		);
+}
+
+unsigned FLAC__format_get_max_rice_partition_order_from_blocksize(unsigned blocksize)
+{
+	unsigned max_rice_partition_order = 0;
+	while(!(blocksize & 1)) {
+		max_rice_partition_order++;
+		blocksize >>= 1;
+	}
+	return min(FLAC__MAX_RICE_PARTITION_ORDER, max_rice_partition_order);
+}
+
+unsigned FLAC__format_get_max_rice_partition_order_from_blocksize_limited_max_and_predictor_order(unsigned limit, unsigned blocksize, unsigned predictor_order)
+{
+	unsigned max_rice_partition_order = limit;
+
+	while(max_rice_partition_order > 0 && (blocksize >> max_rice_partition_order) <= predictor_order)
+		max_rice_partition_order--;
+
+	FLAC__ASSERT(
+		(max_rice_partition_order == 0 && blocksize >= predictor_order) ||
+		(max_rice_partition_order > 0 && blocksize >> max_rice_partition_order > predictor_order)
+	);
+
+	return max_rice_partition_order;
+}
+
+void FLAC__format_entropy_coding_method_partitioned_rice_contents_init(FLAC__EntropyCodingMethod_PartitionedRiceContents *object)
+{
+	FLAC__ASSERT(0 != object);
+
+	object->parameters = 0;
+	object->raw_bits = 0;
+	object->capacity_by_order = 0;
+}
+
+void FLAC__format_entropy_coding_method_partitioned_rice_contents_clear(FLAC__EntropyCodingMethod_PartitionedRiceContents *object)
+{
+	FLAC__ASSERT(0 != object);
+
+	if(0 != object->parameters)
+		free(object->parameters);
+	if(0 != object->raw_bits)
+		free(object->raw_bits);
+	FLAC__format_entropy_coding_method_partitioned_rice_contents_init(object);
+}
+
+FLAC__bool FLAC__format_entropy_coding_method_partitioned_rice_contents_ensure_size(FLAC__EntropyCodingMethod_PartitionedRiceContents *object, unsigned max_partition_order)
+{
+	FLAC__ASSERT(0 != object);
+
+	FLAC__ASSERT(object->capacity_by_order > 0 || (0 == object->parameters && 0 == object->raw_bits));
+
+	if(object->capacity_by_order < max_partition_order) {
+		if(0 == (object->parameters = (unsigned*)realloc(object->parameters, sizeof(unsigned)*(1 << max_partition_order))))
+			return false;
+		if(0 == (object->raw_bits = (unsigned*)realloc(object->raw_bits, sizeof(unsigned)*(1 << max_partition_order))))
+			return false;
+		memset(object->raw_bits, 0, sizeof(unsigned)*(1 << max_partition_order));
+		object->capacity_by_order = max_partition_order;
+	}
+
+	return true;
+}

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/all.h

@@ -0,0 +1,49 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__ALL_H
+#define FLAC__PRIVATE__ALL_H
+
+#include "bitmath.h"
+#include "bitreader.h"
+#include "bitwriter.h"
+#include "cpu.h"
+#include "crc.h"
+#include "fixed.h"
+#include "float.h"
+#include "format.h"
+#include "lpc.h"
+#include "md5.h"
+#include "memory.h"
+#include "metadata.h"
+#include "stream_encoder_framing.h"
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/bitmath.h

@@ -0,0 +1,42 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__BITMATH_H
+#define FLAC__PRIVATE__BITMATH_H
+
+#include "../../../ordinals.h"
+
+unsigned FLAC__bitmath_ilog2(FLAC__uint32 v);
+unsigned FLAC__bitmath_ilog2_wide(FLAC__uint64 v);
+unsigned FLAC__bitmath_silog2(int v);
+unsigned FLAC__bitmath_silog2_wide(FLAC__int64 v);
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/bitreader.h

@@ -0,0 +1,99 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__BITREADER_H
+#define FLAC__PRIVATE__BITREADER_H
+
+#include <stdio.h> /* for FILE */
+#include "../../../ordinals.h"
+#include "cpu.h"
+
+/*
+ * opaque structure definition
+ */
+struct FLAC__BitReader;
+typedef struct FLAC__BitReader FLAC__BitReader;
+
+typedef FLAC__bool (*FLAC__BitReaderReadCallback)(FLAC__byte buffer[], size_t *bytes, void *client_data);
+
+/*
+ * construction, deletion, initialization, etc functions
+ */
+FLAC__BitReader *FLAC__bitreader_new(void);
+void FLAC__bitreader_delete(FLAC__BitReader *br);
+FLAC__bool FLAC__bitreader_init(FLAC__BitReader *br, FLAC__CPUInfo cpu, FLAC__BitReaderReadCallback rcb, void *cd);
+void FLAC__bitreader_free(FLAC__BitReader *br); /* does not 'free(br)' */
+FLAC__bool FLAC__bitreader_clear(FLAC__BitReader *br);
+void FLAC__bitreader_dump(const FLAC__BitReader *br, FILE *out);
+
+/*
+ * CRC functions
+ */
+void FLAC__bitreader_reset_read_crc16(FLAC__BitReader *br, FLAC__uint16 seed);
+FLAC__uint16 FLAC__bitreader_get_read_crc16(FLAC__BitReader *br);
+
+/*
+ * info functions
+ */
+FLAC__bool FLAC__bitreader_is_consumed_byte_aligned(const FLAC__BitReader *br);
+unsigned FLAC__bitreader_bits_left_for_byte_alignment(const FLAC__BitReader *br);
+unsigned FLAC__bitreader_get_input_bits_unconsumed(const FLAC__BitReader *br);
+
+/*
+ * read functions
+ */
+
+FLAC__bool FLAC__bitreader_read_raw_uint32(FLAC__BitReader *br, FLAC__uint32 *val, unsigned bits);
+FLAC__bool FLAC__bitreader_read_raw_int32(FLAC__BitReader *br, FLAC__int32 *val, unsigned bits);
+FLAC__bool FLAC__bitreader_read_raw_uint64(FLAC__BitReader *br, FLAC__uint64 *val, unsigned bits);
+FLAC__bool FLAC__bitreader_read_uint32_little_endian(FLAC__BitReader *br, FLAC__uint32 *val); /*only for bits=32*/
+FLAC__bool FLAC__bitreader_skip_bits_no_crc(FLAC__BitReader *br, unsigned bits); /* WATCHOUT: does not CRC the skipped data! */ /*@@@@ add to unit tests */
+FLAC__bool FLAC__bitreader_skip_byte_block_aligned_no_crc(FLAC__BitReader *br, unsigned nvals); /* WATCHOUT: does not CRC the read data! */
+FLAC__bool FLAC__bitreader_read_byte_block_aligned_no_crc(FLAC__BitReader *br, FLAC__byte *val, unsigned nvals); /* WATCHOUT: does not CRC the read data! */
+FLAC__bool FLAC__bitreader_read_unary_unsigned(FLAC__BitReader *br, unsigned *val);
+FLAC__bool FLAC__bitreader_read_rice_signed(FLAC__BitReader *br, int *val, unsigned parameter);
+FLAC__bool FLAC__bitreader_read_rice_signed_block(FLAC__BitReader *br, int vals[], unsigned nvals, unsigned parameter);
+#ifndef FLAC__NO_ASM
+#  ifdef FLAC__CPU_IA32
+#    ifdef FLAC__HAS_NASM
+FLAC__bool FLAC__bitreader_read_rice_signed_block_asm_ia32_bswap(FLAC__BitReader *br, int vals[], unsigned nvals, unsigned parameter);
+#    endif
+#  endif
+#endif
+#if 0 /* UNUSED */
+FLAC__bool FLAC__bitreader_read_golomb_signed(FLAC__BitReader *br, int *val, unsigned parameter);
+FLAC__bool FLAC__bitreader_read_golomb_unsigned(FLAC__BitReader *br, unsigned *val, unsigned parameter);
+#endif
+FLAC__bool FLAC__bitreader_read_utf8_uint32(FLAC__BitReader *br, FLAC__uint32 *val, FLAC__byte *raw, unsigned *rawlen);
+FLAC__bool FLAC__bitreader_read_utf8_uint64(FLAC__BitReader *br, FLAC__uint64 *val, FLAC__byte *raw, unsigned *rawlen);
+
+FLAC__bool bitreader_read_from_client_(FLAC__BitReader *br);
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/bitwriter.h

@@ -0,0 +1,103 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__BITWRITER_H
+#define FLAC__PRIVATE__BITWRITER_H
+
+#include <stdio.h> /* for FILE */
+#include "../../../ordinals.h"
+
+/*
+ * opaque structure definition
+ */
+struct FLAC__BitWriter;
+typedef struct FLAC__BitWriter FLAC__BitWriter;
+
+/*
+ * construction, deletion, initialization, etc functions
+ */
+FLAC__BitWriter *FLAC__bitwriter_new(void);
+void FLAC__bitwriter_delete(FLAC__BitWriter *bw);
+FLAC__bool FLAC__bitwriter_init(FLAC__BitWriter *bw);
+void FLAC__bitwriter_free(FLAC__BitWriter *bw); /* does not 'free(buffer)' */
+void FLAC__bitwriter_clear(FLAC__BitWriter *bw);
+void FLAC__bitwriter_dump(const FLAC__BitWriter *bw, FILE *out);
+
+/*
+ * CRC functions
+ *
+ * non-const *bw because they have to cal FLAC__bitwriter_get_buffer()
+ */
+FLAC__bool FLAC__bitwriter_get_write_crc16(FLAC__BitWriter *bw, FLAC__uint16 *crc);
+FLAC__bool FLAC__bitwriter_get_write_crc8(FLAC__BitWriter *bw, FLAC__byte *crc);
+
+/*
+ * info functions
+ */
+FLAC__bool FLAC__bitwriter_is_byte_aligned(const FLAC__BitWriter *bw);
+unsigned FLAC__bitwriter_get_input_bits_unconsumed(const FLAC__BitWriter *bw); /* can be called anytime, returns total # of bits unconsumed */
+
+/*
+ * direct buffer access
+ *
+ * there may be no calls on the bitwriter between get and release.
+ * the bitwriter continues to own the returned buffer.
+ * before get, bitwriter MUST be byte aligned: check with FLAC__bitwriter_is_byte_aligned()
+ */
+FLAC__bool FLAC__bitwriter_get_buffer(FLAC__BitWriter *bw, const FLAC__byte **buffer, size_t *bytes);
+void FLAC__bitwriter_release_buffer(FLAC__BitWriter *bw);
+
+/*
+ * write functions
+ */
+FLAC__bool FLAC__bitwriter_write_zeroes(FLAC__BitWriter *bw, unsigned bits);
+FLAC__bool FLAC__bitwriter_write_raw_uint32(FLAC__BitWriter *bw, FLAC__uint32 val, unsigned bits);
+FLAC__bool FLAC__bitwriter_write_raw_int32(FLAC__BitWriter *bw, FLAC__int32 val, unsigned bits);
+FLAC__bool FLAC__bitwriter_write_raw_uint64(FLAC__BitWriter *bw, FLAC__uint64 val, unsigned bits);
+FLAC__bool FLAC__bitwriter_write_raw_uint32_little_endian(FLAC__BitWriter *bw, FLAC__uint32 val); /*only for bits=32*/
+FLAC__bool FLAC__bitwriter_write_byte_block(FLAC__BitWriter *bw, const FLAC__byte vals[], unsigned nvals);
+FLAC__bool FLAC__bitwriter_write_unary_unsigned(FLAC__BitWriter *bw, unsigned val);
+unsigned FLAC__bitwriter_rice_bits(FLAC__int32 val, unsigned parameter);
+#if 0 /* UNUSED */
+unsigned FLAC__bitwriter_golomb_bits_signed(int val, unsigned parameter);
+unsigned FLAC__bitwriter_golomb_bits_unsigned(unsigned val, unsigned parameter);
+#endif
+FLAC__bool FLAC__bitwriter_write_rice_signed(FLAC__BitWriter *bw, FLAC__int32 val, unsigned parameter);
+FLAC__bool FLAC__bitwriter_write_rice_signed_block(FLAC__BitWriter *bw, const FLAC__int32 *vals, unsigned nvals, unsigned parameter);
+#if 0 /* UNUSED */
+FLAC__bool FLAC__bitwriter_write_golomb_signed(FLAC__BitWriter *bw, int val, unsigned parameter);
+FLAC__bool FLAC__bitwriter_write_golomb_unsigned(FLAC__BitWriter *bw, unsigned val, unsigned parameter);
+#endif
+FLAC__bool FLAC__bitwriter_write_utf8_uint32(FLAC__BitWriter *bw, FLAC__uint32 val);
+FLAC__bool FLAC__bitwriter_write_utf8_uint64(FLAC__BitWriter *bw, FLAC__uint64 val);
+FLAC__bool FLAC__bitwriter_zero_pad_to_byte_boundary(FLAC__BitWriter *bw);
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/cpu.h

@@ -0,0 +1,88 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__CPU_H
+#define FLAC__PRIVATE__CPU_H
+
+#include "../../../ordinals.h"
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+typedef enum {
+	FLAC__CPUINFO_TYPE_IA32,
+	FLAC__CPUINFO_TYPE_PPC,
+	FLAC__CPUINFO_TYPE_UNKNOWN
+} FLAC__CPUInfo_Type;
+
+typedef struct {
+	FLAC__bool cpuid;
+	FLAC__bool bswap;
+	FLAC__bool cmov;
+	FLAC__bool mmx;
+	FLAC__bool fxsr;
+	FLAC__bool sse;
+	FLAC__bool sse2;
+	FLAC__bool sse3;
+	FLAC__bool ssse3;
+	FLAC__bool _3dnow;
+	FLAC__bool ext3dnow;
+	FLAC__bool extmmx;
+} FLAC__CPUInfo_IA32;
+
+typedef struct {
+	FLAC__bool altivec;
+	FLAC__bool ppc64;
+} FLAC__CPUInfo_PPC;
+
+typedef struct {
+	FLAC__bool use_asm;
+	FLAC__CPUInfo_Type type;
+	union {
+		FLAC__CPUInfo_IA32 ia32;
+		FLAC__CPUInfo_PPC ppc;
+	} data;
+} FLAC__CPUInfo;
+
+void FLAC__cpu_info(FLAC__CPUInfo *info);
+
+#ifndef FLAC__NO_ASM
+#ifdef FLAC__CPU_IA32
+#ifdef FLAC__HAS_NASM
+FLAC__uint32 FLAC__cpu_have_cpuid_asm_ia32(void);
+void         FLAC__cpu_info_asm_ia32(FLAC__uint32 *flags_edx, FLAC__uint32 *flags_ecx);
+FLAC__uint32 FLAC__cpu_info_extended_amd_asm_ia32(void);
+#endif
+#endif
+#endif
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/crc.h

@@ -0,0 +1,61 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__CRC_H
+#define FLAC__PRIVATE__CRC_H
+
+#include "../../../ordinals.h"
+
+/* 8 bit CRC generator, MSB shifted first
+** polynomial = x^8 + x^2 + x^1 + x^0
+** init = 0
+*/
+extern FLAC__byte const FLAC__crc8_table[256];
+#define FLAC__CRC8_UPDATE(data, crc) (crc) = FLAC__crc8_table[(crc) ^ (data)];
+void FLAC__crc8_update(const FLAC__byte data, FLAC__uint8 *crc);
+void FLAC__crc8_update_block(const FLAC__byte *data, unsigned len, FLAC__uint8 *crc);
+FLAC__uint8 FLAC__crc8(const FLAC__byte *data, unsigned len);
+
+/* 16 bit CRC generator, MSB shifted first
+** polynomial = x^16 + x^15 + x^2 + x^0
+** init = 0
+*/
+extern unsigned FLAC__crc16_table[256];
+
+#define FLAC__CRC16_UPDATE(data, crc) (((((crc)<<8) & 0xffff) ^ FLAC__crc16_table[((crc)>>8) ^ (data)]))
+/* this alternate may be faster on some systems/compilers */
+#if 0
+#define FLAC__CRC16_UPDATE(data, crc) ((((crc)<<8) ^ FLAC__crc16_table[((crc)>>8) ^ (data)]) & 0xffff)
+#endif
+
+unsigned FLAC__crc16(const FLAC__byte *data, unsigned len);
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/fixed.h

@@ -0,0 +1,97 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__FIXED_H
+#define FLAC__PRIVATE__FIXED_H
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "float.h"
+#include "format.h"
+
+/*
+ *	FLAC__fixed_compute_best_predictor()
+ *	--------------------------------------------------------------------
+ *	Compute the best fixed predictor and the expected bits-per-sample
+ *  of the residual signal for each order.  The _wide() version uses
+ *  64-bit integers which is statistically necessary when bits-per-
+ *  sample + log2(blocksize) > 30
+ *
+ *	IN data[0,data_len-1]
+ *	IN data_len
+ *	OUT residual_bits_per_sample[0,FLAC__MAX_FIXED_ORDER]
+ */
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+unsigned FLAC__fixed_compute_best_predictor(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
+# ifndef FLAC__NO_ASM
+#  ifdef FLAC__CPU_IA32
+#   ifdef FLAC__HAS_NASM
+unsigned FLAC__fixed_compute_best_predictor_asm_ia32_mmx_cmov(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
+#   endif
+#  endif
+# endif
+unsigned FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
+#else
+unsigned FLAC__fixed_compute_best_predictor(const FLAC__int32 data[], unsigned data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
+unsigned FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], unsigned data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
+#endif
+
+/*
+ *	FLAC__fixed_compute_residual()
+ *	--------------------------------------------------------------------
+ *	Compute the residual signal obtained from sutracting the predicted
+ *	signal from the original.
+ *
+ *	IN data[-order,data_len-1]        original signal (NOTE THE INDICES!)
+ *	IN data_len                       length of original signal
+ *	IN order <= FLAC__MAX_FIXED_ORDER fixed-predictor order
+ *	OUT residual[0,data_len-1]        residual signal
+ */
+void FLAC__fixed_compute_residual(const FLAC__int32 data[], unsigned data_len, unsigned order, FLAC__int32 residual[]);
+
+/*
+ *	FLAC__fixed_restore_signal()
+ *	--------------------------------------------------------------------
+ *	Restore the original signal by summing the residual and the
+ *	predictor.
+ *
+ *	IN residual[0,data_len-1]         residual signal
+ *	IN data_len                       length of original signal
+ *	IN order <= FLAC__MAX_FIXED_ORDER fixed-predictor order
+ *	*** IMPORTANT: the caller must pass in the historical samples:
+ *	IN  data[-order,-1]               previously-reconstructed historical samples
+ *	OUT data[0,data_len-1]            original signal
+ */
+void FLAC__fixed_restore_signal(const FLAC__int32 residual[], unsigned data_len, unsigned order, FLAC__int32 data[]);
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/float.h

@@ -0,0 +1,97 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__FLOAT_H
+#define FLAC__PRIVATE__FLOAT_H
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "../../../ordinals.h"
+
+/*
+ * These typedefs make it easier to ensure that integer versions of
+ * the library really only contain integer operations.  All the code
+ * in libFLAC should use FLAC__float and FLAC__double in place of
+ * float and double, and be protected by checks of the macro
+ * FLAC__INTEGER_ONLY_LIBRARY.
+ *
+ * FLAC__real is the basic floating point type used in LPC analysis.
+ */
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+typedef double FLAC__double;
+typedef float FLAC__float;
+/*
+ * WATCHOUT: changing FLAC__real will change the signatures of many
+ * functions that have assembly language equivalents and break them.
+ */
+typedef float FLAC__real;
+#else
+/*
+ * The convention for FLAC__fixedpoint is to use the upper 16 bits
+ * for the integer part and lower 16 bits for the fractional part.
+ */
+typedef FLAC__int32 FLAC__fixedpoint;
+extern const FLAC__fixedpoint FLAC__FP_ZERO;
+extern const FLAC__fixedpoint FLAC__FP_ONE_HALF;
+extern const FLAC__fixedpoint FLAC__FP_ONE;
+extern const FLAC__fixedpoint FLAC__FP_LN2;
+extern const FLAC__fixedpoint FLAC__FP_E;
+
+#define FLAC__fixedpoint_trunc(x) ((x)>>16)
+
+#define FLAC__fixedpoint_mul(x, y) ( (FLAC__fixedpoint) ( ((FLAC__int64)(x)*(FLAC__int64)(y)) >> 16 ) )
+
+#define FLAC__fixedpoint_div(x, y) ( (FLAC__fixedpoint) ( ( ((FLAC__int64)(x)<<32) / (FLAC__int64)(y) ) >> 16 ) )
+
+/*
+ *	FLAC__fixedpoint_log2()
+ *	--------------------------------------------------------------------
+ *	Returns the base-2 logarithm of the fixed-point number 'x' using an
+ *	algorithm by Knuth for x >= 1.0
+ *
+ *	'fracbits' is the number of fractional bits of 'x'.  'fracbits' must
+ *	be < 32 and evenly divisible by 4 (0 is OK but not very precise).
+ *
+ *	'precision' roughly limits the number of iterations that are done;
+ *	use (unsigned)(-1) for maximum precision.
+ *
+ *	If 'x' is less than one -- that is, x < (1<<fracbits) -- then this
+ *	function will punt and return 0.
+ *
+ *	The return value will also have 'fracbits' fractional bits.
+ */
+FLAC__uint32 FLAC__fixedpoint_log2(FLAC__uint32 x, unsigned fracbits, unsigned precision);
+
+#endif
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/format.h

@@ -0,0 +1,44 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__FORMAT_H
+#define FLAC__PRIVATE__FORMAT_H
+
+#include "../../../format.h"
+
+unsigned FLAC__format_get_max_rice_partition_order(unsigned blocksize, unsigned predictor_order);
+unsigned FLAC__format_get_max_rice_partition_order_from_blocksize(unsigned blocksize);
+unsigned FLAC__format_get_max_rice_partition_order_from_blocksize_limited_max_and_predictor_order(unsigned limit, unsigned blocksize, unsigned predictor_order);
+void FLAC__format_entropy_coding_method_partitioned_rice_contents_init(FLAC__EntropyCodingMethod_PartitionedRiceContents *object);
+void FLAC__format_entropy_coding_method_partitioned_rice_contents_clear(FLAC__EntropyCodingMethod_PartitionedRiceContents *object);
+FLAC__bool FLAC__format_entropy_coding_method_partitioned_rice_contents_ensure_size(FLAC__EntropyCodingMethod_PartitionedRiceContents *object, unsigned max_partition_order);
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/lpc.h

@@ -0,0 +1,214 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__LPC_H
+#define FLAC__PRIVATE__LPC_H
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "float.h"
+#include "../../../format.h"
+
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+
+/*
+ *	FLAC__lpc_window_data()
+ *	--------------------------------------------------------------------
+ *	Applies the given window to the data.
+ *  OPT: asm implementation
+ *
+ *	IN in[0,data_len-1]
+ *	IN window[0,data_len-1]
+ *	OUT out[0,lag-1]
+ *	IN data_len
+ */
+void FLAC__lpc_window_data(const FLAC__int32 in[], const FLAC__real window[], FLAC__real out[], unsigned data_len);
+
+/*
+ *	FLAC__lpc_compute_autocorrelation()
+ *	--------------------------------------------------------------------
+ *	Compute the autocorrelation for lags between 0 and lag-1.
+ *	Assumes data[] outside of [0,data_len-1] == 0.
+ *	Asserts that lag > 0.
+ *
+ *	IN data[0,data_len-1]
+ *	IN data_len
+ *	IN 0 < lag <= data_len
+ *	OUT autoc[0,lag-1]
+ */
+void FLAC__lpc_compute_autocorrelation(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+#ifndef FLAC__NO_ASM
+#  ifdef FLAC__CPU_IA32
+#    ifdef FLAC__HAS_NASM
+void FLAC__lpc_compute_autocorrelation_asm_ia32(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+void FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_4(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+void FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_8(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+void FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_12(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+void FLAC__lpc_compute_autocorrelation_asm_ia32_3dnow(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+#    endif
+#  endif
+#endif
+
+/*
+ *	FLAC__lpc_compute_lp_coefficients()
+ *	--------------------------------------------------------------------
+ *	Computes LP coefficients for orders 1..max_order.
+ *	Do not call if autoc[0] == 0.0.  This means the signal is zero
+ *	and there is no point in calculating a predictor.
+ *
+ *	IN autoc[0,max_order]                      autocorrelation values
+ *	IN 0 < max_order <= FLAC__MAX_LPC_ORDER    max LP order to compute
+ *	OUT lp_coeff[0,max_order-1][0,max_order-1] LP coefficients for each order
+ *	*** IMPORTANT:
+ *	*** lp_coeff[0,max_order-1][max_order,FLAC__MAX_LPC_ORDER-1] are untouched
+ *	OUT error[0,max_order-1]                   error for each order (more
+ *	                                           specifically, the variance of
+ *	                                           the error signal times # of
+ *	                                           samples in the signal)
+ *
+ *	Example: if max_order is 9, the LP coefficients for order 9 will be
+ *	         in lp_coeff[8][0,8], the LP coefficients for order 8 will be
+ *			 in lp_coeff[7][0,7], etc.
+ */
+void FLAC__lpc_compute_lp_coefficients(const FLAC__real autoc[], unsigned *max_order, FLAC__real lp_coeff[][FLAC__MAX_LPC_ORDER], FLAC__double error[]);
+
+/*
+ *	FLAC__lpc_quantize_coefficients()
+ *	--------------------------------------------------------------------
+ *	Quantizes the LP coefficients.  NOTE: precision + bits_per_sample
+ *	must be less than 32 (sizeof(FLAC__int32)*8).
+ *
+ *	IN lp_coeff[0,order-1]    LP coefficients
+ *	IN order                  LP order
+ *	IN FLAC__MIN_QLP_COEFF_PRECISION < precision
+ *	                          desired precision (in bits, including sign
+ *	                          bit) of largest coefficient
+ *	OUT qlp_coeff[0,order-1]  quantized coefficients
+ *	OUT shift                 # of bits to shift right to get approximated
+ *	                          LP coefficients.  NOTE: could be negative.
+ *	RETURN 0 => quantization OK
+ *	       1 => coefficients require too much shifting for *shift to
+ *              fit in the LPC subframe header.  'shift' is unset.
+ *         2 => coefficients are all zero, which is bad.  'shift' is
+ *              unset.
+ */
+int FLAC__lpc_quantize_coefficients(const FLAC__real lp_coeff[], unsigned order, unsigned precision, FLAC__int32 qlp_coeff[], int *shift);
+
+/*
+ *	FLAC__lpc_compute_residual_from_qlp_coefficients()
+ *	--------------------------------------------------------------------
+ *	Compute the residual signal obtained from sutracting the predicted
+ *	signal from the original.
+ *
+ *	IN data[-order,data_len-1] original signal (NOTE THE INDICES!)
+ *	IN data_len                length of original signal
+ *	IN qlp_coeff[0,order-1]    quantized LP coefficients
+ *	IN order > 0               LP order
+ *	IN lp_quantization         quantization of LP coefficients in bits
+ *	OUT residual[0,data_len-1] residual signal
+ */
+void FLAC__lpc_compute_residual_from_qlp_coefficients(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+void FLAC__lpc_compute_residual_from_qlp_coefficients_wide(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+#ifndef FLAC__NO_ASM
+#  ifdef FLAC__CPU_IA32
+#    ifdef FLAC__HAS_NASM
+void FLAC__lpc_compute_residual_from_qlp_coefficients_asm_ia32(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+void FLAC__lpc_compute_residual_from_qlp_coefficients_asm_ia32_mmx(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+#    endif
+#  endif
+#endif
+
+#endif /* !defined FLAC__INTEGER_ONLY_LIBRARY */
+
+/*
+ *	FLAC__lpc_restore_signal()
+ *	--------------------------------------------------------------------
+ *	Restore the original signal by summing the residual and the
+ *	predictor.
+ *
+ *	IN residual[0,data_len-1]  residual signal
+ *	IN data_len                length of original signal
+ *	IN qlp_coeff[0,order-1]    quantized LP coefficients
+ *	IN order > 0               LP order
+ *	IN lp_quantization         quantization of LP coefficients in bits
+ *	*** IMPORTANT: the caller must pass in the historical samples:
+ *	IN  data[-order,-1]        previously-reconstructed historical samples
+ *	OUT data[0,data_len-1]     original signal
+ */
+void FLAC__lpc_restore_signal(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
+void FLAC__lpc_restore_signal_wide(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
+#ifndef FLAC__NO_ASM
+#  ifdef FLAC__CPU_IA32
+#    ifdef FLAC__HAS_NASM
+void FLAC__lpc_restore_signal_asm_ia32(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
+void FLAC__lpc_restore_signal_asm_ia32_mmx(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
+#    endif /* FLAC__HAS_NASM */
+#  elif defined FLAC__CPU_PPC
+void FLAC__lpc_restore_signal_asm_ppc_altivec_16(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
+void FLAC__lpc_restore_signal_asm_ppc_altivec_16_order8(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
+#  endif/* FLAC__CPU_IA32 || FLAC__CPU_PPC */
+#endif /* FLAC__NO_ASM */
+
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+
+/*
+ *	FLAC__lpc_compute_expected_bits_per_residual_sample()
+ *	--------------------------------------------------------------------
+ *	Compute the expected number of bits per residual signal sample
+ *	based on the LP error (which is related to the residual variance).
+ *
+ *	IN lpc_error >= 0.0   error returned from calculating LP coefficients
+ *	IN total_samples > 0  # of samples in residual signal
+ *	RETURN                expected bits per sample
+ */
+FLAC__double FLAC__lpc_compute_expected_bits_per_residual_sample(FLAC__double lpc_error, unsigned total_samples);
+FLAC__double FLAC__lpc_compute_expected_bits_per_residual_sample_with_error_scale(FLAC__double lpc_error, FLAC__double error_scale);
+
+/*
+ *	FLAC__lpc_compute_best_order()
+ *	--------------------------------------------------------------------
+ *	Compute the best order from the array of signal errors returned
+ *	during coefficient computation.
+ *
+ *	IN lpc_error[0,max_order-1] >= 0.0  error returned from calculating LP coefficients
+ *	IN max_order > 0                    max LP order
+ *	IN total_samples > 0                # of samples in residual signal
+ *	IN overhead_bits_per_order          # of bits overhead for each increased LP order
+ *	                                    (includes warmup sample size and quantized LP coefficient)
+ *	RETURN [1,max_order]                best order
+ */
+unsigned FLAC__lpc_compute_best_order(const FLAC__double lpc_error[], unsigned max_order, unsigned total_samples, unsigned overhead_bits_per_order);
+
+#endif /* !defined FLAC__INTEGER_ONLY_LIBRARY */
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/md5.h

@@ -0,0 +1,44 @@
+#ifndef FLAC__PRIVATE__MD5_H
+#define FLAC__PRIVATE__MD5_H
+
+/*
+ * This is the header file for the MD5 message-digest algorithm.
+ * The algorithm is due to Ron Rivest.  This code was
+ * written by Colin Plumb in 1993, no copyright is claimed.
+ * This code is in the public domain; do with it what you wish.
+ *
+ * Equivalent code is available from RSA Data Security, Inc.
+ * This code has been tested against that, and is equivalent,
+ * except that you don't need to include two pages of legalese
+ * with every copy.
+ *
+ * To compute the message digest of a chunk of bytes, declare an
+ * MD5Context structure, pass it to MD5Init, call MD5Update as
+ * needed on buffers full of bytes, and then call MD5Final, which
+ * will fill a supplied 16-byte array with the digest.
+ *
+ * Changed so as no longer to depend on Colin Plumb's `usual.h'
+ * header definitions; now uses stuff from dpkg's config.h
+ *  - Ian Jackson <ijackson@nyx.cs.du.edu>.
+ * Still in the public domain.
+ *
+ * Josh Coalson: made some changes to integrate with libFLAC.
+ * Still in the public domain, with no warranty.
+ */
+
+#include "../../../ordinals.h"
+
+typedef struct {
+	FLAC__uint32 in[16];
+	FLAC__uint32 buf[4];
+	FLAC__uint32 bytes[2];
+	FLAC__byte *internal_buf;
+	size_t capacity;
+} FLAC__MD5Context;
+
+void FLAC__MD5Init(FLAC__MD5Context *context);
+void FLAC__MD5Final(FLAC__byte digest[16], FLAC__MD5Context *context);
+
+FLAC__bool FLAC__MD5Accumulate(FLAC__MD5Context *ctx, const FLAC__int32 * const signal[], unsigned channels, unsigned samples, unsigned bytes_per_sample);
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/memory.h

@@ -0,0 +1,56 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__MEMORY_H
+#define FLAC__PRIVATE__MEMORY_H
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdlib.h> /* for size_t */
+
+#include "float.h"
+#include "../../../ordinals.h" /* for FLAC__bool */
+
+/* Returns the unaligned address returned by malloc.
+ * Use free() on this address to deallocate.
+ */
+void *FLAC__memory_alloc_aligned(size_t bytes, void **aligned_address);
+FLAC__bool FLAC__memory_alloc_aligned_int32_array(unsigned elements, FLAC__int32 **unaligned_pointer, FLAC__int32 **aligned_pointer);
+FLAC__bool FLAC__memory_alloc_aligned_uint32_array(unsigned elements, FLAC__uint32 **unaligned_pointer, FLAC__uint32 **aligned_pointer);
+FLAC__bool FLAC__memory_alloc_aligned_uint64_array(unsigned elements, FLAC__uint64 **unaligned_pointer, FLAC__uint64 **aligned_pointer);
+FLAC__bool FLAC__memory_alloc_aligned_unsigned_array(unsigned elements, unsigned **unaligned_pointer, unsigned **aligned_pointer);
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+FLAC__bool FLAC__memory_alloc_aligned_real_array(unsigned elements, FLAC__real **unaligned_pointer, FLAC__real **aligned_pointer);
+#endif
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/metadata.h

@@ -0,0 +1,45 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__METADATA_H
+#define FLAC__PRIVATE__METADATA_H
+
+#include "metadata.h"
+
+/* WATCHOUT: all malloc()ed data in the block is free()ed; this may not
+ * be a consistent state (e.g. PICTURE) or equivalent to the initial
+ * state after FLAC__metadata_object_new()
+ */
+void FLAC__metadata_object_delete_data(FLAC__StreamMetadata *object);
+
+void FLAC__metadata_object_cuesheet_track_delete_data(FLAC__StreamMetadata_CueSheet_Track *object);
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/stream_encoder_framing.h

@@ -0,0 +1,45 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__STREAM_ENCODER_FRAMING_H
+#define FLAC__PRIVATE__STREAM_ENCODER_FRAMING_H
+
+#include "../../../format.h"
+#include "bitwriter.h"
+
+FLAC__bool FLAC__add_metadata_block(const FLAC__StreamMetadata *metadata, FLAC__BitWriter *bw);
+FLAC__bool FLAC__frame_add_header(const FLAC__FrameHeader *header, FLAC__BitWriter *bw);
+FLAC__bool FLAC__subframe_add_constant(const FLAC__Subframe_Constant *subframe, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw);
+FLAC__bool FLAC__subframe_add_fixed(const FLAC__Subframe_Fixed *subframe, unsigned residual_samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw);
+FLAC__bool FLAC__subframe_add_lpc(const FLAC__Subframe_LPC *subframe, unsigned residual_samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw);
+FLAC__bool FLAC__subframe_add_verbatim(const FLAC__Subframe_Verbatim *subframe, unsigned samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw);
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/window.h

@@ -0,0 +1,71 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PRIVATE__WINDOW_H
+#define FLAC__PRIVATE__WINDOW_H
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include "float.h"
+#include "../../../format.h"
+
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+
+/*
+ *	FLAC__window_*()
+ *	--------------------------------------------------------------------
+ *	Calculates window coefficients according to different apodization
+ *	functions.
+ *
+ *	OUT window[0,L-1]
+ *	IN L (number of points in window)
+ */
+void FLAC__window_bartlett(FLAC__real *window, const FLAC__int32 L);
+void FLAC__window_bartlett_hann(FLAC__real *window, const FLAC__int32 L);
+void FLAC__window_blackman(FLAC__real *window, const FLAC__int32 L);
+void FLAC__window_blackman_harris_4term_92db_sidelobe(FLAC__real *window, const FLAC__int32 L);
+void FLAC__window_connes(FLAC__real *window, const FLAC__int32 L);
+void FLAC__window_flattop(FLAC__real *window, const FLAC__int32 L);
+void FLAC__window_gauss(FLAC__real *window, const FLAC__int32 L, const FLAC__real stddev); /* 0.0 < stddev <= 0.5 */
+void FLAC__window_hamming(FLAC__real *window, const FLAC__int32 L);
+void FLAC__window_hann(FLAC__real *window, const FLAC__int32 L);
+void FLAC__window_kaiser_bessel(FLAC__real *window, const FLAC__int32 L);
+void FLAC__window_nuttall(FLAC__real *window, const FLAC__int32 L);
+void FLAC__window_rectangle(FLAC__real *window, const FLAC__int32 L);
+void FLAC__window_triangle(FLAC__real *window, const FLAC__int32 L);
+void FLAC__window_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p);
+void FLAC__window_welch(FLAC__real *window, const FLAC__int32 L);
+
+#endif /* !defined FLAC__INTEGER_ONLY_LIBRARY */
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/protected/all.h

@@ -0,0 +1,38 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PROTECTED__ALL_H
+#define FLAC__PROTECTED__ALL_H
+
+#include "stream_decoder.h"
+#include "stream_encoder.h"
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/protected/stream_decoder.h

@@ -0,0 +1,58 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PROTECTED__STREAM_DECODER_H
+#define FLAC__PROTECTED__STREAM_DECODER_H
+
+#include "../../../stream_decoder.h"
+#if FLAC__HAS_OGG
+#include "include/private/ogg_decoder_aspect.h"
+#endif
+
+typedef struct FLAC__StreamDecoderProtected {
+	FLAC__StreamDecoderState state;
+	unsigned channels;
+	FLAC__ChannelAssignment channel_assignment;
+	unsigned bits_per_sample;
+	unsigned sample_rate; /* in Hz */
+	unsigned blocksize; /* in samples (per channel) */
+	FLAC__bool md5_checking; /* if true, generate MD5 signature of decoded data and compare against signature in the STREAMINFO metadata block */
+#if FLAC__HAS_OGG
+	FLAC__OggDecoderAspect ogg_decoder_aspect;
+#endif
+} FLAC__StreamDecoderProtected;
+
+/*
+ * return the number of input bytes consumed
+ */
+unsigned FLAC__stream_decoder_get_input_bytes_unconsumed(const FLAC__StreamDecoder *decoder);
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/include/protected/stream_encoder.h

@@ -0,0 +1,110 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__PROTECTED__STREAM_ENCODER_H
+#define FLAC__PROTECTED__STREAM_ENCODER_H
+
+#include "../../../stream_encoder.h"
+#if FLAC__HAS_OGG
+#include "private/ogg_encoder_aspect.h"
+#endif
+
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+
+#include "../private/float.h"
+
+#define FLAC__MAX_APODIZATION_FUNCTIONS 32
+
+typedef enum {
+	FLAC__APODIZATION_BARTLETT,
+	FLAC__APODIZATION_BARTLETT_HANN,
+	FLAC__APODIZATION_BLACKMAN,
+	FLAC__APODIZATION_BLACKMAN_HARRIS_4TERM_92DB_SIDELOBE,
+	FLAC__APODIZATION_CONNES,
+	FLAC__APODIZATION_FLATTOP,
+	FLAC__APODIZATION_GAUSS,
+	FLAC__APODIZATION_HAMMING,
+	FLAC__APODIZATION_HANN,
+	FLAC__APODIZATION_KAISER_BESSEL,
+	FLAC__APODIZATION_NUTTALL,
+	FLAC__APODIZATION_RECTANGLE,
+	FLAC__APODIZATION_TRIANGLE,
+	FLAC__APODIZATION_TUKEY,
+	FLAC__APODIZATION_WELCH
+} FLAC__ApodizationFunction;
+
+typedef struct {
+	FLAC__ApodizationFunction type;
+	union {
+		struct {
+			FLAC__real stddev;
+		} gauss;
+		struct {
+			FLAC__real p;
+		} tukey;
+	} parameters;
+} FLAC__ApodizationSpecification;
+
+#endif // #ifndef FLAC__INTEGER_ONLY_LIBRARY
+
+typedef struct FLAC__StreamEncoderProtected {
+	FLAC__StreamEncoderState state;
+	FLAC__bool verify;
+	FLAC__bool streamable_subset;
+	FLAC__bool do_md5;
+	FLAC__bool do_mid_side_stereo;
+	FLAC__bool loose_mid_side_stereo;
+	unsigned channels;
+	unsigned bits_per_sample;
+	unsigned sample_rate;
+	unsigned blocksize;
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+	unsigned num_apodizations;
+	FLAC__ApodizationSpecification apodizations[FLAC__MAX_APODIZATION_FUNCTIONS];
+#endif
+	unsigned max_lpc_order;
+	unsigned qlp_coeff_precision;
+	FLAC__bool do_qlp_coeff_prec_search;
+	FLAC__bool do_exhaustive_model_search;
+	FLAC__bool do_escape_coding;
+	unsigned min_residual_partition_order;
+	unsigned max_residual_partition_order;
+	unsigned rice_parameter_search_dist;
+	FLAC__uint64 total_samples_estimate;
+	FLAC__StreamMetadata **metadata;
+	unsigned num_metadata_blocks;
+	FLAC__uint64 streaminfo_offset, seektable_offset, audio_offset;
+#if FLAC__HAS_OGG
+	FLAC__OggEncoderAspect ogg_encoder_aspect;
+#endif
+} FLAC__StreamEncoderProtected;
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/md5.c

@@ -0,0 +1,425 @@
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include <stdlib.h>		/* for malloc() */
+#include <string.h>		/* for memcpy() */
+
+#include "include/private/md5.h"
+#include "../alloc.h"
+
+#ifndef FLaC__INLINE
+#define FLaC__INLINE
+#endif
+
+/*
+ * This code implements the MD5 message-digest algorithm.
+ * The algorithm is due to Ron Rivest.  This code was
+ * written by Colin Plumb in 1993, no copyright is claimed.
+ * This code is in the public domain; do with it what you wish.
+ *
+ * Equivalent code is available from RSA Data Security, Inc.
+ * This code has been tested against that, and is equivalent,
+ * except that you don't need to include two pages of legalese
+ * with every copy.
+ *
+ * To compute the message digest of a chunk of bytes, declare an
+ * MD5Context structure, pass it to MD5Init, call MD5Update as
+ * needed on buffers full of bytes, and then call MD5Final, which
+ * will fill a supplied 16-byte array with the digest.
+ *
+ * Changed so as no longer to depend on Colin Plumb's `usual.h' header
+ * definitions; now uses stuff from dpkg's config.h.
+ *  - Ian Jackson <ijackson@nyx.cs.du.edu>.
+ * Still in the public domain.
+ *
+ * Josh Coalson: made some changes to integrate with libFLAC.
+ * Still in the public domain.
+ */
+
+/* The four core functions - F1 is optimized somewhat */
+
+/* #define F1(x, y, z) (x & y | ~x & z) */
+#define F1(x, y, z) (z ^ (x & (y ^ z)))
+#define F2(x, y, z) F1(z, x, y)
+#define F3(x, y, z) (x ^ y ^ z)
+#define F4(x, y, z) (y ^ (x | ~z))
+
+/* This is the central step in the MD5 algorithm. */
+#define MD5STEP(f,w,x,y,z,in,s) \
+	 (w += f(x,y,z) + in, w = (w<<s | w>>(32-s)) + x)
+
+/*
+ * The core of the MD5 algorithm, this alters an existing MD5 hash to
+ * reflect the addition of 16 longwords of new data.  MD5Update blocks
+ * the data and converts bytes into longwords for this routine.
+ */
+static void FLAC__MD5Transform(FLAC__uint32 buf[4], FLAC__uint32 const in[16])
+{
+	register FLAC__uint32 a, b, c, d;
+
+	a = buf[0];
+	b = buf[1];
+	c = buf[2];
+	d = buf[3];
+
+	MD5STEP(F1, a, b, c, d, in[0] + 0xd76aa478, 7);
+	MD5STEP(F1, d, a, b, c, in[1] + 0xe8c7b756, 12);
+	MD5STEP(F1, c, d, a, b, in[2] + 0x242070db, 17);
+	MD5STEP(F1, b, c, d, a, in[3] + 0xc1bdceee, 22);
+	MD5STEP(F1, a, b, c, d, in[4] + 0xf57c0faf, 7);
+	MD5STEP(F1, d, a, b, c, in[5] + 0x4787c62a, 12);
+	MD5STEP(F1, c, d, a, b, in[6] + 0xa8304613, 17);
+	MD5STEP(F1, b, c, d, a, in[7] + 0xfd469501, 22);
+	MD5STEP(F1, a, b, c, d, in[8] + 0x698098d8, 7);
+	MD5STEP(F1, d, a, b, c, in[9] + 0x8b44f7af, 12);
+	MD5STEP(F1, c, d, a, b, in[10] + 0xffff5bb1, 17);
+	MD5STEP(F1, b, c, d, a, in[11] + 0x895cd7be, 22);
+	MD5STEP(F1, a, b, c, d, in[12] + 0x6b901122, 7);
+	MD5STEP(F1, d, a, b, c, in[13] + 0xfd987193, 12);
+	MD5STEP(F1, c, d, a, b, in[14] + 0xa679438e, 17);
+	MD5STEP(F1, b, c, d, a, in[15] + 0x49b40821, 22);
+
+	MD5STEP(F2, a, b, c, d, in[1] + 0xf61e2562, 5);
+	MD5STEP(F2, d, a, b, c, in[6] + 0xc040b340, 9);
+	MD5STEP(F2, c, d, a, b, in[11] + 0x265e5a51, 14);
+	MD5STEP(F2, b, c, d, a, in[0] + 0xe9b6c7aa, 20);
+	MD5STEP(F2, a, b, c, d, in[5] + 0xd62f105d, 5);
+	MD5STEP(F2, d, a, b, c, in[10] + 0x02441453, 9);
+	MD5STEP(F2, c, d, a, b, in[15] + 0xd8a1e681, 14);
+	MD5STEP(F2, b, c, d, a, in[4] + 0xe7d3fbc8, 20);
+	MD5STEP(F2, a, b, c, d, in[9] + 0x21e1cde6, 5);
+	MD5STEP(F2, d, a, b, c, in[14] + 0xc33707d6, 9);
+	MD5STEP(F2, c, d, a, b, in[3] + 0xf4d50d87, 14);
+	MD5STEP(F2, b, c, d, a, in[8] + 0x455a14ed, 20);
+	MD5STEP(F2, a, b, c, d, in[13] + 0xa9e3e905, 5);
+	MD5STEP(F2, d, a, b, c, in[2] + 0xfcefa3f8, 9);
+	MD5STEP(F2, c, d, a, b, in[7] + 0x676f02d9, 14);
+	MD5STEP(F2, b, c, d, a, in[12] + 0x8d2a4c8a, 20);
+
+	MD5STEP(F3, a, b, c, d, in[5] + 0xfffa3942, 4);
+	MD5STEP(F3, d, a, b, c, in[8] + 0x8771f681, 11);
+	MD5STEP(F3, c, d, a, b, in[11] + 0x6d9d6122, 16);
+	MD5STEP(F3, b, c, d, a, in[14] + 0xfde5380c, 23);
+	MD5STEP(F3, a, b, c, d, in[1] + 0xa4beea44, 4);
+	MD5STEP(F3, d, a, b, c, in[4] + 0x4bdecfa9, 11);
+	MD5STEP(F3, c, d, a, b, in[7] + 0xf6bb4b60, 16);
+	MD5STEP(F3, b, c, d, a, in[10] + 0xbebfbc70, 23);
+	MD5STEP(F3, a, b, c, d, in[13] + 0x289b7ec6, 4);
+	MD5STEP(F3, d, a, b, c, in[0] + 0xeaa127fa, 11);
+	MD5STEP(F3, c, d, a, b, in[3] + 0xd4ef3085, 16);
+	MD5STEP(F3, b, c, d, a, in[6] + 0x04881d05, 23);
+	MD5STEP(F3, a, b, c, d, in[9] + 0xd9d4d039, 4);
+	MD5STEP(F3, d, a, b, c, in[12] + 0xe6db99e5, 11);
+	MD5STEP(F3, c, d, a, b, in[15] + 0x1fa27cf8, 16);
+	MD5STEP(F3, b, c, d, a, in[2] + 0xc4ac5665, 23);
+
+	MD5STEP(F4, a, b, c, d, in[0] + 0xf4292244, 6);
+	MD5STEP(F4, d, a, b, c, in[7] + 0x432aff97, 10);
+	MD5STEP(F4, c, d, a, b, in[14] + 0xab9423a7, 15);
+	MD5STEP(F4, b, c, d, a, in[5] + 0xfc93a039, 21);
+	MD5STEP(F4, a, b, c, d, in[12] + 0x655b59c3, 6);
+	MD5STEP(F4, d, a, b, c, in[3] + 0x8f0ccc92, 10);
+	MD5STEP(F4, c, d, a, b, in[10] + 0xffeff47d, 15);
+	MD5STEP(F4, b, c, d, a, in[1] + 0x85845dd1, 21);
+	MD5STEP(F4, a, b, c, d, in[8] + 0x6fa87e4f, 6);
+	MD5STEP(F4, d, a, b, c, in[15] + 0xfe2ce6e0, 10);
+	MD5STEP(F4, c, d, a, b, in[6] + 0xa3014314, 15);
+	MD5STEP(F4, b, c, d, a, in[13] + 0x4e0811a1, 21);
+	MD5STEP(F4, a, b, c, d, in[4] + 0xf7537e82, 6);
+	MD5STEP(F4, d, a, b, c, in[11] + 0xbd3af235, 10);
+	MD5STEP(F4, c, d, a, b, in[2] + 0x2ad7d2bb, 15);
+	MD5STEP(F4, b, c, d, a, in[9] + 0xeb86d391, 21);
+
+	buf[0] += a;
+	buf[1] += b;
+	buf[2] += c;
+	buf[3] += d;
+}
+
+#if WORDS_BIGENDIAN
+//@@@@@@ OPT: use bswap/intrinsics
+static void byteSwap(FLAC__uint32 *buf, unsigned words)
+{
+	register FLAC__uint32 x;
+	do {
+		x = *buf;
+		x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff);
+		*buf++ = (x >> 16) | (x << 16);
+	} while (--words);
+}
+static void byteSwapX16(FLAC__uint32 *buf)
+{
+	register FLAC__uint32 x;
+
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf   = (x >> 16) | (x << 16);
+}
+#else
+#define byteSwap(buf, words)
+#define byteSwapX16(buf)
+#endif
+
+/*
+ * Update context to reflect the concatenation of another buffer full
+ * of bytes.
+ */
+static void FLAC__MD5Update(FLAC__MD5Context *ctx, FLAC__byte const *buf, unsigned len)
+{
+	FLAC__uint32 t;
+
+	/* Update byte count */
+
+	t = ctx->bytes[0];
+	if ((ctx->bytes[0] = t + len) < t)
+		ctx->bytes[1]++;	/* Carry from low to high */
+
+	t = 64 - (t & 0x3f);	/* Space available in ctx->in (at least 1) */
+	if (t > len) {
+		memcpy((FLAC__byte *)ctx->in + 64 - t, buf, len);
+		return;
+	}
+	/* First chunk is an odd size */
+	memcpy((FLAC__byte *)ctx->in + 64 - t, buf, t);
+	byteSwapX16(ctx->in);
+	FLAC__MD5Transform(ctx->buf, ctx->in);
+	buf += t;
+	len -= t;
+
+	/* Process data in 64-byte chunks */
+	while (len >= 64) {
+		memcpy(ctx->in, buf, 64);
+		byteSwapX16(ctx->in);
+		FLAC__MD5Transform(ctx->buf, ctx->in);
+		buf += 64;
+		len -= 64;
+	}
+
+	/* Handle any remaining bytes of data. */
+	memcpy(ctx->in, buf, len);
+}
+
+/*
+ * Start MD5 accumulation.  Set bit count to 0 and buffer to mysterious
+ * initialization constants.
+ */
+void FLAC__MD5Init(FLAC__MD5Context *ctx)
+{
+	ctx->buf[0] = 0x67452301;
+	ctx->buf[1] = 0xefcdab89;
+	ctx->buf[2] = 0x98badcfe;
+	ctx->buf[3] = 0x10325476;
+
+	ctx->bytes[0] = 0;
+	ctx->bytes[1] = 0;
+
+	ctx->internal_buf = 0;
+	ctx->capacity = 0;
+}
+
+/*
+ * Final wrapup - pad to 64-byte boundary with the bit pattern
+ * 1 0* (64-bit count of bits processed, MSB-first)
+ */
+void FLAC__MD5Final(FLAC__byte digest[16], FLAC__MD5Context *ctx)
+{
+	int count = ctx->bytes[0] & 0x3f;	/* Number of bytes in ctx->in */
+	FLAC__byte *p = (FLAC__byte *)ctx->in + count;
+
+	/* Set the first char of padding to 0x80.  There is always room. */
+	*p++ = 0x80;
+
+	/* Bytes of padding needed to make 56 bytes (-8..55) */
+	count = 56 - 1 - count;
+
+	if (count < 0) {	/* Padding forces an extra block */
+		memset(p, 0, count + 8);
+		byteSwapX16(ctx->in);
+		FLAC__MD5Transform(ctx->buf, ctx->in);
+		p = (FLAC__byte *)ctx->in;
+		count = 56;
+	}
+	memset(p, 0, count);
+	byteSwap(ctx->in, 14);
+
+	/* Append length in bits and transform */
+	ctx->in[14] = ctx->bytes[0] << 3;
+	ctx->in[15] = ctx->bytes[1] << 3 | ctx->bytes[0] >> 29;
+	FLAC__MD5Transform(ctx->buf, ctx->in);
+
+	byteSwap(ctx->buf, 4);
+	memcpy(digest, ctx->buf, 16);
+	//memset(ctx, 0, sizeof(ctx));	/* In case it's sensitive */
+	if(0 != ctx->internal_buf) {
+		free(ctx->internal_buf);
+		ctx->internal_buf = 0;
+		ctx->capacity = 0;
+	}
+}
+
+/*
+ * Convert the incoming audio signal to a byte stream
+ */
+static void format_input_(FLAC__byte *buf, const FLAC__int32 * const signal[], unsigned channels, unsigned samples, unsigned bytes_per_sample)
+{
+	unsigned channel, sample;
+	register FLAC__int32 a_word;
+	register FLAC__byte *buf_ = buf;
+
+#if WORDS_BIGENDIAN
+#else
+	if(channels == 2 && bytes_per_sample == 2) {
+		FLAC__int16 *buf1_ = ((FLAC__int16*)buf_) + 1;
+		memcpy(buf_, signal[0], sizeof(FLAC__int32) * samples);
+		for(sample = 0; sample < samples; sample++, buf1_+=2)
+			*buf1_ = (FLAC__int16)signal[1][sample];
+	}
+	else if(channels == 1 && bytes_per_sample == 2) {
+		FLAC__int16 *buf1_ = (FLAC__int16*)buf_;
+		for(sample = 0; sample < samples; sample++)
+			*buf1_++ = (FLAC__int16)signal[0][sample];
+	}
+	else
+#endif
+	if(bytes_per_sample == 2) {
+		if(channels == 2) {
+			for(sample = 0; sample < samples; sample++) {
+				a_word = signal[0][sample];
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word;
+				a_word = signal[1][sample];
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word;
+			}
+		}
+		else if(channels == 1) {
+			for(sample = 0; sample < samples; sample++) {
+				a_word = signal[0][sample];
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word;
+			}
+		}
+		else {
+			for(sample = 0; sample < samples; sample++) {
+				for(channel = 0; channel < channels; channel++) {
+					a_word = signal[channel][sample];
+					*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+					*buf_++ = (FLAC__byte)a_word;
+				}
+			}
+		}
+	}
+	else if(bytes_per_sample == 3) {
+		if(channels == 2) {
+			for(sample = 0; sample < samples; sample++) {
+				a_word = signal[0][sample];
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word;
+				a_word = signal[1][sample];
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word;
+			}
+		}
+		else if(channels == 1) {
+			for(sample = 0; sample < samples; sample++) {
+				a_word = signal[0][sample];
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word;
+			}
+		}
+		else {
+			for(sample = 0; sample < samples; sample++) {
+				for(channel = 0; channel < channels; channel++) {
+					a_word = signal[channel][sample];
+					*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+					*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+					*buf_++ = (FLAC__byte)a_word;
+				}
+			}
+		}
+	}
+	else if(bytes_per_sample == 1) {
+		if(channels == 2) {
+			for(sample = 0; sample < samples; sample++) {
+				a_word = signal[0][sample];
+				*buf_++ = (FLAC__byte)a_word;
+				a_word = signal[1][sample];
+				*buf_++ = (FLAC__byte)a_word;
+			}
+		}
+		else if(channels == 1) {
+			for(sample = 0; sample < samples; sample++) {
+				a_word = signal[0][sample];
+				*buf_++ = (FLAC__byte)a_word;
+			}
+		}
+		else {
+			for(sample = 0; sample < samples; sample++) {
+				for(channel = 0; channel < channels; channel++) {
+					a_word = signal[channel][sample];
+					*buf_++ = (FLAC__byte)a_word;
+				}
+			}
+		}
+	}
+	else { /* bytes_per_sample == 4, maybe optimize more later */
+		for(sample = 0; sample < samples; sample++) {
+			for(channel = 0; channel < channels; channel++) {
+				a_word = signal[channel][sample];
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word;
+			}
+		}
+	}
+}
+
+/*
+ * Convert the incoming audio signal to a byte stream and FLAC__MD5Update it.
+ */
+FLAC__bool FLAC__MD5Accumulate(FLAC__MD5Context *ctx, const FLAC__int32 * const signal[], unsigned channels, unsigned samples, unsigned bytes_per_sample)
+{
+	const size_t bytes_needed = (size_t)channels * (size_t)samples * (size_t)bytes_per_sample;
+
+	/* overflow check */
+	if((size_t)channels > SIZE_MAX / (size_t)bytes_per_sample)
+		return false;
+	if((size_t)channels * (size_t)bytes_per_sample > SIZE_MAX / (size_t)samples)
+		return false;
+
+	if(ctx->capacity < bytes_needed) {
+		FLAC__byte *tmp = (FLAC__byte*)realloc(ctx->internal_buf, bytes_needed);
+		if(0 == tmp) {
+			free(ctx->internal_buf);
+			if(0 == (ctx->internal_buf = (FLAC__byte*)safe_malloc_(bytes_needed)))
+				return false;
+		}
+		ctx->internal_buf = tmp;
+		ctx->capacity = bytes_needed;
+	}
+
+	format_input_(ctx->internal_buf, signal, channels, samples, bytes_per_sample);
+
+	FLAC__MD5Update(ctx, ctx->internal_buf, bytes_needed);
+
+	return true;
+}

source/modules/juce_audio_formats/codecs/flac/libFLAC/memory.c

@@ -0,0 +1,206 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include "include/private/memory.h"
+#include "../assert.h"
+#include "../alloc.h"
+
+void *FLAC__memory_alloc_aligned(size_t bytes, void **aligned_address)
+{
+	void *x;
+
+	FLAC__ASSERT(0 != aligned_address);
+
+#ifdef FLAC__ALIGN_MALLOC_DATA
+	/* align on 32-byte (256-bit) boundary */
+	x = safe_malloc_add_2op_(bytes, /*+*/31);
+#ifdef SIZEOF_VOIDP
+#if SIZEOF_VOIDP == 4
+		/* could do  *aligned_address = x + ((unsigned) (32 - (((unsigned)x) & 31))) & 31; */
+		*aligned_address = (void*)(((unsigned)x + 31) & -32);
+#elif SIZEOF_VOIDP == 8
+		*aligned_address = (void*)(((FLAC__uint64)x + 31) & (FLAC__uint64)(-((FLAC__int64)32)));
+#else
+# error  Unsupported sizeof(void*)
+#endif
+#else
+	/* there's got to be a better way to do this right for all archs */
+	if(sizeof(void*) == sizeof(unsigned))
+		*aligned_address = (void*)(((unsigned)x + 31) & -32);
+	else if(sizeof(void*) == sizeof(FLAC__uint64))
+		*aligned_address = (void*)(((FLAC__uint64)x + 31) & (FLAC__uint64)(-((FLAC__int64)32)));
+	else
+		return 0;
+#endif
+#else
+	x = safe_malloc_(bytes);
+	*aligned_address = x;
+#endif
+	return x;
+}
+
+FLAC__bool FLAC__memory_alloc_aligned_int32_array(unsigned elements, FLAC__int32 **unaligned_pointer, FLAC__int32 **aligned_pointer)
+{
+	FLAC__int32 *pu; /* unaligned pointer */
+	union { /* union needed to comply with C99 pointer aliasing rules */
+		FLAC__int32 *pa; /* aligned pointer */
+		void        *pv; /* aligned pointer alias */
+	} u;
+
+	FLAC__ASSERT(elements > 0);
+	FLAC__ASSERT(0 != unaligned_pointer);
+	FLAC__ASSERT(0 != aligned_pointer);
+	FLAC__ASSERT(unaligned_pointer != aligned_pointer);
+
+	pu = (FLAC__int32*)FLAC__memory_alloc_aligned(sizeof(*pu) * (size_t)elements, &u.pv);
+	if(0 == pu) {
+		return false;
+	}
+	else {
+		if(*unaligned_pointer != 0)
+			free(*unaligned_pointer);
+		*unaligned_pointer = pu;
+		*aligned_pointer = u.pa;
+		return true;
+	}
+}
+
+FLAC__bool FLAC__memory_alloc_aligned_uint32_array(unsigned elements, FLAC__uint32 **unaligned_pointer, FLAC__uint32 **aligned_pointer)
+{
+	FLAC__uint32 *pu; /* unaligned pointer */
+	union { /* union needed to comply with C99 pointer aliasing rules */
+		FLAC__uint32 *pa; /* aligned pointer */
+		void         *pv; /* aligned pointer alias */
+	} u;
+
+	FLAC__ASSERT(elements > 0);
+	FLAC__ASSERT(0 != unaligned_pointer);
+	FLAC__ASSERT(0 != aligned_pointer);
+	FLAC__ASSERT(unaligned_pointer != aligned_pointer);
+
+	pu = (FLAC__uint32*)FLAC__memory_alloc_aligned(sizeof(*pu) * elements, &u.pv);
+	if(0 == pu) {
+		return false;
+	}
+	else {
+		if(*unaligned_pointer != 0)
+			free(*unaligned_pointer);
+		*unaligned_pointer = pu;
+		*aligned_pointer = u.pa;
+		return true;
+	}
+}
+
+FLAC__bool FLAC__memory_alloc_aligned_uint64_array(unsigned elements, FLAC__uint64 **unaligned_pointer, FLAC__uint64 **aligned_pointer)
+{
+	FLAC__uint64 *pu; /* unaligned pointer */
+	union { /* union needed to comply with C99 pointer aliasing rules */
+		FLAC__uint64 *pa; /* aligned pointer */
+		void         *pv; /* aligned pointer alias */
+	} u;
+
+	FLAC__ASSERT(elements > 0);
+	FLAC__ASSERT(0 != unaligned_pointer);
+	FLAC__ASSERT(0 != aligned_pointer);
+	FLAC__ASSERT(unaligned_pointer != aligned_pointer);
+
+	pu = (FLAC__uint64*)FLAC__memory_alloc_aligned(sizeof(*pu) * elements, &u.pv);
+	if(0 == pu) {
+		return false;
+	}
+	else {
+		if(*unaligned_pointer != 0)
+			free(*unaligned_pointer);
+		*unaligned_pointer = pu;
+		*aligned_pointer = u.pa;
+		return true;
+	}
+}
+
+FLAC__bool FLAC__memory_alloc_aligned_unsigned_array(unsigned elements, unsigned **unaligned_pointer, unsigned **aligned_pointer)
+{
+	unsigned *pu; /* unaligned pointer */
+	union { /* union needed to comply with C99 pointer aliasing rules */
+		unsigned *pa; /* aligned pointer */
+		void     *pv; /* aligned pointer alias */
+	} u;
+
+	FLAC__ASSERT(elements > 0);
+	FLAC__ASSERT(0 != unaligned_pointer);
+	FLAC__ASSERT(0 != aligned_pointer);
+	FLAC__ASSERT(unaligned_pointer != aligned_pointer);
+
+	pu = (unsigned*)FLAC__memory_alloc_aligned(sizeof(*pu) * elements, &u.pv);
+	if(0 == pu) {
+		return false;
+	}
+	else {
+		if(*unaligned_pointer != 0)
+			free(*unaligned_pointer);
+		*unaligned_pointer = pu;
+		*aligned_pointer = u.pa;
+		return true;
+	}
+}
+
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+
+FLAC__bool FLAC__memory_alloc_aligned_real_array(unsigned elements, FLAC__real **unaligned_pointer, FLAC__real **aligned_pointer)
+{
+	FLAC__real *pu; /* unaligned pointer */
+	union { /* union needed to comply with C99 pointer aliasing rules */
+		FLAC__real *pa; /* aligned pointer */
+		void       *pv; /* aligned pointer alias */
+	} u;
+
+	FLAC__ASSERT(elements > 0);
+	FLAC__ASSERT(0 != unaligned_pointer);
+	FLAC__ASSERT(0 != aligned_pointer);
+	FLAC__ASSERT(unaligned_pointer != aligned_pointer);
+
+	pu = (FLAC__real*)FLAC__memory_alloc_aligned(sizeof(*pu) * elements, &u.pv);
+	if(0 == pu) {
+		return false;
+	}
+	else {
+		if(*unaligned_pointer != 0)
+			free(*unaligned_pointer);
+		*unaligned_pointer = pu;
+		*aligned_pointer = u.pa;
+		return true;
+	}
+}
+
+#endif

source/modules/juce_audio_formats/codecs/flac/libFLAC/stream_encoder_framing.c

@@ -0,0 +1,553 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include <stdio.h>
+#include <string.h> /* for strlen() */
+#include "include/private/stream_encoder_framing.h"
+#include "include/private/crc.h"
+#include "../assert.h"
+
+#ifdef max
+#undef max
+#endif
+#define max(x,y) ((x)>(y)?(x):(y))
+
+static FLAC__bool add_entropy_coding_method_(FLAC__BitWriter *bw, const FLAC__EntropyCodingMethod *method);
+static FLAC__bool add_residual_partitioned_rice_(FLAC__BitWriter *bw, const FLAC__int32 residual[], const unsigned residual_samples, const unsigned predictor_order, const unsigned rice_parameters[], const unsigned raw_bits[], const unsigned partition_order, const FLAC__bool is_extended);
+
+FLAC__bool FLAC__add_metadata_block(const FLAC__StreamMetadata *metadata, FLAC__BitWriter *bw)
+{
+	unsigned i, j;
+	const unsigned vendor_string_length = (unsigned)strlen(FLAC__VENDOR_STRING);
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->is_last, FLAC__STREAM_METADATA_IS_LAST_LEN))
+		return false;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->type, FLAC__STREAM_METADATA_TYPE_LEN))
+		return false;
+
+	/*
+	 * First, for VORBIS_COMMENTs, adjust the length to reflect our vendor string
+	 */
+	i = metadata->length;
+	if(metadata->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) {
+		FLAC__ASSERT(metadata->data.vorbis_comment.vendor_string.length == 0 || 0 != metadata->data.vorbis_comment.vendor_string.entry);
+		i -= metadata->data.vorbis_comment.vendor_string.length;
+		i += vendor_string_length;
+	}
+	FLAC__ASSERT(i < (1u << FLAC__STREAM_METADATA_LENGTH_LEN));
+	if(!FLAC__bitwriter_write_raw_uint32(bw, i, FLAC__STREAM_METADATA_LENGTH_LEN))
+		return false;
+
+	switch(metadata->type) {
+		case FLAC__METADATA_TYPE_STREAMINFO:
+			FLAC__ASSERT(metadata->data.stream_info.min_blocksize < (1u << FLAC__STREAM_METADATA_STREAMINFO_MIN_BLOCK_SIZE_LEN));
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.min_blocksize, FLAC__STREAM_METADATA_STREAMINFO_MIN_BLOCK_SIZE_LEN))
+				return false;
+			FLAC__ASSERT(metadata->data.stream_info.max_blocksize < (1u << FLAC__STREAM_METADATA_STREAMINFO_MAX_BLOCK_SIZE_LEN));
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.max_blocksize, FLAC__STREAM_METADATA_STREAMINFO_MAX_BLOCK_SIZE_LEN))
+				return false;
+			FLAC__ASSERT(metadata->data.stream_info.min_framesize < (1u << FLAC__STREAM_METADATA_STREAMINFO_MIN_FRAME_SIZE_LEN));
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.min_framesize, FLAC__STREAM_METADATA_STREAMINFO_MIN_FRAME_SIZE_LEN))
+				return false;
+			FLAC__ASSERT(metadata->data.stream_info.max_framesize < (1u << FLAC__STREAM_METADATA_STREAMINFO_MAX_FRAME_SIZE_LEN));
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.max_framesize, FLAC__STREAM_METADATA_STREAMINFO_MAX_FRAME_SIZE_LEN))
+				return false;
+			FLAC__ASSERT(FLAC__format_sample_rate_is_valid(metadata->data.stream_info.sample_rate));
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.sample_rate, FLAC__STREAM_METADATA_STREAMINFO_SAMPLE_RATE_LEN))
+				return false;
+			FLAC__ASSERT(metadata->data.stream_info.channels > 0);
+			FLAC__ASSERT(metadata->data.stream_info.channels <= (1u << FLAC__STREAM_METADATA_STREAMINFO_CHANNELS_LEN));
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.channels-1, FLAC__STREAM_METADATA_STREAMINFO_CHANNELS_LEN))
+				return false;
+			FLAC__ASSERT(metadata->data.stream_info.bits_per_sample > 0);
+			FLAC__ASSERT(metadata->data.stream_info.bits_per_sample <= (1u << FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN));
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.bits_per_sample-1, FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN))
+				return false;
+			if(!FLAC__bitwriter_write_raw_uint64(bw, metadata->data.stream_info.total_samples, FLAC__STREAM_METADATA_STREAMINFO_TOTAL_SAMPLES_LEN))
+				return false;
+			if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.stream_info.md5sum, 16))
+				return false;
+			break;
+		case FLAC__METADATA_TYPE_PADDING:
+			if(!FLAC__bitwriter_write_zeroes(bw, metadata->length * 8))
+				return false;
+			break;
+		case FLAC__METADATA_TYPE_APPLICATION:
+			if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.application.id, FLAC__STREAM_METADATA_APPLICATION_ID_LEN / 8))
+				return false;
+			if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.application.data, metadata->length - (FLAC__STREAM_METADATA_APPLICATION_ID_LEN / 8)))
+				return false;
+			break;
+		case FLAC__METADATA_TYPE_SEEKTABLE:
+			for(i = 0; i < metadata->data.seek_table.num_points; i++) {
+				if(!FLAC__bitwriter_write_raw_uint64(bw, metadata->data.seek_table.points[i].sample_number, FLAC__STREAM_METADATA_SEEKPOINT_SAMPLE_NUMBER_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint64(bw, metadata->data.seek_table.points[i].stream_offset, FLAC__STREAM_METADATA_SEEKPOINT_STREAM_OFFSET_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.seek_table.points[i].frame_samples, FLAC__STREAM_METADATA_SEEKPOINT_FRAME_SAMPLES_LEN))
+					return false;
+			}
+			break;
+		case FLAC__METADATA_TYPE_VORBIS_COMMENT:
+			if(!FLAC__bitwriter_write_raw_uint32_little_endian(bw, vendor_string_length))
+				return false;
+			if(!FLAC__bitwriter_write_byte_block(bw, (const FLAC__byte*)FLAC__VENDOR_STRING, vendor_string_length))
+				return false;
+			if(!FLAC__bitwriter_write_raw_uint32_little_endian(bw, metadata->data.vorbis_comment.num_comments))
+				return false;
+			for(i = 0; i < metadata->data.vorbis_comment.num_comments; i++) {
+				if(!FLAC__bitwriter_write_raw_uint32_little_endian(bw, metadata->data.vorbis_comment.comments[i].length))
+					return false;
+				if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.vorbis_comment.comments[i].entry, metadata->data.vorbis_comment.comments[i].length))
+					return false;
+			}
+			break;
+		case FLAC__METADATA_TYPE_CUESHEET:
+			FLAC__ASSERT(FLAC__STREAM_METADATA_CUESHEET_MEDIA_CATALOG_NUMBER_LEN % 8 == 0);
+			if(!FLAC__bitwriter_write_byte_block(bw, (const FLAC__byte*)metadata->data.cue_sheet.media_catalog_number, FLAC__STREAM_METADATA_CUESHEET_MEDIA_CATALOG_NUMBER_LEN/8))
+				return false;
+			if(!FLAC__bitwriter_write_raw_uint64(bw, metadata->data.cue_sheet.lead_in, FLAC__STREAM_METADATA_CUESHEET_LEAD_IN_LEN))
+				return false;
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.cue_sheet.is_cd? 1 : 0, FLAC__STREAM_METADATA_CUESHEET_IS_CD_LEN))
+				return false;
+			if(!FLAC__bitwriter_write_zeroes(bw, FLAC__STREAM_METADATA_CUESHEET_RESERVED_LEN))
+				return false;
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.cue_sheet.num_tracks, FLAC__STREAM_METADATA_CUESHEET_NUM_TRACKS_LEN))
+				return false;
+			for(i = 0; i < metadata->data.cue_sheet.num_tracks; i++) {
+				const FLAC__StreamMetadata_CueSheet_Track *track = metadata->data.cue_sheet.tracks + i;
+
+				if(!FLAC__bitwriter_write_raw_uint64(bw, track->offset, FLAC__STREAM_METADATA_CUESHEET_TRACK_OFFSET_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, track->number, FLAC__STREAM_METADATA_CUESHEET_TRACK_NUMBER_LEN))
+					return false;
+				FLAC__ASSERT(FLAC__STREAM_METADATA_CUESHEET_TRACK_ISRC_LEN % 8 == 0);
+				if(!FLAC__bitwriter_write_byte_block(bw, (const FLAC__byte*)track->isrc, FLAC__STREAM_METADATA_CUESHEET_TRACK_ISRC_LEN/8))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, track->type, FLAC__STREAM_METADATA_CUESHEET_TRACK_TYPE_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, track->pre_emphasis, FLAC__STREAM_METADATA_CUESHEET_TRACK_PRE_EMPHASIS_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_zeroes(bw, FLAC__STREAM_METADATA_CUESHEET_TRACK_RESERVED_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, track->num_indices, FLAC__STREAM_METADATA_CUESHEET_TRACK_NUM_INDICES_LEN))
+					return false;
+				for(j = 0; j < track->num_indices; j++) {
+					const FLAC__StreamMetadata_CueSheet_Index *index = track->indices + j;
+
+					if(!FLAC__bitwriter_write_raw_uint64(bw, index->offset, FLAC__STREAM_METADATA_CUESHEET_INDEX_OFFSET_LEN))
+						return false;
+					if(!FLAC__bitwriter_write_raw_uint32(bw, index->number, FLAC__STREAM_METADATA_CUESHEET_INDEX_NUMBER_LEN))
+						return false;
+					if(!FLAC__bitwriter_write_zeroes(bw, FLAC__STREAM_METADATA_CUESHEET_INDEX_RESERVED_LEN))
+						return false;
+				}
+			}
+			break;
+		case FLAC__METADATA_TYPE_PICTURE:
+			{
+				size_t len;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.type, FLAC__STREAM_METADATA_PICTURE_TYPE_LEN))
+					return false;
+				len = strlen(metadata->data.picture.mime_type);
+				if(!FLAC__bitwriter_write_raw_uint32(bw, len, FLAC__STREAM_METADATA_PICTURE_MIME_TYPE_LENGTH_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_byte_block(bw, (const FLAC__byte*)metadata->data.picture.mime_type, len))
+					return false;
+				len = strlen((const char *)metadata->data.picture.description);
+				if(!FLAC__bitwriter_write_raw_uint32(bw, len, FLAC__STREAM_METADATA_PICTURE_DESCRIPTION_LENGTH_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.picture.description, len))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.width, FLAC__STREAM_METADATA_PICTURE_WIDTH_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.height, FLAC__STREAM_METADATA_PICTURE_HEIGHT_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.depth, FLAC__STREAM_METADATA_PICTURE_DEPTH_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.colors, FLAC__STREAM_METADATA_PICTURE_COLORS_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.data_length, FLAC__STREAM_METADATA_PICTURE_DATA_LENGTH_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.picture.data, metadata->data.picture.data_length))
+					return false;
+			}
+			break;
+		default:
+			if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.unknown.data, metadata->length))
+				return false;
+			break;
+	}
+
+	FLAC__ASSERT(FLAC__bitwriter_is_byte_aligned(bw));
+	return true;
+}
+
+FLAC__bool FLAC__frame_add_header(const FLAC__FrameHeader *header, FLAC__BitWriter *bw)
+{
+	unsigned u, blocksize_hint, sample_rate_hint;
+	FLAC__byte crc;
+
+	FLAC__ASSERT(FLAC__bitwriter_is_byte_aligned(bw));
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, FLAC__FRAME_HEADER_SYNC, FLAC__FRAME_HEADER_SYNC_LEN))
+		return false;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, 0, FLAC__FRAME_HEADER_RESERVED_LEN))
+		return false;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, (header->number_type == FLAC__FRAME_NUMBER_TYPE_FRAME_NUMBER)? 0 : 1, FLAC__FRAME_HEADER_BLOCKING_STRATEGY_LEN))
+		return false;
+
+	FLAC__ASSERT(header->blocksize > 0 && header->blocksize <= FLAC__MAX_BLOCK_SIZE);
+	/* when this assertion holds true, any legal blocksize can be expressed in the frame header */
+	FLAC__ASSERT(FLAC__MAX_BLOCK_SIZE <= 65535u);
+	blocksize_hint = 0;
+	switch(header->blocksize) {
+		case   192: u = 1; break;
+		case   576: u = 2; break;
+		case  1152: u = 3; break;
+		case  2304: u = 4; break;
+		case  4608: u = 5; break;
+		case   256: u = 8; break;
+		case   512: u = 9; break;
+		case  1024: u = 10; break;
+		case  2048: u = 11; break;
+		case  4096: u = 12; break;
+		case  8192: u = 13; break;
+		case 16384: u = 14; break;
+		case 32768: u = 15; break;
+		default:
+			if(header->blocksize <= 0x100)
+				blocksize_hint = u = 6;
+			else
+				blocksize_hint = u = 7;
+			break;
+	}
+	if(!FLAC__bitwriter_write_raw_uint32(bw, u, FLAC__FRAME_HEADER_BLOCK_SIZE_LEN))
+		return false;
+
+	FLAC__ASSERT(FLAC__format_sample_rate_is_valid(header->sample_rate));
+	sample_rate_hint = 0;
+	switch(header->sample_rate) {
+		case  88200: u = 1; break;
+		case 176400: u = 2; break;
+		case 192000: u = 3; break;
+		case   8000: u = 4; break;
+		case  16000: u = 5; break;
+		case  22050: u = 6; break;
+		case  24000: u = 7; break;
+		case  32000: u = 8; break;
+		case  44100: u = 9; break;
+		case  48000: u = 10; break;
+		case  96000: u = 11; break;
+		default:
+			if(header->sample_rate <= 255000 && header->sample_rate % 1000 == 0)
+				sample_rate_hint = u = 12;
+			else if(header->sample_rate % 10 == 0)
+				sample_rate_hint = u = 14;
+			else if(header->sample_rate <= 0xffff)
+				sample_rate_hint = u = 13;
+			else
+				u = 0;
+			break;
+	}
+	if(!FLAC__bitwriter_write_raw_uint32(bw, u, FLAC__FRAME_HEADER_SAMPLE_RATE_LEN))
+		return false;
+
+	FLAC__ASSERT(header->channels > 0 && header->channels <= (1u << FLAC__STREAM_METADATA_STREAMINFO_CHANNELS_LEN) && header->channels <= FLAC__MAX_CHANNELS);
+	switch(header->channel_assignment) {
+		case FLAC__CHANNEL_ASSIGNMENT_INDEPENDENT:
+			u = header->channels - 1;
+			break;
+		case FLAC__CHANNEL_ASSIGNMENT_LEFT_SIDE:
+			FLAC__ASSERT(header->channels == 2);
+			u = 8;
+			break;
+		case FLAC__CHANNEL_ASSIGNMENT_RIGHT_SIDE:
+			FLAC__ASSERT(header->channels == 2);
+			u = 9;
+			break;
+		case FLAC__CHANNEL_ASSIGNMENT_MID_SIDE:
+			FLAC__ASSERT(header->channels == 2);
+			u = 10;
+			break;
+		default:
+			FLAC__ASSERT(0);
+	}
+	if(!FLAC__bitwriter_write_raw_uint32(bw, u, FLAC__FRAME_HEADER_CHANNEL_ASSIGNMENT_LEN))
+		return false;
+
+	FLAC__ASSERT(header->bits_per_sample > 0 && header->bits_per_sample <= (1u << FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN));
+	switch(header->bits_per_sample) {
+		case 8 : u = 1; break;
+		case 12: u = 2; break;
+		case 16: u = 4; break;
+		case 20: u = 5; break;
+		case 24: u = 6; break;
+		default: u = 0; break;
+	}
+	if(!FLAC__bitwriter_write_raw_uint32(bw, u, FLAC__FRAME_HEADER_BITS_PER_SAMPLE_LEN))
+		return false;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, 0, FLAC__FRAME_HEADER_ZERO_PAD_LEN))
+		return false;
+
+	if(header->number_type == FLAC__FRAME_NUMBER_TYPE_FRAME_NUMBER) {
+		if(!FLAC__bitwriter_write_utf8_uint32(bw, header->number.frame_number))
+			return false;
+	}
+	else {
+		if(!FLAC__bitwriter_write_utf8_uint64(bw, header->number.sample_number))
+			return false;
+	}
+
+	if(blocksize_hint)
+		if(!FLAC__bitwriter_write_raw_uint32(bw, header->blocksize-1, (blocksize_hint==6)? 8:16))
+			return false;
+
+	switch(sample_rate_hint) {
+		case 12:
+			if(!FLAC__bitwriter_write_raw_uint32(bw, header->sample_rate / 1000, 8))
+				return false;
+			break;
+		case 13:
+			if(!FLAC__bitwriter_write_raw_uint32(bw, header->sample_rate, 16))
+				return false;
+			break;
+		case 14:
+			if(!FLAC__bitwriter_write_raw_uint32(bw, header->sample_rate / 10, 16))
+				return false;
+			break;
+	}
+
+	/* write the CRC */
+	if(!FLAC__bitwriter_get_write_crc8(bw, &crc))
+		return false;
+	if(!FLAC__bitwriter_write_raw_uint32(bw, crc, FLAC__FRAME_HEADER_CRC_LEN))
+		return false;
+
+	return true;
+}
+
+FLAC__bool FLAC__subframe_add_constant(const FLAC__Subframe_Constant *subframe, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw)
+{
+	FLAC__bool ok;
+
+	ok =
+		FLAC__bitwriter_write_raw_uint32(bw, FLAC__SUBFRAME_TYPE_CONSTANT_BYTE_ALIGNED_MASK | (wasted_bits? 1:0), FLAC__SUBFRAME_ZERO_PAD_LEN + FLAC__SUBFRAME_TYPE_LEN + FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN) &&
+		(wasted_bits? FLAC__bitwriter_write_unary_unsigned(bw, wasted_bits-1) : true) &&
+		FLAC__bitwriter_write_raw_int32(bw, subframe->value, subframe_bps)
+	;
+
+	return ok;
+}
+
+FLAC__bool FLAC__subframe_add_fixed(const FLAC__Subframe_Fixed *subframe, unsigned residual_samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw)
+{
+	unsigned i;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, FLAC__SUBFRAME_TYPE_FIXED_BYTE_ALIGNED_MASK | (subframe->order<<1) | (wasted_bits? 1:0), FLAC__SUBFRAME_ZERO_PAD_LEN + FLAC__SUBFRAME_TYPE_LEN + FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN))
+		return false;
+	if(wasted_bits)
+		if(!FLAC__bitwriter_write_unary_unsigned(bw, wasted_bits-1))
+			return false;
+
+	for(i = 0; i < subframe->order; i++)
+		if(!FLAC__bitwriter_write_raw_int32(bw, subframe->warmup[i], subframe_bps))
+			return false;
+
+	if(!add_entropy_coding_method_(bw, &subframe->entropy_coding_method))
+		return false;
+	switch(subframe->entropy_coding_method.type) {
+		case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE:
+		case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2:
+			if(!add_residual_partitioned_rice_(
+				bw,
+				subframe->residual,
+				residual_samples,
+				subframe->order,
+				subframe->entropy_coding_method.data.partitioned_rice.contents->parameters,
+				subframe->entropy_coding_method.data.partitioned_rice.contents->raw_bits,
+				subframe->entropy_coding_method.data.partitioned_rice.order,
+				/*is_extended=*/subframe->entropy_coding_method.type == FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2
+			))
+				return false;
+			break;
+		default:
+			FLAC__ASSERT(0);
+	}
+
+	return true;
+}
+
+FLAC__bool FLAC__subframe_add_lpc(const FLAC__Subframe_LPC *subframe, unsigned residual_samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw)
+{
+	unsigned i;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, FLAC__SUBFRAME_TYPE_LPC_BYTE_ALIGNED_MASK | ((subframe->order-1)<<1) | (wasted_bits? 1:0), FLAC__SUBFRAME_ZERO_PAD_LEN + FLAC__SUBFRAME_TYPE_LEN + FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN))
+		return false;
+	if(wasted_bits)
+		if(!FLAC__bitwriter_write_unary_unsigned(bw, wasted_bits-1))
+			return false;
+
+	for(i = 0; i < subframe->order; i++)
+		if(!FLAC__bitwriter_write_raw_int32(bw, subframe->warmup[i], subframe_bps))
+			return false;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, subframe->qlp_coeff_precision-1, FLAC__SUBFRAME_LPC_QLP_COEFF_PRECISION_LEN))
+		return false;
+	if(!FLAC__bitwriter_write_raw_int32(bw, subframe->quantization_level, FLAC__SUBFRAME_LPC_QLP_SHIFT_LEN))
+		return false;
+	for(i = 0; i < subframe->order; i++)
+		if(!FLAC__bitwriter_write_raw_int32(bw, subframe->qlp_coeff[i], subframe->qlp_coeff_precision))
+			return false;
+
+	if(!add_entropy_coding_method_(bw, &subframe->entropy_coding_method))
+		return false;
+	switch(subframe->entropy_coding_method.type) {
+		case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE:
+		case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2:
+			if(!add_residual_partitioned_rice_(
+				bw,
+				subframe->residual,
+				residual_samples,
+				subframe->order,
+				subframe->entropy_coding_method.data.partitioned_rice.contents->parameters,
+				subframe->entropy_coding_method.data.partitioned_rice.contents->raw_bits,
+				subframe->entropy_coding_method.data.partitioned_rice.order,
+				/*is_extended=*/subframe->entropy_coding_method.type == FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2
+			))
+				return false;
+			break;
+		default:
+			FLAC__ASSERT(0);
+	}
+
+	return true;
+}
+
+FLAC__bool FLAC__subframe_add_verbatim(const FLAC__Subframe_Verbatim *subframe, unsigned samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw)
+{
+	unsigned i;
+	const FLAC__int32 *signal = subframe->data;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, FLAC__SUBFRAME_TYPE_VERBATIM_BYTE_ALIGNED_MASK | (wasted_bits? 1:0), FLAC__SUBFRAME_ZERO_PAD_LEN + FLAC__SUBFRAME_TYPE_LEN + FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN))
+		return false;
+	if(wasted_bits)
+		if(!FLAC__bitwriter_write_unary_unsigned(bw, wasted_bits-1))
+			return false;
+
+	for(i = 0; i < samples; i++)
+		if(!FLAC__bitwriter_write_raw_int32(bw, signal[i], subframe_bps))
+			return false;
+
+	return true;
+}
+
+FLAC__bool add_entropy_coding_method_(FLAC__BitWriter *bw, const FLAC__EntropyCodingMethod *method)
+{
+	if(!FLAC__bitwriter_write_raw_uint32(bw, method->type, FLAC__ENTROPY_CODING_METHOD_TYPE_LEN))
+		return false;
+	switch(method->type) {
+		case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE:
+		case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2:
+			if(!FLAC__bitwriter_write_raw_uint32(bw, method->data.partitioned_rice.order, FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_ORDER_LEN))
+				return false;
+			break;
+		default:
+			FLAC__ASSERT(0);
+	}
+	return true;
+}
+
+FLAC__bool add_residual_partitioned_rice_(FLAC__BitWriter *bw, const FLAC__int32 residual[], const unsigned residual_samples, const unsigned predictor_order, const unsigned rice_parameters[], const unsigned raw_bits[], const unsigned partition_order, const FLAC__bool is_extended)
+{
+	const unsigned plen = is_extended? FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_PARAMETER_LEN : FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_PARAMETER_LEN;
+	const unsigned pesc = is_extended? FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_ESCAPE_PARAMETER : FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_ESCAPE_PARAMETER;
+
+	if(partition_order == 0) {
+		unsigned i;
+
+		if(raw_bits[0] == 0) {
+			if(!FLAC__bitwriter_write_raw_uint32(bw, rice_parameters[0], plen))
+				return false;
+			if(!FLAC__bitwriter_write_rice_signed_block(bw, residual, residual_samples, rice_parameters[0]))
+				return false;
+		}
+		else {
+			FLAC__ASSERT(rice_parameters[0] == 0);
+			if(!FLAC__bitwriter_write_raw_uint32(bw, pesc, plen))
+				return false;
+			if(!FLAC__bitwriter_write_raw_uint32(bw, raw_bits[0], FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_RAW_LEN))
+				return false;
+			for(i = 0; i < residual_samples; i++) {
+				if(!FLAC__bitwriter_write_raw_int32(bw, residual[i], raw_bits[0]))
+					return false;
+			}
+		}
+		return true;
+	}
+	else {
+		unsigned i, j, k = 0, k_last = 0;
+		unsigned partition_samples;
+		const unsigned default_partition_samples = (residual_samples+predictor_order) >> partition_order;
+		for(i = 0; i < (1u<<partition_order); i++) {
+			partition_samples = default_partition_samples;
+			if(i == 0)
+				partition_samples -= predictor_order;
+			k += partition_samples;
+			if(raw_bits[i] == 0) {
+				if(!FLAC__bitwriter_write_raw_uint32(bw, rice_parameters[i], plen))
+					return false;
+				if(!FLAC__bitwriter_write_rice_signed_block(bw, residual+k_last, k-k_last, rice_parameters[i]))
+					return false;
+			}
+			else {
+				if(!FLAC__bitwriter_write_raw_uint32(bw, pesc, plen))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, raw_bits[i], FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_RAW_LEN))
+					return false;
+				for(j = k_last; j < k; j++) {
+					if(!FLAC__bitwriter_write_raw_int32(bw, residual[j], raw_bits[i]))
+						return false;
+				}
+			}
+			k_last = k;
+		}
+		return true;
+	}
+}

source/modules/juce_audio_formats/codecs/flac/libFLAC/window_flac.c

@@ -0,0 +1,225 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include <math.h>
+#include "../assert.h"
+#include "../format.h"
+#include "include/private/window.h"
+
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+
+#ifndef M_PI
+/* math.h in VC++ doesn't seem to have this (how Microsoft is that?) */
+#define M_PI 3.14159265358979323846
+#endif
+
+
+void FLAC__window_bartlett(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	if (L & 1) {
+		for (n = 0; n <= N/2; n++)
+			window[n] = 2.0f * n / (float)N;
+		for (; n <= N; n++)
+			window[n] = 2.0f - 2.0f * n / (float)N;
+	}
+	else {
+		for (n = 0; n <= L/2-1; n++)
+			window[n] = 2.0f * n / (float)N;
+		for (; n <= N; n++)
+			window[n] = 2.0f - 2.0f * (N-n) / (float)N;
+	}
+}
+
+void FLAC__window_bartlett_hann(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = (FLAC__real)(0.62f - 0.48f * fabs((float)n/(float)N+0.5f) + 0.38f * cos(2.0f * M_PI * ((float)n/(float)N+0.5f)));
+}
+
+void FLAC__window_blackman(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = (FLAC__real)(0.42f - 0.5f * cos(2.0f * M_PI * n / N) + 0.08f * cos(4.0f * M_PI * n / N));
+}
+
+/* 4-term -92dB side-lobe */
+void FLAC__window_blackman_harris_4term_92db_sidelobe(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n <= N; n++)
+		window[n] = (FLAC__real)(0.35875f - 0.48829f * cos(2.0f * M_PI * n / N) + 0.14128f * cos(4.0f * M_PI * n / N) - 0.01168f * cos(6.0f * M_PI * n / N));
+}
+
+void FLAC__window_connes(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	const double N2 = (double)N / 2.;
+	FLAC__int32 n;
+
+	for (n = 0; n <= N; n++) {
+		double k = ((double)n - N2) / N2;
+		k = 1.0f - k * k;
+		window[n] = (FLAC__real)(k * k);
+	}
+}
+
+void FLAC__window_flattop(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = (FLAC__real)(1.0f - 1.93f * cos(2.0f * M_PI * n / N) + 1.29f * cos(4.0f * M_PI * n / N) - 0.388f * cos(6.0f * M_PI * n / N) + 0.0322f * cos(8.0f * M_PI * n / N));
+}
+
+void FLAC__window_gauss(FLAC__real *window, const FLAC__int32 L, const FLAC__real stddev)
+{
+	const FLAC__int32 N = L - 1;
+	const double N2 = (double)N / 2.;
+	FLAC__int32 n;
+
+	for (n = 0; n <= N; n++) {
+		const double k = ((double)n - N2) / (stddev * N2);
+		window[n] = (FLAC__real)exp(-0.5f * k * k);
+	}
+}
+
+void FLAC__window_hamming(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = (FLAC__real)(0.54f - 0.46f * cos(2.0f * M_PI * n / N));
+}
+
+void FLAC__window_hann(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = (FLAC__real)(0.5f - 0.5f * cos(2.0f * M_PI * n / N));
+}
+
+void FLAC__window_kaiser_bessel(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = (FLAC__real)(0.402f - 0.498f * cos(2.0f * M_PI * n / N) + 0.098f * cos(4.0f * M_PI * n / N) - 0.001f * cos(6.0f * M_PI * n / N));
+}
+
+void FLAC__window_nuttall(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = (FLAC__real)(0.3635819f - 0.4891775f*cos(2.0f*M_PI*n/N) + 0.1365995f*cos(4.0f*M_PI*n/N) - 0.0106411f*cos(6.0f*M_PI*n/N));
+}
+
+void FLAC__window_rectangle(FLAC__real *window, const FLAC__int32 L)
+{
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = 1.0f;
+}
+
+void FLAC__window_triangle(FLAC__real *window, const FLAC__int32 L)
+{
+	FLAC__int32 n;
+
+	if (L & 1) {
+		for (n = 1; n <= L+1/2; n++)
+			window[n-1] = 2.0f * n / ((float)L + 1.0f);
+		for (; n <= L; n++)
+			window[n-1] = - (float)(2 * (L - n + 1)) / ((float)L + 1.0f);
+	}
+	else {
+		for (n = 1; n <= L/2; n++)
+			window[n-1] = 2.0f * n / (float)L;
+		for (; n <= L; n++)
+			window[n-1] = ((float)(2 * (L - n)) + 1.0f) / (float)L;
+	}
+}
+
+void FLAC__window_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p)
+{
+	if (p <= 0.0)
+		FLAC__window_rectangle(window, L);
+	else if (p >= 1.0)
+		FLAC__window_hann(window, L);
+	else {
+		const FLAC__int32 Np = (FLAC__int32)(p / 2.0f * L) - 1;
+		FLAC__int32 n;
+		/* start with rectangle... */
+		FLAC__window_rectangle(window, L);
+		/* ...replace ends with hann */
+		if (Np > 0) {
+			for (n = 0; n <= Np; n++) {
+				window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * n / Np));
+				window[L-Np-1+n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * (n+Np) / Np));
+			}
+		}
+	}
+}
+
+void FLAC__window_welch(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	const double N2 = (double)N / 2.;
+	FLAC__int32 n;
+
+	for (n = 0; n <= N; n++) {
+		const double k = ((double)n - N2) / N2;
+		window[n] = (FLAC__real)(1.0f - k * k);
+	}
+}
+
+#endif /* !defined FLAC__INTEGER_ONLY_LIBRARY */

source/modules/juce_audio_formats/codecs/flac/ordinals.h

@@ -0,0 +1,80 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#ifndef FLAC__ORDINALS_H
+#define FLAC__ORDINALS_H
+
+#if !(defined(_MSC_VER) || defined(__BORLANDC__) || defined(__EMX__))
+#include <inttypes.h>
+#endif
+
+typedef signed char FLAC__int8;
+typedef unsigned char FLAC__uint8;
+
+#if defined(_MSC_VER) || defined(__BORLANDC__)
+typedef __int16 FLAC__int16;
+typedef __int32 FLAC__int32;
+typedef __int64 FLAC__int64;
+typedef unsigned __int16 FLAC__uint16;
+typedef unsigned __int32 FLAC__uint32;
+typedef unsigned __int64 FLAC__uint64;
+#elif defined(__EMX__)
+typedef short FLAC__int16;
+typedef long FLAC__int32;
+typedef long long FLAC__int64;
+typedef unsigned short FLAC__uint16;
+typedef unsigned long FLAC__uint32;
+typedef unsigned long long FLAC__uint64;
+#else
+typedef int16_t FLAC__int16;
+typedef int32_t FLAC__int32;
+typedef int64_t FLAC__int64;
+typedef uint16_t FLAC__uint16;
+typedef uint32_t FLAC__uint32;
+typedef uint64_t FLAC__uint64;
+#endif
+
+typedef int FLAC__bool;
+
+typedef FLAC__uint8 FLAC__byte;
+
+#ifdef true
+#undef true
+#endif
+#ifdef false
+#undef false
+#endif
+#ifndef __cplusplus
+#define true 1
+#define false 0
+#endif
+
+#endif

source/modules/juce_audio_formats/codecs/juce_AiffAudioFormat.cpp

@@ -0,0 +1,957 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+//==============================================================================
+static const char* const aiffFormatName = "AIFF file";
+static const char* const aiffExtensions[] = { ".aiff", ".aif", 0 };
+
+//==============================================================================
+const char* const AiffAudioFormat::appleOneShot         = "apple one shot";
+const char* const AiffAudioFormat::appleRootSet         = "apple root set";
+const char* const AiffAudioFormat::appleRootNote        = "apple root note";
+const char* const AiffAudioFormat::appleBeats           = "apple beats";
+const char* const AiffAudioFormat::appleDenominator     = "apple denominator";
+const char* const AiffAudioFormat::appleNumerator       = "apple numerator";
+const char* const AiffAudioFormat::appleTag             = "apple tag";
+const char* const AiffAudioFormat::appleKey             = "apple key";
+
+//==============================================================================
+namespace AiffFileHelpers
+{
+    inline int chunkName (const char* const name)   { return (int) ByteOrder::littleEndianInt (name); }
+
+   #if JUCE_MSVC
+    #pragma pack (push, 1)
+   #endif
+
+    //==============================================================================
+    struct InstChunk
+    {
+        struct Loop
+        {
+            uint16 type; // these are different in AIFF and WAV
+            uint16 startIdentifier;
+            uint16 endIdentifier;
+        } JUCE_PACKED;
+
+        int8 baseNote;
+        int8 detune;
+        int8 lowNote;
+        int8 highNote;
+        int8 lowVelocity;
+        int8 highVelocity;
+        int16 gain;
+        Loop sustainLoop;
+        Loop releaseLoop;
+
+        void copyTo (StringPairArray& values) const
+        {
+            values.set ("MidiUnityNote",        String (baseNote));
+            values.set ("Detune",               String (detune));
+
+            values.set ("LowNote",              String (lowNote));
+            values.set ("HighNote",             String (highNote));
+            values.set ("LowVelocity",          String (lowVelocity));
+            values.set ("HighVelocity",         String (highVelocity));
+
+            values.set ("Gain",                 String ((int16) ByteOrder::swapIfLittleEndian ((uint16) gain)));
+
+            values.set ("NumSampleLoops",       String (2));        // always 2 with AIFF, WAV can have more
+            values.set ("Loop0Type",            String (ByteOrder::swapIfLittleEndian (sustainLoop.type)));
+            values.set ("Loop0StartIdentifier", String (ByteOrder::swapIfLittleEndian (sustainLoop.startIdentifier)));
+            values.set ("Loop0EndIdentifier",   String (ByteOrder::swapIfLittleEndian (sustainLoop.endIdentifier)));
+            values.set ("Loop1Type",            String (ByteOrder::swapIfLittleEndian (releaseLoop.type)));
+            values.set ("Loop1StartIdentifier", String (ByteOrder::swapIfLittleEndian (releaseLoop.startIdentifier)));
+            values.set ("Loop1EndIdentifier",   String (ByteOrder::swapIfLittleEndian (releaseLoop.endIdentifier)));
+        }
+
+        static uint16 getValue16 (const StringPairArray& values, const char* name, const char* def)
+        {
+            return ByteOrder::swapIfLittleEndian ((uint16) values.getValue (name, def).getIntValue());
+        }
+
+        static int8 getValue8 (const StringPairArray& values, const char* name, const char* def)
+        {
+            return (int8) values.getValue (name, def).getIntValue();
+        }
+
+        static void create (MemoryBlock& block, const StringPairArray& values)
+        {
+            if (values.getAllKeys().contains ("MidiUnityNote", true))
+            {
+                block.setSize ((sizeof (InstChunk) + 3) & ~(size_t) 3, true);
+                InstChunk& inst = *static_cast <InstChunk*> (block.getData());
+
+                inst.baseNote      = getValue8 (values, "MidiUnityNote", "60");
+                inst.detune        = getValue8 (values, "Detune", "0");
+                inst.lowNote       = getValue8 (values, "LowNote", "0");
+                inst.highNote      = getValue8 (values, "HighNote", "127");
+                inst.lowVelocity   = getValue8 (values, "LowVelocity", "1");
+                inst.highVelocity  = getValue8 (values, "HighVelocity", "127");
+                inst.gain          = (int16) getValue16 (values, "Gain", "0");
+
+                inst.sustainLoop.type              = getValue16 (values, "Loop0Type", "0");
+                inst.sustainLoop.startIdentifier   = getValue16 (values, "Loop0StartIdentifier", "0");
+                inst.sustainLoop.endIdentifier     = getValue16 (values, "Loop0EndIdentifier", "0");
+                inst.releaseLoop.type              = getValue16 (values, "Loop1Type", "0");
+                inst.releaseLoop.startIdentifier   = getValue16 (values, "Loop1StartIdentifier", "0");
+                inst.releaseLoop.endIdentifier     = getValue16 (values, "Loop1EndIdentifier", "0");
+            }
+        }
+
+    } JUCE_PACKED;
+
+    //==============================================================================
+    struct BASCChunk
+    {
+        enum Key
+        {
+            minor = 1,
+            major = 2,
+            neither = 3,
+            both = 4
+        };
+
+        BASCChunk (InputStream& input)
+        {
+            zerostruct (*this);
+
+            flags       = (uint32) input.readIntBigEndian();
+            numBeats    = (uint32) input.readIntBigEndian();
+            rootNote    = (uint16) input.readShortBigEndian();
+            key         = (uint16) input.readShortBigEndian();
+            timeSigNum  = (uint16) input.readShortBigEndian();
+            timeSigDen  = (uint16) input.readShortBigEndian();
+            oneShot     = (uint16) input.readShortBigEndian();
+            input.read (unknown, sizeof (unknown));
+        }
+
+        void addToMetadata (StringPairArray& metadata) const
+        {
+            const bool rootNoteSet = rootNote != 0;
+
+            setBoolFlag (metadata, AiffAudioFormat::appleOneShot, oneShot == 2);
+            setBoolFlag (metadata, AiffAudioFormat::appleRootSet, rootNoteSet);
+
+            if (rootNoteSet)
+                metadata.set (AiffAudioFormat::appleRootNote,   String (rootNote));
+
+            metadata.set (AiffAudioFormat::appleBeats,          String (numBeats));
+            metadata.set (AiffAudioFormat::appleDenominator,    String (timeSigDen));
+            metadata.set (AiffAudioFormat::appleNumerator,      String (timeSigNum));
+
+            const char* keyString = nullptr;
+
+            switch (key)
+            {
+                case minor:     keyString = "major";        break;
+                case major:     keyString = "major";        break;
+                case neither:   keyString = "neither";      break;
+                case both:      keyString = "both";         break;
+            }
+
+            if (keyString != nullptr)
+                metadata.set (AiffAudioFormat::appleKey, keyString);
+        }
+
+        void setBoolFlag (StringPairArray& values, const char* name, bool shouldBeSet) const
+        {
+            values.set (name, shouldBeSet ? "1" : "0");
+        }
+
+        uint32 flags;
+        uint32 numBeats;
+        uint16 rootNote;
+        uint16 key;
+        uint16 timeSigNum;
+        uint16 timeSigDen;
+        uint16 oneShot;
+        uint8 unknown[66];
+    } JUCE_PACKED;
+
+   #if JUCE_MSVC
+    #pragma pack (pop)
+   #endif
+
+    //==============================================================================
+    static String readCATEChunk (InputStream& input, const uint32 length)
+    {
+        MemoryBlock mb;
+        input.skipNextBytes (4);
+        input.readIntoMemoryBlock (mb, (ssize_t) length - 4);
+
+        static const char* appleGenres[] =
+        {
+            "Rock/Blues",
+            "Electronic/Dance",
+            "Jazz",
+            "Urban",
+            "World/Ethnic",
+            "Cinematic/New Age",
+            "Orchestral",
+            "Country/Folk",
+            "Experimental",
+            "Other Genre",
+            nullptr
+        };
+
+        const StringArray genres (appleGenres);
+        StringArray tagsArray;
+
+        int bytesLeft = (int) mb.getSize();
+        const char* data = static_cast <const char*> (mb.getData());
+
+        while (bytesLeft > 0)
+        {
+            const String tag (CharPointer_UTF8 (data),
+                              CharPointer_UTF8 (data + bytesLeft));
+
+            if (tag.isNotEmpty())
+                tagsArray.add (data);
+
+            const int numBytesInTag = genres.contains (tag) ? 118 : 50;
+            data += numBytesInTag;
+            bytesLeft -= numBytesInTag;
+        }
+
+        return tagsArray.joinIntoString (";");
+    }
+
+    //==============================================================================
+    namespace MarkChunk
+    {
+        static bool metaDataContainsZeroIdentifiers (const StringPairArray& values)
+        {
+            // (zero cue identifiers are valid for WAV but not for AIFF)
+            const String cueString ("Cue");
+            const String noteString ("CueNote");
+            const String identifierString ("Identifier");
+
+            const StringArray& keys = values.getAllKeys();
+
+            for (int i = 0; i < keys.size(); ++i)
+            {
+                const String key (keys[i]);
+
+                if (key.startsWith (noteString))
+                    continue; // zero identifier IS valid in a COMT chunk
+
+                if (key.startsWith (cueString) && key.contains (identifierString))
+                {
+                    const int value = values.getValue (key, "-1").getIntValue();
+
+                    if (value == 0)
+                        return true;
+                }
+            }
+
+            return false;
+        }
+
+        static void create (MemoryBlock& block, const StringPairArray& values)
+        {
+            const int numCues = values.getValue ("NumCuePoints", "0").getIntValue();
+
+            if (numCues > 0)
+            {
+                MemoryOutputStream out (block, false);
+
+                out.writeShortBigEndian ((short) numCues);
+
+                const int numCueLabels = values.getValue ("NumCueLabels", "0").getIntValue();
+                const int idOffset = metaDataContainsZeroIdentifiers (values) ? 1 : 0; // can't have zero IDs in AIFF
+
+               #if JUCE_DEBUG
+                Array<int> identifiers;
+               #endif
+
+                for (int i = 0; i < numCues; ++i)
+                {
+                    const String prefixCue ("Cue" + String (i));
+                    const int identifier = idOffset + values.getValue (prefixCue + "Identifier", "1").getIntValue();
+
+                   #if JUCE_DEBUG
+                    jassert (! identifiers.contains (identifier));
+                    identifiers.add (identifier);
+                   #endif
+
+                    const int offset = values.getValue (prefixCue + "Offset", "0").getIntValue();
+                    String label ("CueLabel" + String (i));
+
+                    for (int labelIndex = 0; labelIndex < numCueLabels; ++labelIndex)
+                    {
+                        const String prefixLabel ("CueLabel" + String (labelIndex));
+                        const int labelIdentifier = idOffset + values.getValue (prefixLabel + "Identifier", "1").getIntValue();
+
+                        if (labelIdentifier == identifier)
+                        {
+                            label = values.getValue (prefixLabel + "Text", label);
+                            break;
+                        }
+                    }
+
+                    out.writeShortBigEndian ((short) identifier);
+                    out.writeIntBigEndian (offset);
+
+                    const size_t labelLength = jmin ((size_t) 254, label.getNumBytesAsUTF8()); // seems to need null terminator even though it's a pstring
+                    out.writeByte ((char) labelLength + 1);
+                    out.write (label.toUTF8(), labelLength);
+                    out.writeByte (0);
+                }
+
+                if ((out.getDataSize() & 1) != 0)
+                    out.writeByte (0);
+            }
+        }
+    }
+
+    //==============================================================================
+    namespace COMTChunk
+    {
+        static void create (MemoryBlock& block, const StringPairArray& values)
+        {
+            const int numNotes = values.getValue ("NumCueNotes", "0").getIntValue();
+
+            if (numNotes > 0)
+            {
+                MemoryOutputStream out (block, false);
+                out.writeShortBigEndian ((short) numNotes);
+
+                for (int i = 0; i < numNotes; ++i)
+                {
+                    const String prefix ("CueNote" + String (i));
+
+                    out.writeIntBigEndian (values.getValue (prefix + "TimeStamp", "0").getIntValue());
+                    out.writeShortBigEndian ((short) values.getValue (prefix + "Identifier", "0").getIntValue());
+
+                    const String comment (values.getValue (prefix + "Text", String::empty));
+
+                    const size_t commentLength = jmin (comment.getNumBytesAsUTF8(), (size_t) 65534);
+                    out.writeShortBigEndian ((short) commentLength + 1);
+                    out.write (comment.toUTF8(), commentLength);
+                    out.writeByte (0);
+
+                    if ((out.getDataSize() & 1) != 0)
+                        out.writeByte (0);
+                }
+            }
+        }
+    }
+}
+
+//==============================================================================
+class AiffAudioFormatReader  : public AudioFormatReader
+{
+public:
+    AiffAudioFormatReader (InputStream* in)
+        : AudioFormatReader (in, TRANS (aiffFormatName))
+    {
+        using namespace AiffFileHelpers;
+
+        if (input->readInt() == chunkName ("FORM"))
+        {
+            const int len = input->readIntBigEndian();
+            const int64 end = input->getPosition() + len;
+
+            const int nextType = input->readInt();
+            if (nextType == chunkName ("AIFF") || nextType == chunkName ("AIFC"))
+            {
+                bool hasGotVer = false;
+                bool hasGotData = false;
+                bool hasGotType = false;
+
+                while (input->getPosition() < end)
+                {
+                    const int type = input->readInt();
+                    const uint32 length = (uint32) input->readIntBigEndian();
+                    const int64 chunkEnd = input->getPosition() + length;
+
+                    if (type == chunkName ("FVER"))
+                    {
+                        hasGotVer = true;
+
+                        const int ver = input->readIntBigEndian();
+                        if (ver != 0 && ver != (int) 0xa2805140)
+                            break;
+                    }
+                    else if (type == chunkName ("COMM"))
+                    {
+                        hasGotType = true;
+
+                        numChannels = (unsigned int) input->readShortBigEndian();
+                        lengthInSamples = input->readIntBigEndian();
+                        bitsPerSample = (unsigned int) input->readShortBigEndian();
+                        bytesPerFrame = (int) ((numChannels * bitsPerSample) >> 3);
+
+                        unsigned char sampleRateBytes[10];
+                        input->read (sampleRateBytes, 10);
+                        const int byte0 = sampleRateBytes[0];
+
+                        if ((byte0 & 0x80) != 0
+                             || byte0 <= 0x3F || byte0 > 0x40
+                             || (byte0 == 0x40 && sampleRateBytes[1] > 0x1C))
+                            break;
+
+                        unsigned int sampRate = ByteOrder::bigEndianInt (sampleRateBytes + 2);
+                        sampRate >>= (16414 - ByteOrder::bigEndianShort (sampleRateBytes));
+                        sampleRate = (int) sampRate;
+
+                        if (length <= 18)
+                        {
+                            // some types don't have a chunk large enough to include a compression
+                            // type, so assume it's just big-endian pcm
+                            littleEndian = false;
+                        }
+                        else
+                        {
+                            const int compType = input->readInt();
+
+                            if (compType == chunkName ("NONE") || compType == chunkName ("twos"))
+                            {
+                                littleEndian = false;
+                            }
+                            else if (compType == chunkName ("sowt"))
+                            {
+                                littleEndian = true;
+                            }
+                            else if (compType == chunkName ("fl32") || compType == chunkName ("FL32"))
+                            {
+                                littleEndian = false;
+                                usesFloatingPointData = true;
+                            }
+                            else
+                            {
+                                sampleRate = 0;
+                                break;
+                            }
+                        }
+                    }
+                    else if (type == chunkName ("SSND"))
+                    {
+                        hasGotData = true;
+
+                        const int offset = input->readIntBigEndian();
+                        dataChunkStart = input->getPosition() + 4 + offset;
+                        lengthInSamples = (bytesPerFrame > 0) ? jmin (lengthInSamples, ((int64) length) / (int64) bytesPerFrame) : 0;
+                    }
+                    else if (type == chunkName ("MARK"))
+                    {
+                        const uint16 numCues = (uint16) input->readShortBigEndian();
+
+                        // these two are always the same for AIFF-read files
+                        metadataValues.set ("NumCuePoints", String (numCues));
+                        metadataValues.set ("NumCueLabels", String (numCues));
+
+                        for (uint16 i = 0; i < numCues; ++i)
+                        {
+                            uint16 identifier = (uint16) input->readShortBigEndian();
+                            uint32 offset = (uint32) input->readIntBigEndian();
+                            uint8 stringLength = (uint8) input->readByte();
+                            MemoryBlock textBlock;
+                            input->readIntoMemoryBlock (textBlock, stringLength);
+
+                            // if the stringLength is even then read one more byte as the
+                            // string needs to be an even number of bytes INCLUDING the
+                            // leading length character in the pascal string
+                            if ((stringLength & 1) == 0)
+                                input->readByte();
+
+                            const String prefixCue ("Cue" + String (i));
+                            metadataValues.set (prefixCue + "Identifier", String (identifier));
+                            metadataValues.set (prefixCue + "Offset", String (offset));
+
+                            const String prefixLabel ("CueLabel" + String (i));
+                            metadataValues.set (prefixLabel + "Identifier", String (identifier));
+                            metadataValues.set (prefixLabel + "Text", textBlock.toString());
+                        }
+                    }
+                    else if (type == chunkName ("COMT"))
+                    {
+                        const uint16 numNotes = (uint16) input->readShortBigEndian();
+                        metadataValues.set ("NumCueNotes", String (numNotes));
+
+                        for (uint16 i = 0; i < numNotes; ++i)
+                        {
+                            uint32 timestamp = (uint32) input->readIntBigEndian();
+                            uint16 identifier = (uint16) input->readShortBigEndian(); // may be zero in this case
+                            uint16 stringLength = (uint16) input->readShortBigEndian();
+
+                            MemoryBlock textBlock;
+                            input->readIntoMemoryBlock (textBlock, stringLength + (stringLength & 1));
+
+                            const String prefix ("CueNote" + String (i));
+                            metadataValues.set (prefix + "TimeStamp", String (timestamp));
+                            metadataValues.set (prefix + "Identifier", String (identifier));
+                            metadataValues.set (prefix + "Text", textBlock.toString());
+                        }
+                    }
+                    else if (type == chunkName ("INST"))
+                    {
+                        HeapBlock <InstChunk> inst;
+                        inst.calloc (jmax ((size_t) length + 1, sizeof (InstChunk)), 1);
+                        input->read (inst, (int) length);
+                        inst->copyTo (metadataValues);
+                    }
+                    else if (type == chunkName ("basc"))
+                    {
+                        AiffFileHelpers::BASCChunk (*input).addToMetadata (metadataValues);
+                    }
+                    else if (type == chunkName ("cate"))
+                    {
+                        metadataValues.set (AiffAudioFormat::appleTag,
+                                            AiffFileHelpers::readCATEChunk (*input, length));;
+                    }
+                    else if ((hasGotVer && hasGotData && hasGotType)
+                              || chunkEnd < input->getPosition()
+                              || input->isExhausted())
+                    {
+                        break;
+                    }
+
+                    input->setPosition (chunkEnd + (chunkEnd & 1)); // (chunks should be aligned to an even byte address)
+                }
+            }
+        }
+
+        if (metadataValues.size() > 0)
+            metadataValues.set ("MetaDataSource", "AIFF");
+    }
+
+    //==============================================================================
+    bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                      int64 startSampleInFile, int numSamples) override
+    {
+        clearSamplesBeyondAvailableLength (destSamples, numDestChannels, startOffsetInDestBuffer,
+                                           startSampleInFile, numSamples, lengthInSamples);
+
+        if (numSamples <= 0)
+            return true;
+
+        input->setPosition (dataChunkStart + startSampleInFile * bytesPerFrame);
+
+        while (numSamples > 0)
+        {
+            const int tempBufSize = 480 * 3 * 4; // (keep this a multiple of 3)
+            char tempBuffer [tempBufSize];
+
+            const int numThisTime = jmin (tempBufSize / bytesPerFrame, numSamples);
+            const int bytesRead = input->read (tempBuffer, numThisTime * bytesPerFrame);
+
+            if (bytesRead < numThisTime * bytesPerFrame)
+            {
+                jassert (bytesRead >= 0);
+                zeromem (tempBuffer + bytesRead, (size_t) (numThisTime * bytesPerFrame - bytesRead));
+            }
+
+            if (littleEndian)
+                copySampleData<AudioData::LittleEndian> (bitsPerSample, usesFloatingPointData,
+                                                         destSamples, startOffsetInDestBuffer, numDestChannels,
+                                                         tempBuffer, (int) numChannels, numThisTime);
+            else
+                copySampleData<AudioData::BigEndian> (bitsPerSample, usesFloatingPointData,
+                                                      destSamples, startOffsetInDestBuffer, numDestChannels,
+                                                      tempBuffer, (int) numChannels, numThisTime);
+
+            startOffsetInDestBuffer += numThisTime;
+            numSamples -= numThisTime;
+        }
+
+        return true;
+    }
+
+    template <typename Endianness>
+    static void copySampleData (unsigned int bitsPerSample, const bool usesFloatingPointData,
+                                int* const* destSamples, int startOffsetInDestBuffer, int numDestChannels,
+                                const void* sourceData, int numChannels, int numSamples) noexcept
+    {
+        switch (bitsPerSample)
+        {
+            case 8:     ReadHelper<AudioData::Int32, AudioData::Int8,  Endianness>::read (destSamples, startOffsetInDestBuffer, numDestChannels, sourceData, numChannels, numSamples); break;
+            case 16:    ReadHelper<AudioData::Int32, AudioData::Int16, Endianness>::read (destSamples, startOffsetInDestBuffer, numDestChannels, sourceData, numChannels, numSamples); break;
+            case 24:    ReadHelper<AudioData::Int32, AudioData::Int24, Endianness>::read (destSamples, startOffsetInDestBuffer, numDestChannels, sourceData, numChannels, numSamples); break;
+            case 32:    if (usesFloatingPointData) ReadHelper<AudioData::Float32, AudioData::Float32, Endianness>::read (destSamples, startOffsetInDestBuffer, numDestChannels, sourceData, numChannels, numSamples);
+                        else                       ReadHelper<AudioData::Int32,   AudioData::Int32,   Endianness>::read (destSamples, startOffsetInDestBuffer, numDestChannels, sourceData, numChannels, numSamples); break;
+            default:    jassertfalse; break;
+        }
+    }
+
+    int bytesPerFrame;
+    int64 dataChunkStart;
+    bool littleEndian;
+
+private:
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AiffAudioFormatReader)
+};
+
+//==============================================================================
+class AiffAudioFormatWriter  : public AudioFormatWriter
+{
+public:
+    AiffAudioFormatWriter (OutputStream* out, double rate,
+                           unsigned int numChans, unsigned int bits,
+                           const StringPairArray& metadataValues)
+        : AudioFormatWriter (out, TRANS (aiffFormatName), rate, numChans, bits),
+          lengthInSamples (0),
+          bytesWritten (0),
+          writeFailed (false)
+    {
+        using namespace AiffFileHelpers;
+
+        if (metadataValues.size() > 0)
+        {
+            // The meta data should have been santised for the AIFF format.
+            // If it was originally sourced from a WAV file the MetaDataSource
+            // key should be removed (or set to "AIFF") once this has been done
+            jassert (metadataValues.getValue ("MetaDataSource", "None") != "WAV");
+
+            MarkChunk::create (markChunk, metadataValues);
+            COMTChunk::create (comtChunk, metadataValues);
+            InstChunk::create (instChunk, metadataValues);
+        }
+
+        headerPosition = out->getPosition();
+        writeHeader();
+    }
+
+    ~AiffAudioFormatWriter()
+    {
+        if ((bytesWritten & 1) != 0)
+            output->writeByte (0);
+
+        writeHeader();
+    }
+
+    //==============================================================================
+    bool write (const int** data, int numSamples) override
+    {
+        jassert (data != nullptr && *data != nullptr); // the input must contain at least one channel!
+
+        if (writeFailed)
+            return false;
+
+        const size_t bytes = (size_t) numSamples * numChannels * bitsPerSample / 8;
+        tempBlock.ensureSize ((size_t) bytes, false);
+
+        switch (bitsPerSample)
+        {
+            case 8:     WriteHelper<AudioData::Int8,  AudioData::Int32, AudioData::BigEndian>::write (tempBlock.getData(), (int) numChannels, data, numSamples); break;
+            case 16:    WriteHelper<AudioData::Int16, AudioData::Int32, AudioData::BigEndian>::write (tempBlock.getData(), (int) numChannels, data, numSamples); break;
+            case 24:    WriteHelper<AudioData::Int24, AudioData::Int32, AudioData::BigEndian>::write (tempBlock.getData(), (int) numChannels, data, numSamples); break;
+            case 32:    WriteHelper<AudioData::Int32, AudioData::Int32, AudioData::BigEndian>::write (tempBlock.getData(), (int) numChannels, data, numSamples); break;
+            default:    jassertfalse; break;
+        }
+
+        if (bytesWritten + bytes >= (size_t) 0xfff00000
+             || ! output->write (tempBlock.getData(), bytes))
+        {
+            // failed to write to disk, so let's try writing the header.
+            // If it's just run out of disk space, then if it does manage
+            // to write the header, we'll still have a useable file..
+            writeHeader();
+            writeFailed = true;
+            return false;
+        }
+        else
+        {
+            bytesWritten += bytes;
+            lengthInSamples += (uint64) numSamples;
+
+            return true;
+        }
+    }
+
+private:
+    MemoryBlock tempBlock, markChunk, comtChunk, instChunk;
+    uint64 lengthInSamples, bytesWritten;
+    int64 headerPosition;
+    bool writeFailed;
+
+    void writeHeader()
+    {
+        using namespace AiffFileHelpers;
+
+        const bool couldSeekOk = output->setPosition (headerPosition);
+        (void) couldSeekOk;
+
+        // if this fails, you've given it an output stream that can't seek! It needs
+        // to be able to seek back to write the header
+        jassert (couldSeekOk);
+
+        const int headerLen = (int) (54 + (markChunk.getSize() > 0 ? markChunk.getSize() + 8 : 0)
+                                        + (comtChunk.getSize() > 0 ? comtChunk.getSize() + 8 : 0)
+                                        + (instChunk.getSize() > 0 ? instChunk.getSize() + 8 : 0));
+        int audioBytes = (int) (lengthInSamples * ((bitsPerSample * numChannels) / 8));
+        audioBytes += (audioBytes & 1);
+
+        output->writeInt (chunkName ("FORM"));
+        output->writeIntBigEndian (headerLen + audioBytes - 8);
+        output->writeInt (chunkName ("AIFF"));
+        output->writeInt (chunkName ("COMM"));
+        output->writeIntBigEndian (18);
+        output->writeShortBigEndian ((short) numChannels);
+        output->writeIntBigEndian ((int) lengthInSamples);
+        output->writeShortBigEndian ((short) bitsPerSample);
+
+        uint8 sampleRateBytes[10] = { 0 };
+
+        if (sampleRate <= 1)
+        {
+            sampleRateBytes[0] = 0x3f;
+            sampleRateBytes[1] = 0xff;
+            sampleRateBytes[2] = 0x80;
+        }
+        else
+        {
+            int mask = 0x40000000;
+            sampleRateBytes[0] = 0x40;
+
+            if (sampleRate >= mask)
+            {
+                jassertfalse;
+                sampleRateBytes[1] = 0x1d;
+            }
+            else
+            {
+                int n = (int) sampleRate;
+
+                int i;
+                for (i = 0; i <= 32 ; ++i)
+                {
+                    if ((n & mask) != 0)
+                        break;
+
+                    mask >>= 1;
+                }
+
+                n = n << (i + 1);
+
+                sampleRateBytes[1] = (uint8) (29 - i);
+                sampleRateBytes[2] = (uint8) ((n >> 24) & 0xff);
+                sampleRateBytes[3] = (uint8) ((n >> 16) & 0xff);
+                sampleRateBytes[4] = (uint8) ((n >>  8) & 0xff);
+                sampleRateBytes[5] = (uint8) (n & 0xff);
+            }
+        }
+
+        output->write (sampleRateBytes, 10);
+
+        if (markChunk.getSize() > 0)
+        {
+            output->writeInt (chunkName ("MARK"));
+            output->writeIntBigEndian ((int) markChunk.getSize());
+            *output << markChunk;
+        }
+
+        if (comtChunk.getSize() > 0)
+        {
+            output->writeInt (chunkName ("COMT"));
+            output->writeIntBigEndian ((int) comtChunk.getSize());
+            *output << comtChunk;
+        }
+
+        if (instChunk.getSize() > 0)
+        {
+            output->writeInt (chunkName ("INST"));
+            output->writeIntBigEndian ((int) instChunk.getSize());
+            *output << instChunk;
+        }
+
+        output->writeInt (chunkName ("SSND"));
+        output->writeIntBigEndian (audioBytes + 8);
+        output->writeInt (0);
+        output->writeInt (0);
+
+        jassert (output->getPosition() == headerLen);
+    }
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AiffAudioFormatWriter)
+};
+
+//==============================================================================
+class MemoryMappedAiffReader   : public MemoryMappedAudioFormatReader
+{
+public:
+    MemoryMappedAiffReader (const File& f, const AiffAudioFormatReader& reader)
+        : MemoryMappedAudioFormatReader (f, reader, reader.dataChunkStart,
+                                         reader.bytesPerFrame * reader.lengthInSamples, reader.bytesPerFrame),
+          littleEndian (reader.littleEndian)
+    {
+    }
+
+    bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                      int64 startSampleInFile, int numSamples) override
+    {
+        clearSamplesBeyondAvailableLength (destSamples, numDestChannels, startOffsetInDestBuffer,
+                                           startSampleInFile, numSamples, lengthInSamples);
+
+        if (map == nullptr || ! mappedSection.contains (Range<int64> (startSampleInFile, startSampleInFile + numSamples)))
+        {
+            jassertfalse; // you must make sure that the window contains all the samples you're going to attempt to read.
+            return false;
+        }
+
+        if (littleEndian)
+            AiffAudioFormatReader::copySampleData<AudioData::LittleEndian>
+                    (bitsPerSample, usesFloatingPointData, destSamples, startOffsetInDestBuffer,
+                     numDestChannels, sampleToPointer (startSampleInFile), (int) numChannels, numSamples);
+        else
+            AiffAudioFormatReader::copySampleData<AudioData::BigEndian>
+                    (bitsPerSample, usesFloatingPointData, destSamples, startOffsetInDestBuffer,
+                     numDestChannels, sampleToPointer (startSampleInFile), (int) numChannels, numSamples);
+
+        return true;
+    }
+
+    void readMaxLevels (int64 startSampleInFile, int64 numSamples,
+                        float& min0, float& max0, float& min1, float& max1)
+    {
+        if (numSamples <= 0)
+        {
+            min0 = max0 = min1 = max1 = 0;
+            return;
+        }
+
+        if (map == nullptr || ! mappedSection.contains (Range<int64> (startSampleInFile, startSampleInFile + numSamples)))
+        {
+            jassertfalse; // you must make sure that the window contains all the samples you're going to attempt to read.
+
+            min0 = max0 = min1 = max1 = 0;
+            return;
+        }
+
+        switch (bitsPerSample)
+        {
+            case 8:     scanMinAndMax<AudioData::UInt8> (startSampleInFile, numSamples, min0, max0, min1, max1); break;
+            case 16:    scanMinAndMax<AudioData::Int16> (startSampleInFile, numSamples, min0, max0, min1, max1); break;
+            case 24:    scanMinAndMax<AudioData::Int24> (startSampleInFile, numSamples, min0, max0, min1, max1); break;
+            case 32:    if (usesFloatingPointData) scanMinAndMax<AudioData::Float32> (startSampleInFile, numSamples, min0, max0, min1, max1);
+                        else                       scanMinAndMax<AudioData::Int32>   (startSampleInFile, numSamples, min0, max0, min1, max1); break;
+            default:    jassertfalse; break;
+        }
+    }
+
+private:
+    const bool littleEndian;
+
+    template <typename SampleType>
+    void scanMinAndMax (int64 startSampleInFile, int64 numSamples,
+                        float& min0, float& max0, float& min1, float& max1) const noexcept
+    {
+        scanMinAndMax2<SampleType> (0, startSampleInFile, numSamples, min0, max0);
+
+        if (numChannels > 1)
+            scanMinAndMax2<SampleType> (1, startSampleInFile, numSamples, min1, max1);
+        else
+            min1 = max1 = 0;
+    }
+
+    template <typename SampleType>
+    void scanMinAndMax2 (int channel, int64 startSampleInFile, int64 numSamples, float& mn, float& mx) const noexcept
+    {
+        if (littleEndian)
+            scanMinAndMaxInterleaved<SampleType, AudioData::LittleEndian> (channel, startSampleInFile, numSamples, mn, mx);
+        else
+            scanMinAndMaxInterleaved<SampleType, AudioData::BigEndian>    (channel, startSampleInFile, numSamples, mn, mx);
+    }
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (MemoryMappedAiffReader)
+};
+
+//==============================================================================
+AiffAudioFormat::AiffAudioFormat()
+    : AudioFormat (TRANS (aiffFormatName), StringArray (aiffExtensions))
+{
+}
+
+AiffAudioFormat::~AiffAudioFormat()
+{
+}
+
+Array<int> AiffAudioFormat::getPossibleSampleRates()
+{
+    const int rates[] = { 22050, 32000, 44100, 48000, 88200, 96000, 176400, 192000, 0 };
+    return Array <int> (rates);
+}
+
+Array<int> AiffAudioFormat::getPossibleBitDepths()
+{
+    const int depths[] = { 8, 16, 24, 0 };
+    return Array <int> (depths);
+}
+
+bool AiffAudioFormat::canDoStereo() { return true; }
+bool AiffAudioFormat::canDoMono()   { return true; }
+
+#if JUCE_MAC
+bool AiffAudioFormat::canHandleFile (const File& f)
+{
+    if (AudioFormat::canHandleFile (f))
+        return true;
+
+    const OSType type = f.getMacOSType();
+
+    // (NB: written as hex to avoid four-char-constant warnings)
+    return type == 0x41494646 /* AIFF */ || type == 0x41494643 /* AIFC */
+        || type == 0x61696666 /* aiff */ || type == 0x61696663 /* aifc */;
+}
+#endif
+
+AudioFormatReader* AiffAudioFormat::createReaderFor (InputStream* sourceStream, const bool deleteStreamIfOpeningFails)
+{
+    ScopedPointer <AiffAudioFormatReader> w (new AiffAudioFormatReader (sourceStream));
+
+    if (w->sampleRate > 0 && w->numChannels > 0)
+        return w.release();
+
+    if (! deleteStreamIfOpeningFails)
+        w->input = nullptr;
+
+    return nullptr;
+}
+
+MemoryMappedAudioFormatReader* AiffAudioFormat::createMemoryMappedReader (const File& file)
+{
+    if (FileInputStream* fin = file.createInputStream())
+    {
+        AiffAudioFormatReader reader (fin);
+
+        if (reader.lengthInSamples > 0)
+            return new MemoryMappedAiffReader (file, reader);
+    }
+
+    return nullptr;
+}
+
+AudioFormatWriter* AiffAudioFormat::createWriterFor (OutputStream* out,
+                                                     double sampleRate,
+                                                     unsigned int numberOfChannels,
+                                                     int bitsPerSample,
+                                                     const StringPairArray& metadataValues,
+                                                     int /*qualityOptionIndex*/)
+{
+    if (getPossibleBitDepths().contains (bitsPerSample))
+        return new AiffAudioFormatWriter (out, sampleRate, numberOfChannels, (unsigned int) bitsPerSample, metadataValues);
+
+    return nullptr;
+}

source/modules/juce_audio_formats/codecs/juce_AiffAudioFormat.h

@@ -0,0 +1,84 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+//==============================================================================
+/**
+    Reads and Writes AIFF format audio files.
+
+    @see AudioFormat
+*/
+class JUCE_API  AiffAudioFormat  : public AudioFormat
+{
+public:
+    //==============================================================================
+    /** Creates an format object. */
+    AiffAudioFormat();
+
+    /** Destructor. */
+    ~AiffAudioFormat();
+
+    //==============================================================================
+    /** Metadata property name used when reading a aiff file with a basc chunk. */
+    static const char* const appleOneShot;
+    /** Metadata property name used when reading a aiff file with a basc chunk. */
+    static const char* const appleRootSet;
+    /** Metadata property name used when reading a aiff file with a basc chunk. */
+    static const char* const appleRootNote;
+    /** Metadata property name used when reading a aiff file with a basc chunk. */
+    static const char* const appleBeats;
+    /** Metadata property name used when reading a aiff file with a basc chunk. */
+    static const char* const appleDenominator;
+    /** Metadata property name used when reading a aiff file with a basc chunk. */
+    static const char* const appleNumerator;
+    /** Metadata property name used when reading a aiff file with a basc chunk. */
+    static const char* const appleTag;
+    /** Metadata property name used when reading a aiff file with a basc chunk. */
+    static const char* const appleKey;
+
+    //==============================================================================
+    Array<int> getPossibleSampleRates() override;
+    Array<int> getPossibleBitDepths() override;
+    bool canDoStereo() override;
+    bool canDoMono() override;
+
+   #if JUCE_MAC
+    bool canHandleFile (const File& fileToTest) override;
+   #endif
+
+    //==============================================================================
+    AudioFormatReader* createReaderFor (InputStream* sourceStream,
+                                        bool deleteStreamIfOpeningFails) override;
+
+    MemoryMappedAudioFormatReader* createMemoryMappedReader (const File&) override;
+
+    AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
+                                        double sampleRateToUse,
+                                        unsigned int numberOfChannels,
+                                        int bitsPerSample,
+                                        const StringPairArray& metadataValues,
+                                        int qualityOptionIndex) override;
+
+private:
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR(AiffAudioFormat)
+};

source/modules/juce_audio_formats/codecs/juce_CoreAudioFormat.cpp

@@ -0,0 +1,528 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#if JUCE_MAC || JUCE_IOS
+
+//==============================================================================
+namespace
+{
+    const char* const coreAudioFormatName = "CoreAudio supported file";
+
+    StringArray findFileExtensionsForCoreAudioCodecs()
+    {
+        StringArray extensionsArray;
+        CFArrayRef extensions = nullptr;
+        UInt32 sizeOfArray = sizeof (extensions);
+
+        if (AudioFileGetGlobalInfo (kAudioFileGlobalInfo_AllExtensions, 0, 0, &sizeOfArray, &extensions) == noErr)
+        {
+            const CFIndex numValues = CFArrayGetCount (extensions);
+
+            for (CFIndex i = 0; i < numValues; ++i)
+                extensionsArray.add ("." + String::fromCFString ((CFStringRef) CFArrayGetValueAtIndex (extensions, i)));
+
+            CFRelease (extensions);
+        }
+
+        return extensionsArray;
+    }
+}
+
+//==============================================================================
+const char* const CoreAudioFormat::midiDataBase64   = "midiDataBase64";
+const char* const CoreAudioFormat::tempo            = "tempo";
+const char* const CoreAudioFormat::timeSig          = "time signature";
+const char* const CoreAudioFormat::keySig           = "key signature";
+
+//==============================================================================
+struct CoreAudioFormatMetatdata
+{
+    static uint32 chunkName (const char* const name) noexcept   { return ByteOrder::bigEndianInt (name); }
+
+    //==============================================================================
+    struct FileHeader
+    {
+        FileHeader (InputStream& input)
+        {
+            fileType    = (uint32) input.readIntBigEndian();
+            fileVersion = (uint16) input.readShortBigEndian();
+            fileFlags   = (uint16) input.readShortBigEndian();
+        }
+
+        uint32 fileType;
+        uint16 fileVersion;
+        uint16 fileFlags;
+    };
+
+    //==============================================================================
+    struct ChunkHeader
+    {
+        ChunkHeader (InputStream& input)
+        {
+            chunkType = (uint32) input.readIntBigEndian();
+            chunkSize = (int64)  input.readInt64BigEndian();
+        }
+
+        uint32 chunkType;
+        int64 chunkSize;
+    };
+
+    //==============================================================================
+    struct AudioDescriptionChunk
+    {
+        AudioDescriptionChunk (InputStream& input)
+        {
+            sampleRate          = input.readDoubleBigEndian();
+            formatID            = (uint32) input.readIntBigEndian();
+            formatFlags         = (uint32) input.readIntBigEndian();
+            bytesPerPacket      = (uint32) input.readIntBigEndian();
+            framesPerPacket     = (uint32) input.readIntBigEndian();
+            channelsPerFrame    = (uint32) input.readIntBigEndian();
+            bitsPerChannel      = (uint32) input.readIntBigEndian();
+        }
+
+        double sampleRate;
+        uint32 formatID;
+        uint32 formatFlags;
+        uint32 bytesPerPacket;
+        uint32 framesPerPacket;
+        uint32 channelsPerFrame;
+        uint32 bitsPerChannel;
+    };
+
+    //==============================================================================
+    struct UserDefinedChunk
+    {
+        UserDefinedChunk (InputStream& input, int64 size)
+        {
+            // a user defined chunk contains 16 bytes of a UUID first
+            uuid[1] = input.readInt64BigEndian();
+            uuid[0] = input.readInt64BigEndian();
+
+            input.skipNextBytes (size - 16);
+        }
+
+        int64 uuid[2];
+    };
+
+    //==============================================================================
+    static StringPairArray parseMidiChunk (InputStream& input, int64 size)
+    {
+        const int64 originalPosition = input.getPosition();
+
+        MemoryBlock midiBlock;
+        input.readIntoMemoryBlock (midiBlock, (ssize_t) size);
+        MemoryInputStream midiInputStream (midiBlock, false);
+
+        StringPairArray midiMetadata;
+        MidiFile midiFile;
+
+        if (midiFile.readFrom (midiInputStream))
+        {
+            midiMetadata.set (CoreAudioFormat::midiDataBase64, midiBlock.toBase64Encoding());
+
+            findTempoEvents (midiFile, midiMetadata);
+            findTimeSigEvents (midiFile, midiMetadata);
+            findKeySigEvents (midiFile, midiMetadata);
+        }
+
+        input.setPosition (originalPosition + size);
+        return midiMetadata;
+    }
+
+    static void findTempoEvents (MidiFile& midiFile, StringPairArray& midiMetadata)
+    {
+        MidiMessageSequence tempoEvents;
+        midiFile.findAllTempoEvents (tempoEvents);
+
+        const int numTempoEvents = tempoEvents.getNumEvents();
+        MemoryOutputStream tempoSequence;
+
+        for (int i = 0; i < numTempoEvents; ++i)
+        {
+            const double tempo = getTempoFromTempoMetaEvent (tempoEvents.getEventPointer (i));
+
+            if (tempo > 0.0)
+            {
+                if (i == 0)
+                    midiMetadata.set (CoreAudioFormat::tempo, String (tempo));
+
+                if (numTempoEvents > 1)
+                    tempoSequence << String (tempo) << ',' << tempoEvents.getEventTime (i) << ';';
+            }
+        }
+
+        if (tempoSequence.getDataSize() > 0)
+            midiMetadata.set ("tempo sequence", tempoSequence.toUTF8());
+    }
+
+    static double getTempoFromTempoMetaEvent (MidiMessageSequence::MidiEventHolder* holder)
+    {
+        if (holder != nullptr)
+        {
+            const MidiMessage& midiMessage = holder->message;
+
+            if (midiMessage.isTempoMetaEvent())
+            {
+                const double tempoSecondsPerQuarterNote = midiMessage.getTempoSecondsPerQuarterNote();
+
+                if (tempoSecondsPerQuarterNote > 0.0)
+                    return 60.0 / tempoSecondsPerQuarterNote;
+            }
+        }
+
+        return 0.0;
+    }
+
+    static void findTimeSigEvents (MidiFile& midiFile, StringPairArray& midiMetadata)
+    {
+        MidiMessageSequence timeSigEvents;
+        midiFile.findAllTimeSigEvents (timeSigEvents);
+        const int numTimeSigEvents = timeSigEvents.getNumEvents();
+
+        MemoryOutputStream timeSigSequence;
+
+        for (int i = 0; i < numTimeSigEvents; ++i)
+        {
+            int numerator, denominator;
+            timeSigEvents.getEventPointer(i)->message.getTimeSignatureInfo (numerator, denominator);
+
+            String timeSigString;
+            timeSigString << numerator << '/' << denominator;
+
+            if (i == 0)
+                midiMetadata.set (CoreAudioFormat::timeSig, timeSigString);
+
+            if (numTimeSigEvents > 1)
+                timeSigSequence << timeSigString << ',' << timeSigEvents.getEventTime (i) << ';';
+        }
+
+        if (timeSigSequence.getDataSize() > 0)
+            midiMetadata.set ("time signature sequence", timeSigSequence.toUTF8());
+    }
+
+    static void findKeySigEvents (MidiFile& midiFile, StringPairArray& midiMetadata)
+    {
+        MidiMessageSequence keySigEvents;
+        midiFile.findAllKeySigEvents (keySigEvents);
+        const int numKeySigEvents = keySigEvents.getNumEvents();
+
+        MemoryOutputStream keySigSequence;
+
+        for (int i = 0; i < numKeySigEvents; ++i)
+        {
+            const MidiMessage& message (keySigEvents.getEventPointer (i)->message);
+            const int key = jlimit (0, 14, message.getKeySignatureNumberOfSharpsOrFlats() + 7);
+            const bool isMajor = message.isKeySignatureMajorKey();
+
+            static const char* majorKeys[] = { "Cb", "Gb", "Db", "Ab", "Eb", "Bb", "F", "C", "G", "D", "A", "E", "B", "F#", "C#" };
+            static const char* minorKeys[] = { "Ab", "Eb", "Bb", "F", "C", "G", "D", "A", "E", "B", "F#", "C#", "G#", "D#", "A#" };
+
+            String keySigString (isMajor ? majorKeys[key]
+                                         : minorKeys[key]);
+
+            if (! isMajor)
+                keySigString << 'm';
+
+            if (i == 0)
+                midiMetadata.set (CoreAudioFormat::keySig, keySigString);
+
+            if (numKeySigEvents > 1)
+                keySigSequence << keySigString << ',' << keySigEvents.getEventTime (i) << ';';
+        }
+
+        if (keySigSequence.getDataSize() > 0)
+            midiMetadata.set ("key signature sequence", keySigSequence.toUTF8());
+    }
+
+    //==============================================================================
+    static StringPairArray parseInformationChunk (InputStream& input)
+    {
+        StringPairArray infoStrings;
+        const uint32 numEntries = (uint32) input.readIntBigEndian();
+
+        for (uint32 i = 0; i < numEntries; ++i)
+            infoStrings.set (input.readString(), input.readString());
+
+        return infoStrings;
+    }
+
+    //==============================================================================
+    static bool read (InputStream& input, StringPairArray& metadataValues)
+    {
+        const int64 originalPos = input.getPosition();
+
+        const FileHeader cafFileHeader (input);
+        const bool isCafFile = cafFileHeader.fileType == chunkName ("caff");
+
+        if (isCafFile)
+        {
+            while (! input.isExhausted())
+            {
+                const ChunkHeader chunkHeader (input);
+
+                if (chunkHeader.chunkType == chunkName ("desc"))
+                {
+                    AudioDescriptionChunk audioDescriptionChunk (input);
+                }
+                else if (chunkHeader.chunkType == chunkName ("uuid"))
+                {
+                    UserDefinedChunk userDefinedChunk (input, chunkHeader.chunkSize);
+                }
+                else if (chunkHeader.chunkType == chunkName ("data"))
+                {
+                    // -1 signifies an unknown data size so the data has to be at the
+                    // end of the file so we must have finished the header
+
+                    if (chunkHeader.chunkSize == -1)
+                        break;
+
+                    input.skipNextBytes (chunkHeader.chunkSize);
+                }
+                else if (chunkHeader.chunkType == chunkName ("midi"))
+                {
+                    metadataValues.addArray (parseMidiChunk (input, chunkHeader.chunkSize));
+                }
+                else if (chunkHeader.chunkType == chunkName ("info"))
+                {
+                    metadataValues.addArray (parseInformationChunk (input));
+                }
+                else
+                {
+                    // we aren't decoding this chunk yet so just skip over it
+                    input.skipNextBytes (chunkHeader.chunkSize);
+                }
+            }
+        }
+
+        input.setPosition (originalPos);
+
+        return isCafFile;
+    }
+};
+
+//==============================================================================
+class CoreAudioReader : public AudioFormatReader
+{
+public:
+    CoreAudioReader (InputStream* const inp)
+        : AudioFormatReader (inp, TRANS (coreAudioFormatName)),
+          ok (false), lastReadPosition (0)
+    {
+        usesFloatingPointData = true;
+        bitsPerSample = 32;
+
+        if (input != nullptr)
+            CoreAudioFormatMetatdata::read (*input, metadataValues);
+
+        OSStatus status = AudioFileOpenWithCallbacks (this,
+                                                      &readCallback,
+                                                      nullptr,  // write needs to be null to avoid permisisions errors
+                                                      &getSizeCallback,
+                                                      nullptr,  // setSize needs to be null to avoid permisisions errors
+                                                      0,        // AudioFileTypeID inFileTypeHint
+                                                      &audioFileID);
+        if (status == noErr)
+        {
+            status = ExtAudioFileWrapAudioFileID (audioFileID, false, &audioFileRef);
+
+            if (status == noErr)
+            {
+                AudioStreamBasicDescription sourceAudioFormat;
+                UInt32 audioStreamBasicDescriptionSize = sizeof (AudioStreamBasicDescription);
+                ExtAudioFileGetProperty (audioFileRef,
+                                         kExtAudioFileProperty_FileDataFormat,
+                                         &audioStreamBasicDescriptionSize,
+                                         &sourceAudioFormat);
+
+                numChannels = sourceAudioFormat.mChannelsPerFrame;
+                sampleRate  = sourceAudioFormat.mSampleRate;
+
+                UInt32 sizeOfLengthProperty = sizeof (int64);
+                ExtAudioFileGetProperty (audioFileRef,
+                                         kExtAudioFileProperty_FileLengthFrames,
+                                         &sizeOfLengthProperty,
+                                         &lengthInSamples);
+
+                destinationAudioFormat.mSampleRate       = sampleRate;
+                destinationAudioFormat.mFormatID         = kAudioFormatLinearPCM;
+                destinationAudioFormat.mFormatFlags      = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsNonInterleaved | kAudioFormatFlagsNativeEndian;
+                destinationAudioFormat.mBitsPerChannel   = sizeof (float) * 8;
+                destinationAudioFormat.mChannelsPerFrame = numChannels;
+                destinationAudioFormat.mBytesPerFrame    = sizeof (float);
+                destinationAudioFormat.mFramesPerPacket  = 1;
+                destinationAudioFormat.mBytesPerPacket   = destinationAudioFormat.mFramesPerPacket * destinationAudioFormat.mBytesPerFrame;
+
+                status = ExtAudioFileSetProperty (audioFileRef,
+                                                  kExtAudioFileProperty_ClientDataFormat,
+                                                  sizeof (AudioStreamBasicDescription),
+                                                  &destinationAudioFormat);
+                if (status == noErr)
+                {
+                    bufferList.malloc (1, sizeof (AudioBufferList) + numChannels * sizeof (AudioBuffer));
+                    bufferList->mNumberBuffers = numChannels;
+                    ok = true;
+                }
+            }
+        }
+    }
+
+    ~CoreAudioReader()
+    {
+        ExtAudioFileDispose (audioFileRef);
+        AudioFileClose (audioFileID);
+    }
+
+    //==============================================================================
+    bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                      int64 startSampleInFile, int numSamples) override
+    {
+        clearSamplesBeyondAvailableLength (destSamples, numDestChannels, startOffsetInDestBuffer,
+                                           startSampleInFile, numSamples, lengthInSamples);
+
+        if (numSamples <= 0)
+            return true;
+
+        if (lastReadPosition != startSampleInFile)
+        {
+            OSStatus status = ExtAudioFileSeek (audioFileRef, startSampleInFile);
+            if (status != noErr)
+                return false;
+
+            lastReadPosition = startSampleInFile;
+        }
+
+        while (numSamples > 0)
+        {
+            const int numThisTime = jmin (8192, numSamples);
+            const size_t numBytes = sizeof (float) * (size_t) numThisTime;
+
+            audioDataBlock.ensureSize (numBytes * numChannels, false);
+            float* data = static_cast<float*> (audioDataBlock.getData());
+
+            for (int j = (int) numChannels; --j >= 0;)
+            {
+                bufferList->mBuffers[j].mNumberChannels = 1;
+                bufferList->mBuffers[j].mDataByteSize = (UInt32) numBytes;
+                bufferList->mBuffers[j].mData = data;
+                data += numThisTime;
+            }
+
+            UInt32 numFramesToRead = (UInt32) numThisTime;
+            OSStatus status = ExtAudioFileRead (audioFileRef, &numFramesToRead, bufferList);
+            if (status != noErr)
+                return false;
+
+            for (int i = numDestChannels; --i >= 0;)
+            {
+                if (destSamples[i] != nullptr)
+                {
+                    if (i < (int) numChannels)
+                        memcpy (destSamples[i] + startOffsetInDestBuffer, bufferList->mBuffers[i].mData, numBytes);
+                    else
+                        zeromem (destSamples[i] + startOffsetInDestBuffer, numBytes);
+                }
+            }
+
+            startOffsetInDestBuffer += numThisTime;
+            numSamples -= numThisTime;
+            lastReadPosition += numThisTime;
+        }
+
+        return true;
+    }
+
+    bool ok;
+
+private:
+    AudioFileID audioFileID;
+    ExtAudioFileRef audioFileRef;
+    AudioStreamBasicDescription destinationAudioFormat;
+    MemoryBlock audioDataBlock;
+    HeapBlock<AudioBufferList> bufferList;
+    int64 lastReadPosition;
+
+    static SInt64 getSizeCallback (void* inClientData)
+    {
+        return static_cast<CoreAudioReader*> (inClientData)->input->getTotalLength();
+    }
+
+    static OSStatus readCallback (void* inClientData,
+                                  SInt64 inPosition,
+                                  UInt32 requestCount,
+                                  void* buffer,
+                                  UInt32* actualCount)
+    {
+        CoreAudioReader* const reader = static_cast<CoreAudioReader*> (inClientData);
+
+        reader->input->setPosition (inPosition);
+        *actualCount = (UInt32) reader->input->read (buffer, (int) requestCount);
+
+        return noErr;
+    }
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (CoreAudioReader)
+};
+
+//==============================================================================
+CoreAudioFormat::CoreAudioFormat()
+    : AudioFormat (TRANS (coreAudioFormatName), findFileExtensionsForCoreAudioCodecs())
+{
+}
+
+CoreAudioFormat::~CoreAudioFormat() {}
+
+Array<int> CoreAudioFormat::getPossibleSampleRates()    { return Array<int>(); }
+Array<int> CoreAudioFormat::getPossibleBitDepths()      { return Array<int>(); }
+
+bool CoreAudioFormat::canDoStereo()     { return true; }
+bool CoreAudioFormat::canDoMono()       { return true; }
+
+//==============================================================================
+AudioFormatReader* CoreAudioFormat::createReaderFor (InputStream* sourceStream,
+                                                     bool deleteStreamIfOpeningFails)
+{
+    ScopedPointer<CoreAudioReader> r (new CoreAudioReader (sourceStream));
+
+    if (r->ok)
+        return r.release();
+
+    if (! deleteStreamIfOpeningFails)
+        r->input = nullptr;
+
+    return nullptr;
+}
+
+AudioFormatWriter* CoreAudioFormat::createWriterFor (OutputStream*,
+                                                     double /*sampleRateToUse*/,
+                                                     unsigned int /*numberOfChannels*/,
+                                                     int /*bitsPerSample*/,
+                                                     const StringPairArray& /*metadataValues*/,
+                                                     int /*qualityOptionIndex*/)
+{
+    jassertfalse; // not yet implemented!
+    return nullptr;
+}
+
+#endif

source/modules/juce_audio_formats/codecs/juce_CoreAudioFormat.h

@@ -0,0 +1,77 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#if JUCE_MAC || JUCE_IOS || DOXYGEN
+
+//==============================================================================
+/**
+    OSX and iOS only - This uses the AudioToolbox framework to read any audio
+    format that the system has a codec for.
+
+    This should be able to understand formats such as mp3, m4a, etc.
+
+    @see AudioFormat
+ */
+class JUCE_API  CoreAudioFormat     : public AudioFormat
+{
+public:
+    //==============================================================================
+    /** Creates a format object. */
+    CoreAudioFormat();
+
+    /** Destructor. */
+    ~CoreAudioFormat();
+
+    //==============================================================================
+    /** Metadata property name used when reading a caf file with a MIDI chunk. */
+    static const char* const midiDataBase64;
+    /** Metadata property name used when reading a caf file with tempo information. */
+    static const char* const tempo;
+    /** Metadata property name used when reading a caf file time signature information. */
+    static const char* const timeSig;
+    /** Metadata property name used when reading a caf file time signature information. */
+    static const char* const keySig;
+
+    //==============================================================================
+    Array<int> getPossibleSampleRates() override;
+    Array<int> getPossibleBitDepths() override;
+    bool canDoStereo() override;
+    bool canDoMono() override;
+
+    //==============================================================================
+    AudioFormatReader* createReaderFor (InputStream*,
+                                        bool deleteStreamIfOpeningFails) override;
+
+    AudioFormatWriter* createWriterFor (OutputStream*,
+                                        double sampleRateToUse,
+                                        unsigned int numberOfChannels,
+                                        int bitsPerSample,
+                                        const StringPairArray& metadataValues,
+                                        int qualityOptionIndex) override;
+
+private:
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (CoreAudioFormat)
+};
+
+#endif

source/modules/juce_audio_formats/codecs/juce_FlacAudioFormat.cpp

@@ -0,0 +1,545 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#if JUCE_USE_FLAC
+
+namespace FlacNamespace
+{
+#if JUCE_INCLUDE_FLAC_CODE || ! defined (JUCE_INCLUDE_FLAC_CODE)
+
+ #undef VERSION
+ #define VERSION "1.2.1"
+
+ #define FLAC__NO_DLL 1
+
+ #if JUCE_MSVC
+  #pragma warning (disable: 4267 4127 4244 4996 4100 4701 4702 4013 4133 4206 4312 4505 4365 4005 4334 181 111)
+ #endif
+
+ #if JUCE_MAC
+  #define FLAC__SYS_DARWIN 1
+ #endif
+
+ #ifndef SIZE_MAX
+  #define SIZE_MAX 0xffffffff
+ #endif
+
+ #if JUCE_CLANG
+  #pragma clang diagnostic push
+  #pragma clang diagnostic ignored "-Wconversion"
+  #pragma clang diagnostic ignored "-Wshadow"
+ #endif
+
+ #define __STDC_LIMIT_MACROS 1
+ #include "flac/all.h"
+ #include "flac/libFLAC/bitmath.c"
+ #include "flac/libFLAC/bitreader.c"
+ #include "flac/libFLAC/bitwriter.c"
+ #include "flac/libFLAC/cpu.c"
+ #include "flac/libFLAC/crc.c"
+ #include "flac/libFLAC/fixed.c"
+ #include "flac/libFLAC/float.c"
+ #include "flac/libFLAC/format.c"
+ #include "flac/libFLAC/lpc_flac.c"
+ #include "flac/libFLAC/md5.c"
+ #include "flac/libFLAC/memory.c"
+ #include "flac/libFLAC/stream_decoder.c"
+ #include "flac/libFLAC/stream_encoder.c"
+ #include "flac/libFLAC/stream_encoder_framing.c"
+ #include "flac/libFLAC/window_flac.c"
+ #undef VERSION
+#else
+ #include <FLAC/all.h>
+#endif
+
+ #if JUCE_CLANG
+  #pragma clang diagnostic pop
+ #endif
+}
+
+#undef max
+#undef min
+
+//==============================================================================
+static const char* const flacFormatName = "FLAC file";
+static const char* const flacExtensions[] = { ".flac", 0 };
+
+
+//==============================================================================
+class FlacReader  : public AudioFormatReader
+{
+public:
+    //==============================================================================
+    FlacReader (InputStream* const in)
+        : AudioFormatReader (in, TRANS (flacFormatName)),
+          reservoir (2, 0),
+          reservoirStart (0),
+          samplesInReservoir (0),
+          scanningForLength (false)
+    {
+        using namespace FlacNamespace;
+        lengthInSamples = 0;
+
+        decoder = FLAC__stream_decoder_new();
+
+        ok = FLAC__stream_decoder_init_stream (decoder,
+                                               readCallback_, seekCallback_, tellCallback_, lengthCallback_,
+                                               eofCallback_, writeCallback_, metadataCallback_, errorCallback_,
+                                               this) == FLAC__STREAM_DECODER_INIT_STATUS_OK;
+
+        if (ok)
+        {
+            FLAC__stream_decoder_process_until_end_of_metadata (decoder);
+
+            if (lengthInSamples == 0 && sampleRate > 0)
+            {
+                // the length hasn't been stored in the metadata, so we'll need to
+                // work it out the length the hard way, by scanning the whole file..
+                scanningForLength = true;
+                FLAC__stream_decoder_process_until_end_of_stream (decoder);
+                scanningForLength = false;
+                const int64 tempLength = lengthInSamples;
+
+                FLAC__stream_decoder_reset (decoder);
+                FLAC__stream_decoder_process_until_end_of_metadata (decoder);
+                lengthInSamples = tempLength;
+            }
+        }
+    }
+
+    ~FlacReader()
+    {
+        FlacNamespace::FLAC__stream_decoder_delete (decoder);
+    }
+
+    void useMetadata (const FlacNamespace::FLAC__StreamMetadata_StreamInfo& info)
+    {
+        sampleRate = info.sample_rate;
+        bitsPerSample = info.bits_per_sample;
+        lengthInSamples = (unsigned int) info.total_samples;
+        numChannels = info.channels;
+
+        reservoir.setSize ((int) numChannels, 2 * (int) info.max_blocksize, false, false, true);
+    }
+
+    // returns the number of samples read
+    bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                      int64 startSampleInFile, int numSamples) override
+    {
+        using namespace FlacNamespace;
+
+        if (! ok)
+            return false;
+
+        while (numSamples > 0)
+        {
+            if (startSampleInFile >= reservoirStart
+                 && startSampleInFile < reservoirStart + samplesInReservoir)
+            {
+                const int num = (int) jmin ((int64) numSamples,
+                                            reservoirStart + samplesInReservoir - startSampleInFile);
+
+                jassert (num > 0);
+
+                for (int i = jmin (numDestChannels, reservoir.getNumChannels()); --i >= 0;)
+                    if (destSamples[i] != nullptr)
+                        memcpy (destSamples[i] + startOffsetInDestBuffer,
+                                reservoir.getSampleData (i, (int) (startSampleInFile - reservoirStart)),
+                                sizeof (int) * (size_t) num);
+
+                startOffsetInDestBuffer += num;
+                startSampleInFile += num;
+                numSamples -= num;
+            }
+            else
+            {
+                if (startSampleInFile >= (int) lengthInSamples)
+                {
+                    samplesInReservoir = 0;
+                }
+                else if (startSampleInFile < reservoirStart
+                          || startSampleInFile > reservoirStart + jmax (samplesInReservoir, 511))
+                {
+                    // had some problems with flac crashing if the read pos is aligned more
+                    // accurately than this. Probably fixed in newer versions of the library, though.
+                    reservoirStart = (int) (startSampleInFile & ~511);
+                    samplesInReservoir = 0;
+                    FLAC__stream_decoder_seek_absolute (decoder, (FLAC__uint64) reservoirStart);
+                }
+                else
+                {
+                    reservoirStart += samplesInReservoir;
+                    samplesInReservoir = 0;
+                    FLAC__stream_decoder_process_single (decoder);
+                }
+
+                if (samplesInReservoir == 0)
+                    break;
+            }
+        }
+
+        if (numSamples > 0)
+        {
+            for (int i = numDestChannels; --i >= 0;)
+                if (destSamples[i] != nullptr)
+                    zeromem (destSamples[i] + startOffsetInDestBuffer, sizeof (int) * (size_t) numSamples);
+        }
+
+        return true;
+    }
+
+    void useSamples (const FlacNamespace::FLAC__int32* const buffer[], int numSamples)
+    {
+        if (scanningForLength)
+        {
+            lengthInSamples += numSamples;
+        }
+        else
+        {
+            if (numSamples > reservoir.getNumSamples())
+                reservoir.setSize ((int) numChannels, numSamples, false, false, true);
+
+            const unsigned int bitsToShift = 32 - bitsPerSample;
+
+            for (int i = 0; i < (int) numChannels; ++i)
+            {
+                const FlacNamespace::FLAC__int32* src = buffer[i];
+
+                int n = i;
+                while (src == 0 && n > 0)
+                    src = buffer [--n];
+
+                if (src != nullptr)
+                {
+                    int* const dest = reinterpret_cast<int*> (reservoir.getSampleData(i));
+
+                    for (int j = 0; j < numSamples; ++j)
+                        dest[j] = src[j] << bitsToShift;
+                }
+            }
+
+            samplesInReservoir = numSamples;
+        }
+    }
+
+    //==============================================================================
+    static FlacNamespace::FLAC__StreamDecoderReadStatus readCallback_ (const FlacNamespace::FLAC__StreamDecoder*, FlacNamespace::FLAC__byte buffer[], size_t* bytes, void* client_data)
+    {
+        using namespace FlacNamespace;
+        *bytes = (size_t) static_cast<const FlacReader*> (client_data)->input->read (buffer, (int) *bytes);
+        return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
+    }
+
+    static FlacNamespace::FLAC__StreamDecoderSeekStatus seekCallback_ (const FlacNamespace::FLAC__StreamDecoder*, FlacNamespace::FLAC__uint64 absolute_byte_offset, void* client_data)
+    {
+        using namespace FlacNamespace;
+        static_cast<const FlacReader*> (client_data)->input->setPosition ((int) absolute_byte_offset);
+        return FLAC__STREAM_DECODER_SEEK_STATUS_OK;
+    }
+
+    static FlacNamespace::FLAC__StreamDecoderTellStatus tellCallback_ (const FlacNamespace::FLAC__StreamDecoder*, FlacNamespace::FLAC__uint64* absolute_byte_offset, void* client_data)
+    {
+        using namespace FlacNamespace;
+        *absolute_byte_offset = (uint64) static_cast<const FlacReader*> (client_data)->input->getPosition();
+        return FLAC__STREAM_DECODER_TELL_STATUS_OK;
+    }
+
+    static FlacNamespace::FLAC__StreamDecoderLengthStatus lengthCallback_ (const FlacNamespace::FLAC__StreamDecoder*, FlacNamespace::FLAC__uint64* stream_length, void* client_data)
+    {
+        using namespace FlacNamespace;
+        *stream_length = (uint64) static_cast<const FlacReader*> (client_data)->input->getTotalLength();
+        return FLAC__STREAM_DECODER_LENGTH_STATUS_OK;
+    }
+
+    static FlacNamespace::FLAC__bool eofCallback_ (const FlacNamespace::FLAC__StreamDecoder*, void* client_data)
+    {
+        return static_cast<const FlacReader*> (client_data)->input->isExhausted();
+    }
+
+    static FlacNamespace::FLAC__StreamDecoderWriteStatus writeCallback_ (const FlacNamespace::FLAC__StreamDecoder*,
+                                                                         const FlacNamespace::FLAC__Frame* frame,
+                                                                         const FlacNamespace::FLAC__int32* const buffer[],
+                                                                         void* client_data)
+    {
+        using namespace FlacNamespace;
+        static_cast<FlacReader*> (client_data)->useSamples (buffer, (int) frame->header.blocksize);
+        return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
+    }
+
+    static void metadataCallback_ (const FlacNamespace::FLAC__StreamDecoder*,
+                                   const FlacNamespace::FLAC__StreamMetadata* metadata,
+                                   void* client_data)
+    {
+        static_cast<FlacReader*> (client_data)->useMetadata (metadata->data.stream_info);
+    }
+
+    static void errorCallback_ (const FlacNamespace::FLAC__StreamDecoder*, FlacNamespace::FLAC__StreamDecoderErrorStatus, void*)
+    {
+    }
+
+private:
+    FlacNamespace::FLAC__StreamDecoder* decoder;
+    AudioSampleBuffer reservoir;
+    int reservoirStart, samplesInReservoir;
+    bool ok, scanningForLength;
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (FlacReader)
+};
+
+
+//==============================================================================
+class FlacWriter  : public AudioFormatWriter
+{
+public:
+    FlacWriter (OutputStream* const out, double rate, uint32 numChans, uint32 bits, int qualityOptionIndex)
+        : AudioFormatWriter (out, TRANS (flacFormatName), rate, numChans, bits)
+    {
+        using namespace FlacNamespace;
+        encoder = FLAC__stream_encoder_new();
+
+        if (qualityOptionIndex > 0)
+            FLAC__stream_encoder_set_compression_level (encoder, (uint32) jmin (8, qualityOptionIndex));
+
+        FLAC__stream_encoder_set_do_mid_side_stereo (encoder, numChannels == 2);
+        FLAC__stream_encoder_set_loose_mid_side_stereo (encoder, numChannels == 2);
+        FLAC__stream_encoder_set_channels (encoder, numChannels);
+        FLAC__stream_encoder_set_bits_per_sample (encoder, jmin ((unsigned int) 24, bitsPerSample));
+        FLAC__stream_encoder_set_sample_rate (encoder, (unsigned int) sampleRate);
+        FLAC__stream_encoder_set_blocksize (encoder, 0);
+        FLAC__stream_encoder_set_do_escape_coding (encoder, true);
+
+        ok = FLAC__stream_encoder_init_stream (encoder,
+                                               encodeWriteCallback, encodeSeekCallback,
+                                               encodeTellCallback, encodeMetadataCallback,
+                                               this) == FLAC__STREAM_ENCODER_INIT_STATUS_OK;
+    }
+
+    ~FlacWriter()
+    {
+        if (ok)
+        {
+            FlacNamespace::FLAC__stream_encoder_finish (encoder);
+            output->flush();
+        }
+        else
+        {
+            output = nullptr; // to stop the base class deleting this, as it needs to be returned
+                              // to the caller of createWriter()
+        }
+
+        FlacNamespace::FLAC__stream_encoder_delete (encoder);
+    }
+
+    //==============================================================================
+    bool write (const int** samplesToWrite, int numSamples) override
+    {
+        using namespace FlacNamespace;
+        if (! ok)
+            return false;
+
+        HeapBlock<int*> channels;
+        HeapBlock<int> temp;
+        const int bitsToShift = 32 - (int) bitsPerSample;
+
+        if (bitsToShift > 0)
+        {
+            temp.malloc (numChannels * (size_t) numSamples);
+            channels.calloc (numChannels + 1);
+
+            for (unsigned int i = 0; i < numChannels; ++i)
+            {
+                if (samplesToWrite[i] == nullptr)
+                    break;
+
+                int* const destData = temp.getData() + i * (size_t) numSamples;
+                channels[i] = destData;
+
+                for (int j = 0; j < numSamples; ++j)
+                    destData[j] = (samplesToWrite[i][j] >> bitsToShift);
+            }
+
+            samplesToWrite = const_cast<const int**> (channels.getData());
+        }
+
+        return FLAC__stream_encoder_process (encoder, (const FLAC__int32**) samplesToWrite, (size_t) numSamples) != 0;
+    }
+
+    bool writeData (const void* const data, const int size) const
+    {
+        return output->write (data, (size_t) size);
+    }
+
+    static void packUint32 (FlacNamespace::FLAC__uint32 val, FlacNamespace::FLAC__byte* b, const int bytes)
+    {
+        b += bytes;
+
+        for (int i = 0; i < bytes; ++i)
+        {
+            *(--b) = (FlacNamespace::FLAC__byte) (val & 0xff);
+            val >>= 8;
+        }
+    }
+
+    void writeMetaData (const FlacNamespace::FLAC__StreamMetadata* metadata)
+    {
+        using namespace FlacNamespace;
+        const FLAC__StreamMetadata_StreamInfo& info = metadata->data.stream_info;
+
+        unsigned char buffer [FLAC__STREAM_METADATA_STREAMINFO_LENGTH];
+        const unsigned int channelsMinus1 = info.channels - 1;
+        const unsigned int bitsMinus1 = info.bits_per_sample - 1;
+
+        packUint32 (info.min_blocksize, buffer, 2);
+        packUint32 (info.max_blocksize, buffer + 2, 2);
+        packUint32 (info.min_framesize, buffer + 4, 3);
+        packUint32 (info.max_framesize, buffer + 7, 3);
+        buffer[10] = (uint8) ((info.sample_rate >> 12) & 0xff);
+        buffer[11] = (uint8) ((info.sample_rate >> 4) & 0xff);
+        buffer[12] = (uint8) (((info.sample_rate & 0x0f) << 4) | (channelsMinus1 << 1) | (bitsMinus1 >> 4));
+        buffer[13] = (FLAC__byte) (((bitsMinus1 & 0x0f) << 4) | (unsigned int) ((info.total_samples >> 32) & 0x0f));
+        packUint32 ((FLAC__uint32) info.total_samples, buffer + 14, 4);
+        memcpy (buffer + 18, info.md5sum, 16);
+
+        const bool seekOk = output->setPosition (4);
+        (void) seekOk;
+
+        // if this fails, you've given it an output stream that can't seek! It needs
+        // to be able to seek back to write the header
+        jassert (seekOk);
+
+        output->writeIntBigEndian (FLAC__STREAM_METADATA_STREAMINFO_LENGTH);
+        output->write (buffer, FLAC__STREAM_METADATA_STREAMINFO_LENGTH);
+    }
+
+    //==============================================================================
+    static FlacNamespace::FLAC__StreamEncoderWriteStatus encodeWriteCallback (const FlacNamespace::FLAC__StreamEncoder*,
+                                                                              const FlacNamespace::FLAC__byte buffer[],
+                                                                              size_t bytes,
+                                                                              unsigned int /*samples*/,
+                                                                              unsigned int /*current_frame*/,
+                                                                              void* client_data)
+    {
+        using namespace FlacNamespace;
+        return static_cast<FlacWriter*> (client_data)->writeData (buffer, (int) bytes)
+                ? FLAC__STREAM_ENCODER_WRITE_STATUS_OK
+                : FLAC__STREAM_ENCODER_WRITE_STATUS_FATAL_ERROR;
+    }
+
+    static FlacNamespace::FLAC__StreamEncoderSeekStatus encodeSeekCallback (const FlacNamespace::FLAC__StreamEncoder*, FlacNamespace::FLAC__uint64, void*)
+    {
+        using namespace FlacNamespace;
+        return FLAC__STREAM_ENCODER_SEEK_STATUS_UNSUPPORTED;
+    }
+
+    static FlacNamespace::FLAC__StreamEncoderTellStatus encodeTellCallback (const FlacNamespace::FLAC__StreamEncoder*, FlacNamespace::FLAC__uint64* absolute_byte_offset, void* client_data)
+    {
+        using namespace FlacNamespace;
+        if (client_data == nullptr)
+            return FLAC__STREAM_ENCODER_TELL_STATUS_UNSUPPORTED;
+
+        *absolute_byte_offset = (FLAC__uint64) static_cast<FlacWriter*> (client_data)->output->getPosition();
+        return FLAC__STREAM_ENCODER_TELL_STATUS_OK;
+    }
+
+    static void encodeMetadataCallback (const FlacNamespace::FLAC__StreamEncoder*, const FlacNamespace::FLAC__StreamMetadata* metadata, void* client_data)
+    {
+        static_cast<FlacWriter*> (client_data)->writeMetaData (metadata);
+    }
+
+    bool ok;
+
+private:
+    FlacNamespace::FLAC__StreamEncoder* encoder;
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (FlacWriter)
+};
+
+
+//==============================================================================
+FlacAudioFormat::FlacAudioFormat()
+    : AudioFormat (TRANS (flacFormatName), StringArray (flacExtensions))
+{
+}
+
+FlacAudioFormat::~FlacAudioFormat()
+{
+}
+
+Array<int> FlacAudioFormat::getPossibleSampleRates()
+{
+    const int rates[] = { 8000, 11025, 12000, 16000, 22050, 32000, 44100, 48000,
+                          88200, 96000, 176400, 192000, 352800, 384000 };
+
+    return Array<int> (rates, numElementsInArray (rates));
+}
+
+Array<int> FlacAudioFormat::getPossibleBitDepths()
+{
+    const int depths[] = { 16, 24 };
+
+    return Array<int> (depths, numElementsInArray (depths));
+}
+
+bool FlacAudioFormat::canDoStereo()     { return true; }
+bool FlacAudioFormat::canDoMono()       { return true; }
+bool FlacAudioFormat::isCompressed()    { return true; }
+
+AudioFormatReader* FlacAudioFormat::createReaderFor (InputStream* in, const bool deleteStreamIfOpeningFails)
+{
+    ScopedPointer<FlacReader> r (new FlacReader (in));
+
+    if (r->sampleRate > 0)
+        return r.release();
+
+    if (! deleteStreamIfOpeningFails)
+        r->input = nullptr;
+
+    return nullptr;
+}
+
+AudioFormatWriter* FlacAudioFormat::createWriterFor (OutputStream* out,
+                                                     double sampleRate,
+                                                     unsigned int numberOfChannels,
+                                                     int bitsPerSample,
+                                                     const StringPairArray& /*metadataValues*/,
+                                                     int qualityOptionIndex)
+{
+    if (getPossibleBitDepths().contains (bitsPerSample))
+    {
+        ScopedPointer<FlacWriter> w (new FlacWriter (out, sampleRate, numberOfChannels,
+                                                     (uint32) bitsPerSample, qualityOptionIndex));
+        if (w->ok)
+            return w.release();
+    }
+
+    return nullptr;
+}
+
+StringArray FlacAudioFormat::getQualityOptions()
+{
+    const char* options[] = { "0 (Fastest)", "1", "2", "3", "4", "5 (Default)","6", "7", "8 (Highest quality)", 0 };
+    return StringArray (options);
+}
+
+#endif

source/modules/juce_audio_formats/codecs/juce_FlacAudioFormat.h

@@ -0,0 +1,65 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#if JUCE_USE_FLAC || defined (DOXYGEN)
+
+//==============================================================================
+/**
+    Reads and writes the lossless-compression FLAC audio format.
+
+    To compile this, you'll need to set the JUCE_USE_FLAC flag.
+
+    @see AudioFormat
+*/
+class JUCE_API  FlacAudioFormat    : public AudioFormat
+{
+public:
+    //==============================================================================
+    FlacAudioFormat();
+    ~FlacAudioFormat();
+
+    //==============================================================================
+    Array<int> getPossibleSampleRates() override;
+    Array<int> getPossibleBitDepths() override;
+    bool canDoStereo() override;
+    bool canDoMono() override;
+    bool isCompressed() override;
+    StringArray getQualityOptions() override;
+
+    //==============================================================================
+    AudioFormatReader* createReaderFor (InputStream* sourceStream,
+                                        bool deleteStreamIfOpeningFails) override;
+
+    AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
+                                        double sampleRateToUse,
+                                        unsigned int numberOfChannels,
+                                        int bitsPerSample,
+                                        const StringPairArray& metadataValues,
+                                        int qualityOptionIndex) override;
+private:
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (FlacAudioFormat)
+};
+
+
+#endif

source/modules/juce_audio_formats/codecs/juce_LAMEEncoderAudioFormat.cpp

@@ -0,0 +1,227 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#if JUCE_USE_LAME_AUDIO_FORMAT
+
+class LAMEEncoderAudioFormat::Writer   : public AudioFormatWriter
+{
+public:
+    Writer (OutputStream* destStream, const String& formatName,
+            const File& lameApp, int vbr, int cbr,
+            double sampleRate, unsigned int numberOfChannels,
+            unsigned int bitsPerSample, const StringPairArray& metadata)
+        : AudioFormatWriter (destStream, formatName, sampleRate,
+                             numberOfChannels, bitsPerSample),
+          vbrLevel (vbr), cbrBitrate (cbr),
+          tempWav (".wav")
+    {
+        WavAudioFormat wavFormat;
+
+        if (FileOutputStream* out = tempWav.getFile().createOutputStream())
+        {
+            writer = wavFormat.createWriterFor (out, sampleRate, numChannels,
+                                                bitsPerSample, metadata, 0);
+
+            args.add (lameApp.getFullPathName());
+
+            args.add ("--quiet");
+
+            if (cbrBitrate == 0)
+            {
+                args.add ("--vbr-new");
+                args.add ("-V");
+                args.add (String (vbrLevel));
+            }
+            else
+            {
+                args.add ("--cbr");
+                args.add ("-b");
+                args.add (String (cbrBitrate));
+            }
+
+            addMetadataArg (metadata, "id3title",       "--tt");
+            addMetadataArg (metadata, "id3artist",      "--ta");
+            addMetadataArg (metadata, "id3album",       "--tl");
+            addMetadataArg (metadata, "id3comment",     "--tc");
+            addMetadataArg (metadata, "id3date",        "--ty");
+            addMetadataArg (metadata, "id3genre",       "--tg");
+            addMetadataArg (metadata, "id3trackNumber", "--tn");
+        }
+    }
+
+    void addMetadataArg (const StringPairArray& metadata, const char* key, const char* lameFlag)
+    {
+        const String value (metadata.getValue (key, String::empty));
+
+        if (value.isNotEmpty())
+        {
+            args.add (lameFlag);
+            args.add (value);
+        }
+    }
+
+    ~Writer()
+    {
+        if (writer != nullptr)
+        {
+            writer = nullptr;
+
+            if (! convertToMP3())
+                convertToMP3(); // try again
+        }
+    }
+
+    bool write (const int** samplesToWrite, int numSamples)
+    {
+        return writer != nullptr && writer->write (samplesToWrite, numSamples);
+    }
+
+private:
+    int vbrLevel, cbrBitrate;
+    TemporaryFile tempWav;
+    ScopedPointer<AudioFormatWriter> writer;
+    StringArray args;
+
+    bool runLameChildProcess (const TemporaryFile& tempMP3, const StringArray& args) const
+    {
+        ChildProcess cp;
+
+        if (cp.start (args))
+        {
+            const String childOutput (cp.readAllProcessOutput());
+            DBG (childOutput); (void) childOutput;
+
+            cp.waitForProcessToFinish (10000);
+            return tempMP3.getFile().getSize() > 0;
+        }
+
+        return false;
+    }
+
+    bool convertToMP3() const
+    {
+        TemporaryFile tempMP3 (".mp3");
+
+        StringArray args2 (args);
+        args2.add (tempWav.getFile().getFullPathName());
+        args2.add (tempMP3.getFile().getFullPathName());
+
+        DBG (args2.joinIntoString (" "));
+
+        if (runLameChildProcess (tempMP3, args2))
+        {
+            FileInputStream fis (tempMP3.getFile());
+
+            if (fis.openedOk() && output->writeFromInputStream (fis, -1) > 0)
+            {
+                output->flush();
+                return true;
+            }
+        }
+
+        return false;
+    }
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (Writer)
+};
+
+//==============================================================================
+static const char* const lameFormatName = "MP3 file";
+static const char* const lameExtensions[] = { ".mp3", nullptr };
+
+LAMEEncoderAudioFormat::LAMEEncoderAudioFormat (const File& lameApplication)
+   : AudioFormat (TRANS (lameFormatName), StringArray (lameExtensions)),
+     lameApp (lameApplication)
+{
+}
+
+LAMEEncoderAudioFormat::~LAMEEncoderAudioFormat()
+{
+}
+
+bool LAMEEncoderAudioFormat::canHandleFile (const File&)
+{
+    return false;
+}
+
+Array<int> LAMEEncoderAudioFormat::getPossibleSampleRates()
+{
+    const int rates[] = { 32000, 44100, 48000, 0 };
+    return Array <int> (rates);
+}
+
+Array<int> LAMEEncoderAudioFormat::getPossibleBitDepths()
+{
+    const int depths[] = { 16, 0 };
+    return Array <int> (depths);
+}
+
+bool LAMEEncoderAudioFormat::canDoStereo()      { return true; }
+bool LAMEEncoderAudioFormat::canDoMono()        { return true; }
+bool LAMEEncoderAudioFormat::isCompressed()     { return true; }
+
+StringArray LAMEEncoderAudioFormat::getQualityOptions()
+{
+    const char* vbrOptions[] = { "VBR quality 0 (best)", "VBR quality 1", "VBR quality 2", "VBR quality 3",
+                                 "VBR quality 4 (normal)", "VBR quality 5", "VBR quality 6", "VBR quality 7", "VBR quality 8",
+                                 "VBR quality 9 (smallest)",
+                                 nullptr };
+
+    StringArray opts (vbrOptions);
+
+    const int cbrRates[] = { 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 };
+
+    for (int i = 0; i < numElementsInArray (cbrRates); ++i)
+        opts.add (String (cbrRates[i]) + " Kb/s CBR");
+
+    return opts;
+}
+
+AudioFormatReader* LAMEEncoderAudioFormat::createReaderFor (InputStream*, const bool)
+{
+    return nullptr;
+}
+
+AudioFormatWriter* LAMEEncoderAudioFormat::createWriterFor (OutputStream* streamToWriteTo,
+                                                            double sampleRateToUse,
+                                                            unsigned int numberOfChannels,
+                                                            int bitsPerSample,
+                                                            const StringPairArray& metadataValues,
+                                                            int qualityOptionIndex)
+{
+    int vbr = 4;
+    int cbr = 0;
+
+    const String qual (getQualityOptions() [qualityOptionIndex]);
+
+    if (qual.contains ("VBR"))
+        vbr = qual.retainCharacters ("0123456789").getIntValue();
+    else
+        cbr = qual.getIntValue();
+
+    return new Writer (streamToWriteTo, getFormatName(), lameApp, vbr, cbr,
+                       sampleRateToUse, numberOfChannels, bitsPerSample, metadataValues);
+}
+
+#endif

source/modules/juce_audio_formats/codecs/juce_LAMEEncoderAudioFormat.h

@@ -0,0 +1,71 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2013 - Raw Material Software Ltd.
+
+   Permission is granted to use this software under the terms of either:
+   a) the GPL v2 (or any later version)
+   b) the Affero GPL v3
+
+   Details of these licenses can be found at: www.gnu.org/licenses
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+   ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.juce.com for more information.
+
+  ==============================================================================
+*/
+
+#if JUCE_USE_LAME_AUDIO_FORMAT || defined (DOXYGEN)
+
+//==============================================================================
+/**
+    An AudioFormat class which can use an installed version of the LAME mp3
+    encoder to encode a file.
+
+    This format can't read MP3s, it just writes them. Internally, the
+    AudioFormatWriter object that is returned writes the incoming audio data
+    to a temporary WAV file, and then when the writer is deleted, it invokes
+    the LAME executable to convert the data to an MP3, whose data is then
+    piped into the original OutputStream that was used when first creating
+    the writer.
+
+    @see AudioFormat
+*/
+class JUCE_API  LAMEEncoderAudioFormat    : public AudioFormat
+{
+public:
+    /** Creates a LAMEEncoderAudioFormat that expects to find a working LAME
+        executable at the location given.
+    */
+    LAMEEncoderAudioFormat (const File& lameExecutableToUse);
+    ~LAMEEncoderAudioFormat();
+
+    bool canHandleFile (const File&);
+    Array<int> getPossibleSampleRates();
+    Array<int> getPossibleBitDepths();
+    bool canDoStereo();
+    bool canDoMono();
+    bool isCompressed();
+    StringArray getQualityOptions();
+
+    AudioFormatReader* createReaderFor (InputStream*, bool deleteStreamIfOpeningFails);
+
+    AudioFormatWriter* createWriterFor (OutputStream*, double sampleRateToUse,
+                                        unsigned int numberOfChannels, int bitsPerSample,
+                                        const StringPairArray& metadataValues, int qualityOptionIndex);
+
+private:
+    File lameApp;
+    class Writer;
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (LAMEEncoderAudioFormat)
+};
+
+#endif