JUCE/build/linux/platform_specific_code/juce_linux_Audio.cpp

/*
  ==============================================================================

   This file is part of the JUCE library - "Jules' Utility Class Extensions"
   Copyright 2004-7 by Raw Material Software ltd.

  ------------------------------------------------------------------------------

   JUCE can be redistributed and/or modified under the terms of the
   GNU General Public License, as published by the Free Software Foundation;
   either version 2 of the License, or (at your option) any later version.

   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.

   You should have received a copy of the GNU General Public License
   along with JUCE; if not, visit www.gnu.org/licenses or write to the
   Free Software Foundation, Inc., 59 Temple Place, Suite 330,
   Boston, MA 02111-1307 USA

  ------------------------------------------------------------------------------

   If you'd like to release a closed-source product which uses JUCE, commercial
   licenses are also available: visit www.rawmaterialsoftware.com/juce for
   more information.

  ==============================================================================
*/

#include "../../../juce_Config.h"
#if JUCE_BUILD_GUI_CLASSES

#if JUCE_ALSA

#include "linuxincludes.h"

//==============================================================================
/* 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 disable the JUCE_ALSA flag in juce_Config.h
*/
#include <alsa/asoundlib.h>


//==============================================================================
#include "../../../src/juce_core/basics/juce_StandardHeader.h"

BEGIN_JUCE_NAMESPACE

#include "../../../src/juce_appframework/audio/devices/juce_AudioIODeviceType.h"
#include "../../../src/juce_core/threads/juce_Thread.h"
#include "../../../src/juce_core/threads/juce_ScopedLock.h"
#include "../../../src/juce_core/basics/juce_Time.h"
#include "../../../src/juce_core/io/files/juce_File.h"
#include "../../../src/juce_core/io/files/juce_FileInputStream.h"
#include "../../../src/juce_core/basics/juce_Singleton.h"
#include "../../../src/juce_appframework/audio/dsp/juce_AudioDataConverters.h"

static const int maxNumChans = 64;


//==============================================================================
static void getDeviceSampleRates (snd_pcm_t* handle, Array <int>& rates)
{
    const int ratesToTry[] = { 22050, 32000, 44100, 48000, 88200, 96000, 176400, 192000, 0 };

    snd_pcm_hw_params_t* hwParams;
    snd_pcm_hw_params_alloca (&hwParams);

    for (int i = 0; ratesToTry[i] != 0; ++i)
    {
        if (snd_pcm_hw_params_any (handle, hwParams) >= 0
             && snd_pcm_hw_params_test_rate (handle, hwParams, ratesToTry[i], 0) == 0)
        {
            rates.add (ratesToTry[i]);
        }
    }
}

static void getDeviceNumChannels (snd_pcm_t* handle, unsigned int* minChans, unsigned int* maxChans)
{
    snd_pcm_hw_params_t *params;
    snd_pcm_hw_params_alloca (&params);

    if (snd_pcm_hw_params_any (handle, params) >= 0)
    {
        snd_pcm_hw_params_get_channels_min (params, minChans);
        snd_pcm_hw_params_get_channels_max (params, maxChans);
    }
}

static void getDeviceProperties (const String& id,
                                 unsigned int& minChansOut,
                                 unsigned int& maxChansOut,
                                 unsigned int& minChansIn,
                                 unsigned int& maxChansIn,
                                 Array <int>& rates)
{
    snd_ctl_t* handle;

    if (snd_ctl_open (&handle, id.upToLastOccurrenceOf (T(","), false, false), SND_CTL_NONBLOCK) >= 0)
    {
        snd_pcm_info_t* info;
        snd_pcm_info_alloca (&info);

        snd_pcm_info_set_stream (info, SND_PCM_STREAM_PLAYBACK);
        snd_pcm_info_set_device (info, id.fromLastOccurrenceOf (T(","), false, false).getIntValue());
        snd_pcm_info_set_subdevice (info, 0);

        if (snd_ctl_pcm_info (handle, info) >= 0)
        {
            snd_pcm_t* pcmHandle;
            if (snd_pcm_open (&pcmHandle, id, SND_PCM_STREAM_PLAYBACK, SND_PCM_ASYNC | SND_PCM_NONBLOCK ) >= 0)
            {
                getDeviceNumChannels (pcmHandle, &minChansOut, &maxChansOut);
                getDeviceSampleRates (pcmHandle, rates);

                snd_pcm_close (pcmHandle);
            }
        }

        snd_pcm_info_set_stream (info, SND_PCM_STREAM_CAPTURE);

        if (snd_ctl_pcm_info (handle, info) >= 0)
        {
            snd_pcm_t* pcmHandle;
            if (snd_pcm_open (&pcmHandle, id, SND_PCM_STREAM_CAPTURE, SND_PCM_ASYNC | SND_PCM_NONBLOCK ) >= 0)
            {
                getDeviceNumChannels (pcmHandle, &minChansIn, &maxChansIn);

                if (rates.size() == 0)
                    getDeviceSampleRates (pcmHandle, rates);

                snd_pcm_close (pcmHandle);
            }
        }

        snd_ctl_close (handle);
    }
}

//==============================================================================
class ALSADevice
{
public:
    ALSADevice (const String& deviceName,
                const bool forInput)
        : handle (0),
          bitDepth (16),
          numChannelsRunning (0),
          isInput (forInput),
          sampleFormat (AudioDataConverters::int16LE)
    {
        failed (snd_pcm_open (&handle, deviceName,
                              forInput ? SND_PCM_STREAM_CAPTURE : SND_PCM_STREAM_PLAYBACK,
                              SND_PCM_ASYNC));
    }

    ~ALSADevice()
    {
        if (handle != 0)
            snd_pcm_close (handle);
    }

    bool setParameters (unsigned int sampleRate, int numChannels, int bufferSize)
    {
        if (handle == 0)
            return false;

        snd_pcm_hw_params_t* hwParams;
        snd_pcm_hw_params_alloca (&hwParams);

        if (failed (snd_pcm_hw_params_any (handle, hwParams)))
            return false;

        if (snd_pcm_hw_params_set_access (handle, hwParams, SND_PCM_ACCESS_RW_NONINTERLEAVED) >= 0)
            isInterleaved = false;
        else if (snd_pcm_hw_params_set_access (handle, hwParams, SND_PCM_ACCESS_RW_INTERLEAVED) >= 0)
            isInterleaved = true;
        else
        {
            jassertfalse
            return false;
        }

        const int formatsToTry[] = { SND_PCM_FORMAT_FLOAT_LE, 32, AudioDataConverters::float32LE,
                                     SND_PCM_FORMAT_FLOAT_BE, 32, AudioDataConverters::float32BE,
                                     SND_PCM_FORMAT_S32_LE, 32, AudioDataConverters::int32LE,
                                     SND_PCM_FORMAT_S32_BE, 32, AudioDataConverters::int32BE,
                                     SND_PCM_FORMAT_S24_LE, 24, AudioDataConverters::int24LE,
                                     SND_PCM_FORMAT_S24_BE, 24, AudioDataConverters::int24BE,
                                     SND_PCM_FORMAT_S16_LE, 16, AudioDataConverters::int16LE,
                                     SND_PCM_FORMAT_S16_BE, 16, AudioDataConverters::int16BE };
        bitDepth = 0;

        for (int i = 0; i < numElementsInArray (formatsToTry); i += 3)
        {
            if (snd_pcm_hw_params_set_format (handle, hwParams, (_snd_pcm_format) formatsToTry [i]) >= 0)
            {
                bitDepth = formatsToTry [i + 1];
                sampleFormat = (AudioDataConverters::DataFormat) formatsToTry [i + 2];
                break;
            }
        }

        if (bitDepth == 0)
        {
            error = "device doesn't support a compatible PCM format";
            DBG (T("ALSA error: ") + error + T("\n"));
            return false;
        }

        int dir = 0;
        unsigned int periods = 4;
        snd_pcm_uframes_t samplesPerPeriod = bufferSize;

        if (failed (snd_pcm_hw_params_set_rate_near (handle, hwParams, &sampleRate, 0))
            || failed (snd_pcm_hw_params_set_channels (handle, hwParams, numChannels))
            || failed (snd_pcm_hw_params_set_periods_near (handle, hwParams, &periods, &dir))
            || failed (snd_pcm_hw_params_set_period_size_near (handle, hwParams, &samplesPerPeriod, &dir))
            || failed (snd_pcm_hw_params (handle, hwParams)))
        {
            return false;
        }

        snd_pcm_sw_params_t* swParams;
        snd_pcm_sw_params_alloca (&swParams);

        if (failed (snd_pcm_sw_params_current (handle, swParams))
            || failed (snd_pcm_sw_params_set_silence_threshold (handle, swParams, 0))
            || failed (snd_pcm_sw_params_set_silence_size (handle, swParams, INT_MAX))
            || failed (snd_pcm_sw_params_set_start_threshold (handle, swParams, samplesPerPeriod))
            || failed (snd_pcm_sw_params_set_stop_threshold (handle, swParams, INT_MAX))
            || failed (snd_pcm_sw_params (handle, swParams)))
        {
            return false;
        }

        /*
#ifdef JUCE_DEBUG
        // enable this to dump the config of the devices that get opened
        snd_output_t* out;
        snd_output_stdio_attach (&out, stderr, 0);
        snd_pcm_hw_params_dump (hwParams, out);
        snd_pcm_sw_params_dump (swParams, out);
#endif
        */

        numChannelsRunning = numChannels;

        return true;
    }

    //==============================================================================
    bool write (float** const data, const int numSamples)
    {
        if (isInterleaved)
        {
            scratch.ensureSize (sizeof (float) * numSamples * numChannelsRunning, false);
            float* interleaved = (float*) scratch;

            AudioDataConverters::interleaveSamples ((const float**) data, interleaved, numSamples, numChannelsRunning);
            AudioDataConverters::convertFloatToFormat (sampleFormat, interleaved, interleaved, numSamples * numChannelsRunning);

            snd_pcm_sframes_t num = snd_pcm_writei (handle, (void*) interleaved, numSamples);

            if (failed (num) && num != -EPIPE && num != -ESTRPIPE)
                return false;
        }
        else
        {
            for (int i = 0; i < numChannelsRunning; ++i)
                if (data[i] != 0)
                    AudioDataConverters::convertFloatToFormat (sampleFormat, data[i], data[i], numSamples);

            snd_pcm_sframes_t num = snd_pcm_writen (handle, (void**) data, numSamples);

            if (failed (num))
            {
                if (num == -EPIPE)
                {
                    if (failed (snd_pcm_prepare (handle)))
                        return false;
                }
                else if (num != -ESTRPIPE)
                    return false;
            }
        }

        return true;
    }

    bool read (float** const data, const int numSamples)
    {
        if (isInterleaved)
        {
            scratch.ensureSize (sizeof (float) * numSamples * numChannelsRunning, false);
            float* interleaved = (float*) scratch;

            snd_pcm_sframes_t num = snd_pcm_readi (handle, (void*) interleaved, numSamples);

            if (failed (num))
            {
                if (num == -EPIPE)
                {
                    if (failed (snd_pcm_prepare (handle)))
                        return false;
                }
                else if (num != -ESTRPIPE)
                    return false;
            }

            AudioDataConverters::convertFormatToFloat (sampleFormat, interleaved, interleaved, numSamples * numChannelsRunning);
            AudioDataConverters::deinterleaveSamples (interleaved, data, numSamples, numChannelsRunning);
        }
        else
        {
            snd_pcm_sframes_t num = snd_pcm_readn (handle, (void**) data, numSamples);

            if (failed (num) && num != -EPIPE && num != -ESTRPIPE)
                return false;

            for (int i = 0; i < numChannelsRunning; ++i)
                if (data[i] != 0)
                    AudioDataConverters::convertFormatToFloat (sampleFormat, data[i], data[i], numSamples);
        }

        return true;
    }

    //==============================================================================
    juce_UseDebuggingNewOperator

    snd_pcm_t* handle;
    String error;
    int bitDepth, numChannelsRunning;

    //==============================================================================
private:
    const bool isInput;
    bool isInterleaved;
    MemoryBlock scratch;
    AudioDataConverters::DataFormat sampleFormat;

    //==============================================================================
    bool failed (const int errorNum)
    {
        if (errorNum >= 0)
            return false;

        error = snd_strerror (errorNum);
        DBG (T("ALSA error: ") + error + T("\n"));
        return true;
    }
};

//==============================================================================
class ALSAThread  : public Thread
{
public:
    ALSAThread (const String& deviceName_)
        : Thread ("Juce ALSA"),
          sampleRate (0),
          bufferSize (0),
          callback (0),
          deviceName (deviceName_),
          outputDevice (0),
          inputDevice (0),
          numCallbacks (0),
          totalNumInputChannels (0),
          totalNumOutputChannels (0)
    {
        zeromem (outputChannelData, sizeof (outputChannelData));
        zeromem (outputChannelDataForCallback, sizeof (outputChannelDataForCallback));
        zeromem (inputChannelData, sizeof (inputChannelData));
        zeromem (inputChannelDataForCallback, sizeof (inputChannelDataForCallback));

        initialiseRatesAndChannels();
    }

    ~ALSAThread()
    {
        close();
    }

    void open (const BitArray& inputChannels,
               const BitArray& outputChannels,
               const double sampleRate_,
               const int bufferSize_)
    {
        close();

        error = String::empty;
        sampleRate = sampleRate_;
        bufferSize = bufferSize_;
        currentInputChans.clear();
        currentOutputChans.clear();

        numChannelsRunning = jmax (inputChannels.getHighestBit(),
                                   outputChannels.getHighestBit()) + 1;

        numChannelsRunning = jmin (maxNumChans, jlimit ((int) minChansIn,
                                                        (int) maxChansIn,
                                                        numChannelsRunning));

        if (inputChannels.getHighestBit() >= 0)
        {
            for (int i = 0; i < numChannelsRunning; ++i)
            {
                inputChannelData [i] = (float*) juce_calloc (sizeof (float) * bufferSize);

                if (inputChannels[i])
                {
                    inputChannelDataForCallback [totalNumInputChannels++] = inputChannelData [i];
                    currentInputChans.setBit (i);
                }
            }
        }

        if (outputChannels.getHighestBit() >= 0)
        {
            for (int i = 0; i < numChannelsRunning; ++i)
            {
                outputChannelData [i] = (float*) juce_calloc (sizeof (float) * bufferSize);

                if (outputChannels[i])
                {
                    outputChannelDataForCallback [totalNumOutputChannels++] = outputChannelData [i];
                    currentOutputChans.setBit (i);
                }
            }
        }

        if (totalNumOutputChannels > 0)
        {
            outputDevice = new ALSADevice (deviceName, false);

            if (outputDevice->error.isNotEmpty())
            {
                error = outputDevice->error;
                deleteAndZero (outputDevice);
                return;
            }

            if (! outputDevice->setParameters ((unsigned int) sampleRate, numChannelsRunning, bufferSize))
            {
                error = outputDevice->error;
                deleteAndZero (outputDevice);
                return;
            }
        }

        if (totalNumInputChannels > 0)
        {
            inputDevice = new ALSADevice (deviceName, true);

            if (inputDevice->error.isNotEmpty())
            {
                error = inputDevice->error;
                deleteAndZero (inputDevice);
                return;
            }

            if (! inputDevice->setParameters ((unsigned int) sampleRate, numChannelsRunning, bufferSize))
            {
                error = inputDevice->error;
                deleteAndZero (inputDevice);
                return;
            }
        }

        if (outputDevice == 0 && inputDevice == 0)
        {
            error = "no channels";
            return;
        }

        if (outputDevice != 0 && inputDevice != 0)
        {
            snd_pcm_link (outputDevice->handle, inputDevice->handle);
        }

        if (inputDevice != 0 && failed (snd_pcm_prepare (inputDevice->handle)))
            return;

        if (outputDevice != 0 && failed (snd_pcm_prepare (outputDevice->handle)))
            return;

        startThread (9);

        int count = 1000;

        while (numCallbacks == 0)
        {
            sleep (5);

            if (--count < 0 || ! isThreadRunning())
            {
                error = "device didn't start";
                break;
            }
        }
    }

    void close()
    {
        stopThread (6000);

        deleteAndZero (inputDevice);
        deleteAndZero (outputDevice);

        for (int i = 0; i < maxNumChans; ++i)
        {
            juce_free (inputChannelData [i]);
            juce_free (outputChannelData [i]);
        }

        zeromem (outputChannelData, sizeof (outputChannelData));
        zeromem (outputChannelDataForCallback, sizeof (outputChannelDataForCallback));
        zeromem (inputChannelData, sizeof (inputChannelData));
        zeromem (inputChannelDataForCallback, sizeof (inputChannelDataForCallback));
        totalNumOutputChannels = 0;
        totalNumInputChannels = 0;
        numChannelsRunning = 0;

        numCallbacks = 0;
    }

    void setCallback (AudioIODeviceCallback* const newCallback) throw()
    {
        const ScopedLock sl (callbackLock);
        callback = newCallback;
    }

    void run()
    {
        while (! threadShouldExit())
        {
            if (inputDevice != 0)
            {
                jassert (numChannelsRunning >= inputDevice->numChannelsRunning);

                if (! inputDevice->read (inputChannelData, bufferSize))
                {
                    DBG ("ALSA: read failure");
                    break;
                }
            }

            if (threadShouldExit())
                break;

            {
                const ScopedLock sl (callbackLock);
                ++numCallbacks;

                if (callback != 0)
                {
                    callback->audioDeviceIOCallback ((const float**) inputChannelDataForCallback,
                                                     totalNumInputChannels,
                                                     outputChannelDataForCallback,
                                                     totalNumOutputChannels,
                                                     bufferSize);
                }
                else
                {
                    for (int i = 0; i < totalNumOutputChannels; ++i)
                        zeromem (outputChannelDataForCallback[i], sizeof (float) * bufferSize);
                }
            }

            if (outputDevice != 0)
            {
                failed (snd_pcm_wait (outputDevice->handle, 2000));

                if (threadShouldExit())
                    break;

                failed (snd_pcm_avail_update (outputDevice->handle));

                jassert (numChannelsRunning >= outputDevice->numChannelsRunning);
                if (! outputDevice->write (outputChannelData, bufferSize))
                {
                    DBG ("ALSA: write failure");
                    break;
                }
            }
        }
    }

    int getBitDepth() const throw()
    {
        if (outputDevice != 0)
            return outputDevice->bitDepth;

        if (inputDevice != 0)
            return inputDevice->bitDepth;

        return 16;
    }

    //==============================================================================
    juce_UseDebuggingNewOperator

    String error;
    double sampleRate;
    int bufferSize;
    BitArray currentInputChans, currentOutputChans;

    Array <int> sampleRates;
    StringArray channelNamesOut, channelNamesIn;
    AudioIODeviceCallback* callback;

private:
    //==============================================================================
    const String deviceName;
    ALSADevice* outputDevice;
    ALSADevice* inputDevice;
    int numCallbacks;

    CriticalSection callbackLock;

    float* outputChannelData [maxNumChans];
    float* outputChannelDataForCallback [maxNumChans];
    int totalNumInputChannels;
    float* inputChannelData [maxNumChans];
    float* inputChannelDataForCallback [maxNumChans];
    int totalNumOutputChannels;
    int numChannelsRunning;

    unsigned int minChansOut, maxChansOut;
    unsigned int minChansIn, maxChansIn;

    bool failed (const int errorNum) throw()
    {
        if (errorNum >= 0)
            return false;

        error = snd_strerror (errorNum);
        DBG (T("ALSA error: ") + error + T("\n"));
        return true;
    }

    void initialiseRatesAndChannels() throw()
    {
        sampleRates.clear();
        channelNamesOut.clear();
        channelNamesIn.clear();
        minChansOut = 0;
        maxChansOut = 0;
        minChansIn = 0;
        maxChansIn = 0;

        getDeviceProperties (deviceName, minChansOut, maxChansOut, minChansIn, maxChansIn, sampleRates);

        unsigned int i;
        for (i = 0; i < maxChansOut; ++i)
            channelNamesOut.add (T("channel ") + String ((int) i + 1));

        for (i = 0; i < maxChansIn; ++i)
            channelNamesIn.add (T("channel ") + String ((int) i + 1));
    }
};


//==============================================================================
class ALSAAudioIODevice   : public AudioIODevice
{
public:
    ALSAAudioIODevice (const String& deviceName,
                       const String& deviceId)
        : AudioIODevice (deviceName, T("ALSA")),
          isOpen_ (false),
          isStarted (false),
          internal (0)
    {
        internal = new ALSAThread (deviceId);
    }

    ~ALSAAudioIODevice()
    {
        delete internal;
    }

    const StringArray getOutputChannelNames()
    {
        return internal->channelNamesOut;
    }

    const StringArray getInputChannelNames()
    {
        return internal->channelNamesIn;
    }

    int getNumSampleRates()
    {
        return internal->sampleRates.size();
    }

    double getSampleRate (int index)
    {
        return internal->sampleRates [index];
    }

    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;
    }

    int getDefaultBufferSize()
    {
        return 512;
    }

    const String open (const BitArray& inputChannels,
                       const BitArray& outputChannels,
                       double sampleRate,
                       int bufferSizeSamples)
    {
        close();

        if (bufferSizeSamples <= 0)
            bufferSizeSamples = getDefaultBufferSize();

        if (sampleRate <= 0)
        {
            for (int i = 0; i < getNumSampleRates(); ++i)
            {
                if (getSampleRate (i) >= 44100)
                {
                    sampleRate = getSampleRate (i);
                    break;
                }
            }
        }

        internal->open (inputChannels, outputChannels,
                        sampleRate, bufferSizeSamples);

        isOpen_ = internal->error.isEmpty();
        return internal->error;
    }

    void close()
    {
        stop();
        internal->close();
        isOpen_ = false;
    }

    bool isOpen()
    {
        return isOpen_;
    }

    int getCurrentBufferSizeSamples()
    {
        return internal->bufferSize;
    }

    double getCurrentSampleRate()
    {
        return internal->sampleRate;
    }

    int getCurrentBitDepth()
    {
        return internal->getBitDepth();
    }

    const BitArray getActiveOutputChannels() const
    {
        return internal->currentOutputChans;
    }

    const BitArray getActiveInputChannels() const
    {
        return internal->currentInputChans;
    }

    int getOutputLatencyInSamples()
    {
        return 0;
    }

    int getInputLatencyInSamples()
    {
        return 0;
    }

    void start (AudioIODeviceCallback* callback)
    {
        if (! isOpen_)
            callback = 0;

        internal->setCallback (callback);

        if (callback != 0)
            callback->audioDeviceAboutToStart (internal->sampleRate,
                                               internal->bufferSize);

        isStarted = (callback != 0);
    }

    void stop()
    {
        AudioIODeviceCallback* const oldCallback = internal->callback;

        start (0);

        if (oldCallback != 0)
            oldCallback->audioDeviceStopped();
    }

    bool isPlaying()
    {
        return isStarted && internal->error.isEmpty();
    }

    const String getLastError()
    {
        return internal->error;
    }

private:
    bool isOpen_, isStarted;
    ALSAThread* internal;
};


//==============================================================================
class ALSAAudioIODeviceType  : public AudioIODeviceType
{
public:
    //==============================================================================
    ALSAAudioIODeviceType()
        : AudioIODeviceType (T("ALSA")),
          hasScanned (false)
    {
    }

    ~ALSAAudioIODeviceType()
    {
    }

    //==============================================================================
    void scanForDevices()
    {
        hasScanned = true;

        names.clear();
        ids.clear();

        snd_ctl_t* handle;
        snd_ctl_card_info_t* info;
        snd_ctl_card_info_alloca (&info);

        int cardNum = -1;

        while (ids.size() <= 24)
        {
            snd_card_next (&cardNum);

            if (cardNum < 0)
                break;

            if (snd_ctl_open (&handle, T("hw:") + String (cardNum), SND_CTL_NONBLOCK) >= 0)
            {
                if (snd_ctl_card_info (handle, info) >= 0)
                {
                    String cardId (snd_ctl_card_info_get_id (info));

                    if (cardId.removeCharacters (T("0123456789")).isEmpty())
                        cardId = String (cardNum);

                    int device = -1;

                    for (;;)
                    {
                        if (snd_ctl_pcm_next_device (handle, &device) < 0 || device < 0)
                            break;

                        String id, name;
                        id << "hw:" << cardId << ',' << device;

                        if (testDevice (id))
                        {
                            name << snd_ctl_card_info_get_name (info);

                            if (name.isEmpty())
                                name = id;

                            if (device > 0)
                                name << " (" << (device + 1) << ')';

                            ids.add (id);
                            names.add (name);
                        }
                    }
                }

                snd_ctl_close (handle);
            }
        }
    }

    const StringArray getDeviceNames (const bool /*preferInputNames*/) const
    {
        jassert (hasScanned); // need to call scanForDevices() before doing this

        StringArray namesCopy (names);
        namesCopy.removeDuplicates (true);

        return namesCopy;
    }

    const String getDefaultDeviceName (const bool /*preferInputNames*/) const
    {
        jassert (hasScanned); // need to call scanForDevices() before doing this

        return names[0];
    }

    AudioIODevice* createDevice (const String& deviceName)
    {
        jassert (hasScanned); // need to call scanForDevices() before doing this

        const int index = names.indexOf (deviceName);

        if (index >= 0)
            return new ALSAAudioIODevice (deviceName, ids [index]);

        return 0;
    }

    //==============================================================================
    juce_UseDebuggingNewOperator

private:
    StringArray names, ids;
    bool hasScanned;

    static bool testDevice (const String& id)
    {
        unsigned int minChansOut = 0, maxChansOut = 0;
        unsigned int minChansIn = 0, maxChansIn = 0;
        Array <int> rates;

        getDeviceProperties (id, minChansOut, maxChansOut, minChansIn, maxChansIn, rates);

        DBG (T("ALSA device: ") + id
              + T(" outs=") + String ((int) minChansOut) + T("-") + String ((int) maxChansOut)
              + T(" ins=") + String ((int) minChansIn) + T("-") + String ((int) maxChansIn)
              + T(" rates=") + String (rates.size()));

        return (maxChansOut > 0 || maxChansIn > 0) && rates.size() > 0;
    }

    ALSAAudioIODeviceType (const ALSAAudioIODeviceType&);
    const ALSAAudioIODeviceType& operator= (const ALSAAudioIODeviceType&);
};

//==============================================================================
AudioIODeviceType* juce_createDefaultAudioIODeviceType()
{
    return new ALSAAudioIODeviceType();
}


END_JUCE_NAMESPACE


//==============================================================================
#else  // if ALSA is turned off..

#include "../../../src/juce_core/basics/juce_StandardHeader.h"

BEGIN_JUCE_NAMESPACE

#include "../../../src/juce_appframework/audio/devices/juce_AudioIODeviceType.h"
AudioIODeviceType* juce_createDefaultAudioIODeviceType()    { return 0; }

END_JUCE_NAMESPACE

#endif

#endif