Revert "Delete juce_audio_formats"

This reverts commit a7d2d6ddf4.

Signed-off-by: falkTX <falktx@gmail.com>

source/modules/juce_audio_formats/Makefile

@@ -0,0 +1,120 @@
+#!/usr/bin/make -f
+# Makefile for juce_audio_formats #
+# ------------------------------- #
+# Created by falkTX
+#
+
+CWD=../..
+MODULENAME=juce_audio_formats
+include ../Makefile.mk
+
+# ----------------------------------------------------------------------------------------------------------------------------
+
+BUILD_CXX_FLAGS += $(JUCE_AUDIO_FORMATS_FLAGS) -I..
+BUILD_CXX_FLAGS += -DHAVE_LROUND
+
+# ----------------------------------------------------------------------------------------------------------------------------
+
+ifeq ($(MACOS),true)
+OBJS         = $(OBJDIR)/$(MODULENAME).mm.o
+OBJS_posix32 = $(OBJDIR)/$(MODULENAME).mm.posix32.o
+OBJS_posix64 = $(OBJDIR)/$(MODULENAME).mm.posix64.o
+else
+OBJS         = $(OBJDIR)/$(MODULENAME).cpp.o
+OBJS_posix32 = $(OBJDIR)/$(MODULENAME).cpp.posix32.o
+OBJS_posix64 = $(OBJDIR)/$(MODULENAME).cpp.posix64.o
+endif
+OBJS_win32   = $(OBJDIR)/$(MODULENAME).cpp.win32.o
+OBJS_win64   = $(OBJDIR)/$(MODULENAME).cpp.win64.o
+
+# ----------------------------------------------------------------------------------------------------------------------------
+
+all:     $(MODULEDIR)/$(MODULENAME).a
+posix32: $(MODULEDIR)/$(MODULENAME).posix32.a
+posix64: $(MODULEDIR)/$(MODULENAME).posix64.a
+win32:   $(MODULEDIR)/$(MODULENAME).win32.a
+win64:   $(MODULEDIR)/$(MODULENAME).win64.a
+
+# ----------------------------------------------------------------------------------------------------------------------------
+
+clean:
+	rm -f $(OBJDIR)/*.o $(MODULEDIR)/$(MODULENAME)*.a
+
+debug:
+	$(MAKE) DEBUG=true
+
+# ----------------------------------------------------------------------------------------------------------------------------
+
+$(MODULEDIR)/$(MODULENAME).a: $(OBJS)
+	-@mkdir -p $(MODULEDIR)
+	@echo "Creating $(MODULENAME).a"
+	@rm -f $@
+	@$(AR) crs $@ $^
+
+$(MODULEDIR)/$(MODULENAME).posix32.a: $(OBJS_posix32)
+	-@mkdir -p $(MODULEDIR)
+	@echo "Creating $(MODULENAME).posix32.a"
+	@rm -f $@
+	@$(AR) crs $@ $^
+
+$(MODULEDIR)/$(MODULENAME).posix64.a: $(OBJS_posix64)
+	-@mkdir -p $(MODULEDIR)
+	@echo "Creating $(MODULENAME).posix64.a"
+	@rm -f $@
+	@$(AR) crs $@ $^
+
+$(MODULEDIR)/$(MODULENAME).win32.a: $(OBJS_win32)
+	-@mkdir -p $(MODULEDIR)
+	@echo "Creating $(MODULENAME).win32.a"
+	@rm -f $@
+	@$(AR) crs $@ $^
+
+$(MODULEDIR)/$(MODULENAME).win64.a: $(OBJS_win64)
+	-@mkdir -p $(MODULEDIR)
+	@echo "Creating $(MODULENAME).win64.a"
+	@rm -f $@
+	@$(AR) crs $@ $^
+
+# ----------------------------------------------------------------------------------------------------------------------------
+
+$(OBJDIR)/$(MODULENAME).cpp.o: $(MODULENAME).cpp
+	-@mkdir -p $(OBJDIR)
+	@echo "Compiling $<"
+	@$(CXX) $< $(BUILD_CXX_FLAGS) -c -o $@
+
+$(OBJDIR)/$(MODULENAME).cpp.%32.o: $(MODULENAME).cpp
+	-@mkdir -p $(OBJDIR)
+	@echo "Compiling $< (32bit)"
+	@$(CXX) $< $(BUILD_CXX_FLAGS) $(32BIT_FLAGS) -c -o $@
+
+$(OBJDIR)/$(MODULENAME).cpp.%64.o: $(MODULENAME).cpp
+	-@mkdir -p $(OBJDIR)
+	@echo "Compiling $< (64bit)"
+	@$(CXX) $< $(BUILD_CXX_FLAGS) $(64BIT_FLAGS) -c -o $@
+
+# ----------------------------------------------------------------------------------------------------------------------------
+
+$(OBJDIR)/$(MODULENAME).mm.o: $(MODULENAME).cpp
+	-@mkdir -p $(OBJDIR)
+	@echo "Compiling $<"
+	@$(CXX) $< $(BUILD_CXX_FLAGS) -ObjC++ -c -o $@
+
+$(OBJDIR)/$(MODULENAME).mm.%32.o: $(MODULENAME).cpp
+	-@mkdir -p $(OBJDIR)
+	@echo "Compiling $< (32bit)"
+	@$(CXX) $< $(BUILD_CXX_FLAGS) $(32BIT_FLAGS) -ObjC++ -c -o $@
+
+$(OBJDIR)/$(MODULENAME).mm.%64.o: $(MODULENAME).cpp
+	-@mkdir -p $(OBJDIR)
+	@echo "Compiling $< (64bit)"
+	@$(CXX) $< $(BUILD_CXX_FLAGS) $(64BIT_FLAGS) -ObjC++ -c -o $@
+
+# ----------------------------------------------------------------------------------------------------------------------------
+
+-include $(OBJS:%.o=%.d)
+-include $(OBJS_posix32:%.o=%.d)
+-include $(OBJS_posix64:%.o=%.d)
+-include $(OBJS_win32:%.o=%.d)
+-include $(OBJS_win64:%.o=%.d)
+
+# ----------------------------------------------------------------------------------------------------------------------------

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 +1,371 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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"
+
+/** \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/libtool.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) 2007-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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__SHARE__ALLOC_H
+#define FLAC__SHARE__ALLOC_H
+
+#ifdef 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
+ */
+
+// JUCE: removed as JUCE already includes standard headers and including
+// these in FlacNamespace will cause problems
+
+//#include <limits.h> /* for SIZE_MAX */
+//#if HAVE_STDINT_H
+//#include <stdint.h> /* for SIZE_MAX in case limits.h didn't get it */
+//#endif
+//#include <stdlib.h> /* for size_t, malloc(), etc */
+#include "compat.h"
+
+#ifndef SIZE_MAX
+# ifndef SIZE_T_MAX
+#  ifdef _MSC_VER
+#   ifdef _WIN64
+#    define SIZE_T_MAX 0xffffffffffffffffui64
+#   else
+#    define SIZE_T_MAX 0xffffffff
+#   endif
+#  else
+#   error
+#  endif
+# endif
+# define SIZE_MAX SIZE_T_MAX
+#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 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 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 inline void *safe_malloc_add_2op_(size_t size1, size_t size2)
+{
+	size2 += size1;
+	if(size2 < size1)
+		return 0;
+	return safe_malloc_(size2);
+}
+
+static 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 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);
+}
+
+void *safe_malloc_mul_2op_(size_t size1, size_t size2) ;
+
+static 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 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 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;
+	if(size1 > SIZE_MAX / size2)
+		return 0;
+	return malloc(size1*size2);
+}
+
+static 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 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 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 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 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 +1,49 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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
+// JUCE: removed as JUCE already includes standard headers and including
+// these in FlacNamespace will cause problems
+
+//#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 +1,188 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2004-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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"
+
+// JUCE: removed as JUCE already includes this and including stdlib
+// in FlacNamespace will cause problems
+//#include <stdlib.h>
+
+/** \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/compat.h

@@ -0,0 +1,167 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2012-2014  Xiph.org Foundation
+ *
+ * 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.
+ */
+
+/* This is the prefered location of all CPP hackery to make $random_compiler
+ * work like something approaching a C99 (or maybe more accurately GNU99)
+ * compiler.
+ *
+ * It is assumed that this header will be included after "config.h".
+ */
+
+#ifndef FLAC__SHARE__COMPAT_H
+#define FLAC__SHARE__COMPAT_H
+
+#if defined _MSC_VER || defined __BORLANDC__ || defined __MINGW32__
+#define FLAC__off_t __int64 /* use this instead of off_t to fix the 2 GB limit */
+#if !defined __MINGW32__
+#define fseeko _fseeki64
+#define ftello _ftelli64
+#else /* MinGW */
+#if !defined(HAVE_FSEEKO)
+#define fseeko fseeko64
+#define ftello ftello64
+#endif
+#endif
+#else
+#define FLAC__off_t off_t
+#endif
+
+#if defined(_MSC_VER)
+#define strtoll _strtoi64
+#define strtoull _strtoui64
+#endif
+
+#if defined(_MSC_VER)
+#define inline __inline
+#endif
+
+#if defined __INTEL_COMPILER || (defined _MSC_VER && defined _WIN64)
+/* MSVS generates VERY slow 32-bit code with __restrict */
+#define flac_restrict __restrict
+#elif defined __GNUC__
+#define flac_restrict __restrict__
+#else
+#define flac_restrict
+#endif
+
+#define FLAC__U64L(x) x##ULL
+
+#if defined _MSC_VER || defined __BORLANDC__ || defined __MINGW32__
+#define FLAC__STRCASECMP stricmp
+#define FLAC__STRNCASECMP strnicmp
+#else
+#define FLAC__STRCASECMP strcasecmp
+#define FLAC__STRNCASECMP strncasecmp
+#endif
+
+#if defined _MSC_VER
+#  if _MSC_VER >= 1600
+/* Visual Studio 2010 has decent C99 support */
+#    define PRIu64 "llu"
+#    define PRId64 "lld"
+#    define PRIx64 "llx"
+#  else
+#    ifndef UINT32_MAX
+#      define UINT32_MAX _UI32_MAX
+#    endif
+     typedef unsigned __int64 uint64_t;
+     typedef unsigned __int32 uint32_t;
+     typedef unsigned __int16 uint16_t;
+     typedef unsigned __int8 uint8_t;
+     typedef __int64 int64_t;
+     typedef __int32 int32_t;
+     typedef __int16 int16_t;
+     typedef __int8  int8_t;
+#    define PRIu64 "I64u"
+#    define PRId64 "I64d"
+#    define PRIx64 "I64x"
+#  endif
+#endif /* defined _MSC_VER */
+
+#ifdef _WIN32
+/* All char* strings are in UTF-8 format. Added to support Unicode files on Windows */
+#include "win_utf8_io.h"
+
+#define flac_printf printf_utf8
+#define flac_fprintf fprintf_utf8
+#define flac_vfprintf vfprintf_utf8
+#define flac_fopen fopen_utf8
+#define flac_chmod chmod_utf8
+#define flac_utime utime_utf8
+#define flac_unlink unlink_utf8
+#define flac_rename rename_utf8
+#define flac_stat _stat64_utf8
+
+#else
+
+#define flac_printf printf
+#define flac_fprintf fprintf
+#define flac_vfprintf vfprintf
+#define flac_fopen fopen
+#define flac_chmod chmod
+#define flac_utime utime
+#define flac_unlink unlink
+#define flac_rename rename
+#define flac_stat stat
+
+#endif
+
+#ifdef _WIN32
+#define flac_stat_s __stat64 /* stat struct */
+#define flac_fstat _fstat64
+#else
+#define flac_stat_s stat /* stat struct */
+#define flac_fstat fstat
+#endif
+
+#ifndef M_LN2
+#define M_LN2 0.69314718055994530942
+#endif
+#ifndef M_PI
+#define M_PI 3.14159265358979323846
+#endif
+
+/* FLAC needs to compile and work correctly on systems with a normal ISO C99
+ * snprintf as well as Microsoft Visual Studio which has an non-standards
+ * conformant snprint_s function.
+ *
+ * This function wraps the MS version to behave more like the the ISO version.
+ */
+#ifdef __cplusplus
+extern "C" {
+#endif
+int flac_snprintf(char *str, size_t size, const char *fmt, ...);
+int flac_vsnprintf(char *str, size_t size, const char *fmt, va_list va);
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* FLAC__SHARE__COMPAT_H */

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

@@ -0,0 +1,80 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2012-2014  Xiph.org Foundation
+ *
+ * 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.
+ */
+
+/* It is assumed that this header will be included after "config.h". */
+
+#if HAVE_BSWAP32			/* GCC and Clang */
+
+/* GCC prior to 4.8 didn't provide bswap16 on x86_64 */
+#if ! HAVE_BSWAP16
+static inline unsigned short __builtin_bswap16(unsigned short a)
+{
+	return (a<<8)|(a>>8);
+}
+#endif
+
+#define	ENDSWAP_16(x)		(__builtin_bswap16 (x))
+#define	ENDSWAP_32(x)		(__builtin_bswap32 (x))
+
+#elif defined _MSC_VER		/* Windows. Apparently in <stdlib.h>. */
+
+#define	ENDSWAP_16(x)		(_byteswap_ushort (x))
+#define	ENDSWAP_32(x)		(_byteswap_ulong (x))
+
+#elif defined HAVE_BYTESWAP_H		/* Linux */
+
+// JUCE: removed as JUCE already includes standard headers and including
+// these in FlacNamespace will cause problems
+//#include <byteswap.h>
+
+#define	ENDSWAP_16(x)		(bswap_16 (x))
+#define	ENDSWAP_32(x)		(bswap_32 (x))
+
+#else
+
+#define	ENDSWAP_16(x)		((((x) >> 8) & 0xFF) | (((x) & 0xFF) << 8))
+#define	ENDSWAP_32(x)		((((x) >> 24) & 0xFF) | (((x) >> 8) & 0xFF00) | (((x) & 0xFF00) << 8) | (((x) & 0xFF) << 24))
+
+#endif
+
+
+/* Host to little-endian byte swapping. */
+#if CPU_IS_BIG_ENDIAN
+
+#define H2LE_16(x)		ENDSWAP_16 (x)
+#define H2LE_32(x)		ENDSWAP_32 (x)
+
+#else
+
+#define H2LE_16(x)		(x)
+#define H2LE_32(x)		(x)
+
+#endif

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

@@ -0,0 +1,97 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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)
+#define FLAC_API
+
+#elif defined(_MSC_VER)
+#ifdef FLAC_API_EXPORTS
+#define	FLAC_API __declspec(dllexport)
+#else
+#define FLAC_API __declspec(dllimport)
+#endif
+
+#elif defined(FLAC__USE_VISIBILITY_ATTR)
+#define FLAC_API __attribute__ ((visibility ("default")))
+
+#else
+#define FLAC_API
+
+#endif
+
+/** These #defines will mirror the libtool-based library version number, see
+ * http://www.gnu.org/software/libtool/manual/libtool.html#Libtool-versioning
+ */
+#define FLAC_API_VERSION_CURRENT 11
+#define FLAC_API_VERSION_REVISION 0 /**< see above */
+#define FLAC_API_VERSION_AGE 3 /**< 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 +1,109 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include "include/private/bitmath.h"
+
+/* 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 +1,842 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include <stdlib.h>
+#include <string.h>
+#include "include/private/bitwriter.h"
+#include "include/private/crc.h"
+#include "../assert.h"
+#include "../alloc.h"
+#include "../compat.h"
+#include "../endswap.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 uint32_t is >= 32 bits wide */
+#define FLAC__BYTES_PER_WORD 4
+#define FLAC__BITS_PER_WORD (8 * FLAC__BYTES_PER_WORD)
+#define FLAC__WORD_ALL_ONES ((FLAC__uint32)0xffffffff)
+/* SWAP_BE_WORD_TO_HOST swaps bytes in a uint32_t (which is always big-endian) if necessary to match host byte order */
+#if WORDS_BIGENDIAN
+#define SWAP_BE_WORD_TO_HOST(x) (x)
+#else
+#define SWAP_BE_WORD_TO_HOST(x) ENDSWAP_32(x)
+#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(uint32_t); /* size in words */
+/* When growing, increment 4K at a time */
+static const unsigned FLAC__BITWRITER_DEFAULT_INCREMENT = 4096u / sizeof(uint32_t); /* 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)
+
+struct FLAC__BitWriter {
+	uint32_t *buffer;
+	uint32_t 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. */
+#ifndef __SUNPRO_C
+static
+#endif
+FLAC__bool bitwriter_grow_(FLAC__BitWriter *bw, unsigned bits_to_add)
+{
+	unsigned new_capacity;
+	uint32_t *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 = (uint32_t*) safe_realloc_mul_2op_(bw->buffer, sizeof(uint32_t), /*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 = (uint32_t*) malloc(sizeof(uint32_t) * 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;
+}
+
+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 = flac_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;
+}
+
+inline FLAC__bool FLAC__bitwriter_write_raw_uint32(FLAC__BitWriter *bw, FLAC__uint32 val, unsigned bits)
+{
+	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;
+}
+
+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);
+}
+
+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);
+}
+
+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;
+}
+
+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(uint32_t)-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(bw->bits && bw->bits + msbits + lsbits < FLAC__BITS_PER_WORD) { /* i.e. if the whole thing fits in the current uint32_t */
+			/* ^^^ if bw->bits is 0 then we may have filled the buffer and have no free uint32_t 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 {
+			/* 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 uint32_t*/ && !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;
+			}
+		}
+		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;
+}
+
+/* These functions are declared inline in this file but are also callable as
+ * externs from elsewhere.
+ * According to the C99 spec, section 6.7.4, simply providing a function
+ * prototype in a header file without 'inline' and making the function inline
+ * in this file should be sufficient.
+ * Unfortunately, the Microsoft VS compiler doesn't pick them up externally. To
+ * fix that we add extern declarations here.
+ */
+extern FLAC__bool FLAC__bitwriter_write_zeroes(FLAC__BitWriter *bw, unsigned bits);
+extern FLAC__bool FLAC__bitwriter_write_raw_int32(FLAC__BitWriter *bw, FLAC__int32 val, unsigned bits);
+extern FLAC__bool FLAC__bitwriter_write_raw_uint64(FLAC__BitWriter *bw, FLAC__uint64 val, unsigned bits);
+extern FLAC__bool FLAC__bitwriter_write_raw_uint32_little_endian(FLAC__BitWriter *bw, FLAC__uint32 val);
+extern FLAC__bool FLAC__bitwriter_write_byte_block(FLAC__BitWriter *bw, const FLAC__byte vals[], unsigned nvals);

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

@@ -0,0 +1,494 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include "include/private/cpu.h"
+
+#if 0
+ #include <stdlib.h>
+ #include <memory.h>
+ #include <stdio.h>
+#endif
+
+#if defined FLAC__CPU_IA32
+# include <signal.h>
+
+static void disable_sse(FLAC__CPUInfo *info)
+{
+	info->ia32.sse   = false;
+	info->ia32.sse2  = false;
+	info->ia32.sse3  = false;
+	info->ia32.ssse3 = false;
+	info->ia32.sse41 = false;
+	info->ia32.sse42 = false;
+}
+
+static void disable_avx(FLAC__CPUInfo *info)
+{
+	info->ia32.avx     = false;
+	info->ia32.avx2    = false;
+	info->ia32.fma     = false;
+}
+
+#elif defined FLAC__CPU_X86_64
+
+static void disable_avx(FLAC__CPUInfo *info)
+{
+	info->x86.avx     = false;
+	info->x86.avx2    = false;
+	info->x86.fma     = false;
+}
+#endif
+
+#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
+
+#ifdef FLAC__CPU_IA32
+/* 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;
+#endif
+
+/* 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;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE41 = 0x00080000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE42 = 0x00100000;
+
+#if defined FLAC__AVX_SUPPORTED
+/* these are flags in ECX of CPUID AX=00000001 */
+static const unsigned FLAC__CPUINFO_IA32_CPUID_OSXSAVE = 0x08000000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_AVX = 0x10000000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_FMA = 0x00001000;
+/* these are flags in EBX of CPUID AX=00000007 */
+static const unsigned FLAC__CPUINFO_IA32_CPUID_AVX2 = 0x00000020;
+#endif
+
+/*
+ * 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__HAS_X86INTRIN) && !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"
+ */
+#   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;
+	}
+# elif defined(_MSC_VER)
+#  include <windows.h>
+# endif
+#endif
+
+
+void FLAC__cpu_info(FLAC__CPUInfo *info)
+{
+/*
+ * IA32-specific
+ */
+#ifdef FLAC__CPU_IA32
+	FLAC__bool ia32_fxsr = false;
+	FLAC__bool ia32_osxsave = false;
+	(void) ia32_fxsr; (void) ia32_osxsave; /* to avoid warnings about unused variables */
+	memset(info, 0, sizeof(*info));
+	info->type = FLAC__CPUINFO_TYPE_IA32;
+#if !defined FLAC__NO_ASM && (defined FLAC__HAS_NASM || defined FLAC__HAS_X86INTRIN)
+	info->use_asm = true; /* we assume a minimum of 80386 with FLAC__CPU_IA32 */
+#ifdef FLAC__HAS_X86INTRIN
+	if(!FLAC__cpu_have_cpuid_x86())
+		return;
+#else
+	if(!FLAC__cpu_have_cpuid_asm_ia32())
+		return;
+#endif
+	{
+		/* http://www.sandpile.org/x86/cpuid.htm */
+#ifdef FLAC__HAS_X86INTRIN
+		FLAC__uint32 flags_eax, flags_ebx, flags_ecx, flags_edx;
+		FLAC__cpu_info_x86(1, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
+#else
+		FLAC__uint32 flags_ecx, flags_edx;
+		FLAC__cpu_info_asm_ia32(&flags_edx, &flags_ecx);
+#endif
+		info->ia32.cmov  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_CMOV )? true : false;
+		info->ia32.mmx   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_MMX  )? true : false;
+		      ia32_fxsr  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_FXSR )? true : false;
+		info->ia32.sse   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_SSE  )? true : false;
+		info->ia32.sse2  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_SSE2 )? true : false;
+		info->ia32.sse3  = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE3 )? true : false;
+		info->ia32.ssse3 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSSE3)? true : false;
+		info->ia32.sse41 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE41)? true : false;
+		info->ia32.sse42 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE42)? true : false;
+#if defined FLAC__HAS_X86INTRIN && defined FLAC__AVX_SUPPORTED
+		    ia32_osxsave = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_OSXSAVE)? true : false;
+		info->ia32.avx   = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_AVX    )? true : false;
+		info->ia32.fma   = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_FMA    )? true : false;
+		FLAC__cpu_info_x86(7, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
+		info->ia32.avx2  = (flags_ebx & FLAC__CPUINFO_IA32_CPUID_AVX2   )? true : false;
+#endif
+	}
+
+#ifdef DEBUG
+	fprintf(stderr, "CPU info (IA-32):\n");
+	fprintf(stderr, "  CMOV ....... %c\n", info->ia32.cmov    ? 'Y' : 'n');
+	fprintf(stderr, "  MMX ........ %c\n", info->ia32.mmx     ? 'Y' : 'n');
+	fprintf(stderr, "  SSE ........ %c\n", info->ia32.sse     ? 'Y' : 'n');
+	fprintf(stderr, "  SSE2 ....... %c\n", info->ia32.sse2    ? 'Y' : 'n');
+	fprintf(stderr, "  SSE3 ....... %c\n", info->ia32.sse3    ? 'Y' : 'n');
+	fprintf(stderr, "  SSSE3 ...... %c\n", info->ia32.ssse3   ? 'Y' : 'n');
+	fprintf(stderr, "  SSE41 ...... %c\n", info->ia32.sse41   ? 'Y' : 'n');
+	fprintf(stderr, "  SSE42 ...... %c\n", info->ia32.sse42   ? 'Y' : 'n');
+# if defined FLAC__HAS_X86INTRIN && defined FLAC__AVX_SUPPORTED
+	fprintf(stderr, "  AVX ........ %c\n", info->ia32.avx     ? 'Y' : 'n');
+	fprintf(stderr, "  FMA ........ %c\n", info->ia32.fma     ? 'Y' : 'n');
+	fprintf(stderr, "  AVX2 ....... %c\n", info->ia32.avx2    ? 'Y' : 'n');
+# endif
+#endif
+
+	/*
+	 * now have to check for OS support of SSE instructions
+	 */
+	if(info->ia32.sse) {
+#if defined FLAC__NO_SSE_OS
+		/* assume user knows better than us; turn it off */
+		disable_sse(info);
+#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)
+			disable_sse(info);
+#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)
+			disable_sse(info);
+		else { /* double-check SSE2 */
+			mib[1] = CPU_SSE2;
+			len = sizeof(val);
+			if(sysctl(mib, 2, &val, &len, NULL, 0) < 0 || !val) {
+				disable_sse(info);
+				info->ia32.sse = true;
+			}
+		}
+# else
+		disable_sse(info);
+# endif
+#elif defined(__linux__)
+		int sse = 0;
+		struct sigaction sigill_save;
+		struct sigaction sigill_sse;
+		sigill_sse.sa_sigaction = sigill_handler_sse_os;
+      #ifdef __ANDROID__
+        sigemptyset (&sigill_sse.sa_mask);
+      #else
+		__sigemptyset(&sigill_sse.sa_mask);
+      #endif
+		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))
+		{
+			/* http://www.ibiblio.org/gferg/ldp/GCC-Inline-Assembly-HOWTO.html */
+			/* see sigill_handler_sse_os() for an explanation of the following: */
+			asm volatile (
+				"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)
+				: "0"(sse)
+			);
+
+			sigaction(SIGILL, &sigill_save, NULL);
+		}
+
+		if(!sse)
+			disable_sse(info);
+#elif defined(_MSC_VER)
+		__try {
+			__asm {
+				xorps xmm0,xmm0
+			}
+		}
+		__except(EXCEPTION_EXECUTE_HANDLER) {
+			if (_exception_code() == STATUS_ILLEGAL_INSTRUCTION)
+				disable_sse(info);
+		}
+#elif defined(__GNUC__) /* MinGW goes here */
+		int sse = 0;
+		/* Based on the idea described in Agner Fog's manual "Optimizing subroutines in assembly language" */
+		/* In theory, not guaranteed to detect lack of OS SSE support on some future Intel CPUs, but in practice works (see the aforementioned manual) */
+		if (ia32_fxsr) {
+			struct {
+				FLAC__uint32 buff[128];
+			} __attribute__((aligned(16))) fxsr;
+			FLAC__uint32 old_val, new_val;
+
+			asm volatile ("fxsave %0"  : "=m" (fxsr) : "m" (fxsr));
+			old_val = fxsr.buff[50];
+			fxsr.buff[50] ^= 0x0013c0de;                             /* change value in the buffer */
+			asm volatile ("fxrstor %0" : "=m" (fxsr) : "m" (fxsr));  /* try to change SSE register */
+			fxsr.buff[50] = old_val;                                 /* restore old value in the buffer */
+			asm volatile ("fxsave %0 " : "=m" (fxsr) : "m" (fxsr));  /* old value will be overwritten if SSE register was changed */
+			new_val = fxsr.buff[50];                                 /* == old_val if FXRSTOR didn't change SSE register and (old_val ^ 0x0013c0de) otherwise */
+			fxsr.buff[50] = old_val;                                 /* again restore old value in the buffer */
+			asm volatile ("fxrstor %0" : "=m" (fxsr) : "m" (fxsr));  /* restore old values of registers */
+
+			if ((old_val^new_val) == 0x0013c0de)
+				sse = 1;
+		}
+		if(!sse)
+			disable_sse(info);
+#else
+		/* no way to test, disable to be safe */
+		disable_sse(info);
+#endif
+#ifdef DEBUG
+		fprintf(stderr, "  SSE OS sup . %c\n", info->ia32.sse ? 'Y' : 'n');
+#endif
+	}
+	else /* info->ia32.sse == false */
+		disable_sse(info);
+
+	/*
+	 * now have to check for OS support of AVX instructions
+	 */
+	if(info->ia32.avx && ia32_osxsave) {
+		FLAC__uint32 ecr = FLAC__cpu_xgetbv_x86();
+		if ((ecr & 0x6) != 0x6)
+			disable_avx(info);
+#ifdef DEBUG
+		fprintf(stderr, "  AVX OS sup . %c\n", info->ia32.avx ? 'Y' : 'n');
+#endif
+	}
+	else /* no OS AVX support*/
+		disable_avx(info);
+#else
+	info->use_asm = false;
+#endif
+
+/*
+ * x86-64-specific
+ */
+#elif defined FLAC__CPU_X86_64
+	FLAC__bool x86_osxsave = false;
+	(void) x86_osxsave; /* to avoid warnings about unused variables */
+	memset(info, 0, sizeof(*info));
+	info->type = FLAC__CPUINFO_TYPE_X86_64;
+#if !defined FLAC__NO_ASM && defined FLAC__HAS_X86INTRIN
+	info->use_asm = true;
+	{
+		/* http://www.sandpile.org/x86/cpuid.htm */
+		FLAC__uint32 flags_eax, flags_ebx, flags_ecx, flags_edx;
+		FLAC__cpu_info_x86(1, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
+		info->x86.sse3  = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE3 )? true : false;
+		info->x86.ssse3 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSSE3)? true : false;
+		info->x86.sse41 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE41)? true : false;
+		info->x86.sse42 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE42)? true : false;
+#if defined FLAC__AVX_SUPPORTED
+		    x86_osxsave = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_OSXSAVE)? true : false;
+		info->x86.avx   = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_AVX    )? true : false;
+		info->x86.fma   = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_FMA    )? true : false;
+		FLAC__cpu_info_x86(7, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
+		info->x86.avx2  = (flags_ebx & FLAC__CPUINFO_IA32_CPUID_AVX2   )? true : false;
+#endif
+	}
+#ifdef DEBUG
+	fprintf(stderr, "CPU info (x86-64):\n");
+	fprintf(stderr, "  SSE3 ....... %c\n", info->x86.sse3  ? 'Y' : 'n');
+	fprintf(stderr, "  SSSE3 ...... %c\n", info->x86.ssse3 ? 'Y' : 'n');
+	fprintf(stderr, "  SSE41 ...... %c\n", info->x86.sse41 ? 'Y' : 'n');
+	fprintf(stderr, "  SSE42 ...... %c\n", info->x86.sse42 ? 'Y' : 'n');
+# if defined FLAC__AVX_SUPPORTED
+	fprintf(stderr, "  AVX ........ %c\n", info->x86.avx   ? 'Y' : 'n');
+	fprintf(stderr, "  FMA ........ %c\n", info->x86.fma   ? 'Y' : 'n');
+	fprintf(stderr, "  AVX2 ....... %c\n", info->x86.avx2  ? 'Y' : 'n');
+# endif
+#endif
+
+	/*
+	 * now have to check for OS support of AVX instructions
+	 */
+	if(info->x86.avx && x86_osxsave) {
+		FLAC__uint32 ecr = FLAC__cpu_xgetbv_x86();
+		if ((ecr & 0x6) != 0x6)
+			disable_avx(info);
+#ifdef DEBUG
+		fprintf(stderr, "  AVX OS sup . %c\n", info->x86.avx ? 'Y' : 'n');
+#endif
+	}
+	else /* no OS AVX support*/
+		disable_avx(info);
+#else
+	info->use_asm = false;
+#endif
+
+/*
+ * unknown CPU
+ */
+#else
+	info->type = FLAC__CPUINFO_TYPE_UNKNOWN;
+	info->use_asm = false;
+#endif
+}
+
+#if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
+
+#if defined _MSC_VER
+#include <intrin.h> /* for __cpuid() and _xgetbv() */
+#elif defined __GNUC__ && defined HAVE_CPUID_H
+#include <cpuid.h> /* for __get_cpuid() and __get_cpuid_max() */
+#endif
+
+FLAC__uint32 FLAC__cpu_have_cpuid_x86(void)
+{
+#ifdef FLAC__CPU_X86_64
+	return 1;
+#else
+# if defined _MSC_VER || defined __INTEL_COMPILER /* Do they support CPUs w/o CPUID support (or OSes that work on those CPUs)? */
+	FLAC__uint32 flags1, flags2;
+	__asm {
+		pushfd
+		pushfd
+		pop		eax
+		mov		flags1, eax
+		xor		eax, 0x200000
+		push	eax
+		popfd
+		pushfd
+		pop		eax
+		mov		flags2, eax
+		popfd
+	}
+	if (((flags1^flags2) & 0x200000) != 0)
+		return 1;
+	else
+		return 0;
+# elif defined __GNUC__ && defined HAVE_CPUID_H
+	if (__get_cpuid_max(0, 0) != 0)
+		return 1;
+	else
+		return 0;
+# else
+	return 0;
+# endif
+#endif
+}
+
+void FLAC__cpu_info_x86(FLAC__uint32 level, FLAC__uint32 *eax, FLAC__uint32 *ebx, FLAC__uint32 *ecx, FLAC__uint32 *edx)
+{
+    (void) level;
+
+#if defined _MSC_VER || defined __INTEL_COMPILER
+	int cpuinfo[4];
+	int ext = level & 0x80000000;
+	__cpuid(cpuinfo, ext);
+	if((unsigned)cpuinfo[0] < level) {
+		*eax = *ebx = *ecx = *edx = 0;
+		return;
+	}
+#if defined FLAC__AVX_SUPPORTED
+	__cpuidex(cpuinfo, level, 0); /* for AVX2 detection */
+#else
+	__cpuid(cpuinfo, level); /* some old compilers don't support __cpuidex */
+#endif
+	*eax = cpuinfo[0]; *ebx = cpuinfo[1]; *ecx = cpuinfo[2]; *edx = cpuinfo[3];
+#elif defined __GNUC__ && defined HAVE_CPUID_H
+	FLAC__uint32 ext = level & 0x80000000;
+	__cpuid(ext, *eax, *ebx, *ecx, *edx);
+	if (*eax < level) {
+		*eax = *ebx = *ecx = *edx = 0;
+		return;
+	}
+	__cpuid_count(level, 0, *eax, *ebx, *ecx, *edx);
+#else
+	*eax = *ebx = *ecx = *edx = 0;
+#endif
+}
+
+FLAC__uint32 FLAC__cpu_xgetbv_x86(void)
+{
+#if (defined _MSC_VER || defined __INTEL_COMPILER) && defined FLAC__AVX_SUPPORTED
+	return (FLAC__uint32)_xgetbv(0);
+#elif defined __GNUC__
+	FLAC__uint32 lo, hi;
+	asm volatile (".byte 0x0f, 0x01, 0xd0" : "=a"(lo), "=d"(hi) : "c" (0));
+	return lo;
+#else
+	return 0;
+#endif
+}
+
+#endif /* (FLAC__CPU_IA32 || FLAC__CPU_X86_64) && FLAC__HAS_X86INTRIN */

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

@@ -0,0 +1,143 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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.
+ */
+
+#ifdef 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 const 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 +1,418 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include <math.h>
+#include <string.h>
+#include "../compat.h"
+#include "include/private/bitmath.h"
+#include "include/private/fixed.h"
+#include "../assert.h"
+
+#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 < flac_min(flac_min(flac_min(total_error_1, total_error_2), total_error_3), total_error_4))
+		order = 0;
+	else if(total_error_1 < flac_min(flac_min(total_error_2, total_error_3), total_error_4))
+		order = 1;
+	else if(total_error_2 < flac_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 < flac_min(flac_min(flac_min(total_error_1, total_error_2), total_error_3), total_error_4))
+		order = 0;
+	else if(total_error_1 < flac_min(flac_min(total_error_2, total_error_3), total_error_4))
+		order = 1;
+	else if(total_error_2 < flac_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_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 +1,302 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2004-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include "../assert.h"
+#include "../compat.h"
+#include "include/private/float.h"
+
+#ifdef FLAC__INTEGER_ONLY_LIBRARY
+
+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 +1,584 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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.
+ */
+
+#ifdef 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 "../compat.h"
+#include "include/private/format.h"
+
+/* VERSION should come from configure */
+FLAC_API const char *FLAC__VERSION_STRING = VERSION;
+
+FLAC_API const char *FLAC__VENDOR_STRING = "reference libFLAC " VERSION " 20141125";
+
+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_blocksize_is_subset(unsigned blocksize, unsigned sample_rate)
+{
+	if(blocksize > 16384)
+		return false;
+	else if(sample_rate <= 48000 && blocksize > 4608)
+		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 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 (*)(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 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 flac_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 int*) realloc(object->parameters, sizeof(unsigned)*(1 << max_partition_order))))
+			return false;
+		if(0 == (object->raw_bits = (unsigned int*) 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 +1,50 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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 +1,186 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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"
+#include "../../../assert.h"
+
+/* for CHAR_BIT */
+#include <limits.h>
+#include "../../../compat.h"
+
+#if defined(_MSC_VER) && (_MSC_VER >= 1400)
+#include <intrin.h> /* for _BitScanReverse* */
+#endif
+
+/* Will never be emitted for MSVC, GCC, Intel compilers */
+static inline unsigned int FLAC__clz_soft_uint32(unsigned int word)
+{
+    static const unsigned char byte_to_unary_table[] = {
+    8, 7, 6, 6, 5, 5, 5, 5, 4, 4, 4, 4, 4, 4, 4, 4,
+    3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3,
+    2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
+    2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    };
+
+    return (word) > 0xffffff ? byte_to_unary_table[(word) >> 24] :
+    (word) > 0xffff ? byte_to_unary_table[(word) >> 16] + 8 :
+    (word) > 0xff ? byte_to_unary_table[(word) >> 8] + 16 :
+    byte_to_unary_table[(word)] + 24;
+}
+
+static inline unsigned int FLAC__clz_uint32(FLAC__uint32 v)
+{
+/* Never used with input 0 */
+    FLAC__ASSERT(v > 0);
+#if defined(__INTEL_COMPILER)
+    return _bit_scan_reverse(v) ^ 31U;
+#elif defined(__GNUC__) && (__GNUC__ >= 4 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 4))
+/* This will translate either to (bsr ^ 31U), clz , ctlz, cntlz, lzcnt depending on
+ * -march= setting or to a software routine in exotic machines. */
+    return __builtin_clz(v);
+#elif defined(_MSC_VER) && (_MSC_VER >= 1400)
+    {
+        unsigned long idx;
+        _BitScanReverse(&idx, v);
+        return idx ^ 31U;
+    }
+#else
+    return FLAC__clz_soft_uint32(v);
+#endif
+}
+
+/* This one works with input 0 */
+static inline unsigned int FLAC__clz2_uint32(FLAC__uint32 v)
+{
+    if (!v)
+        return 32;
+    return FLAC__clz_uint32(v);
+}
+
+/* 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
+ */
+
+static inline unsigned FLAC__bitmath_ilog2(FLAC__uint32 v)
+{
+    FLAC__ASSERT(v > 0);
+#if defined(__INTEL_COMPILER)
+    return _bit_scan_reverse(v);
+#elif defined(_MSC_VER) && (_MSC_VER >= 1400)
+    {
+        unsigned long idx;
+        _BitScanReverse(&idx, v);
+        return idx;
+    }
+#else
+    return sizeof(FLAC__uint32) * CHAR_BIT  - 1 - FLAC__clz_uint32(v);
+#endif
+}
+
+
+#ifdef FLAC__INTEGER_ONLY_LIBRARY /* Unused otherwise */
+
+static inline unsigned FLAC__bitmath_ilog2_wide(FLAC__uint64 v)
+{
+    FLAC__ASSERT(v > 0);
+#if defined(__GNUC__) && (__GNUC__ >= 4 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 4))
+    return sizeof(FLAC__uint64) * CHAR_BIT - 1 - __builtin_clzll(v);
+/* Sorry, only supported in x64/Itanium.. and both have fast FPU which makes integer-only encoder pointless */
+#elif (defined(_MSC_VER) && (_MSC_VER >= 1400)) && (defined(_M_IA64) || defined(_M_X64))
+    {
+        unsigned long idx;
+        _BitScanReverse64(&idx, v);
+        return idx;
+    }
+#else
+/*  Brain-damaged compilers will use the fastest possible way that is,
+    de Bruijn sequences (http://supertech.csail.mit.edu/papers/debruijn.pdf)
+    (C) Timothy B. Terriberry (tterribe@xiph.org) 2001-2009 CC0 (Public domain).
+*/
+    {
+        static const unsigned char DEBRUIJN_IDX64[64]={
+            0, 1, 2, 7, 3,13, 8,19, 4,25,14,28, 9,34,20,40,
+            5,17,26,38,15,46,29,48,10,31,35,54,21,50,41,57,
+            63, 6,12,18,24,27,33,39,16,37,45,47,30,53,49,56,
+            62,11,23,32,36,44,52,55,61,22,43,51,60,42,59,58
+        };
+        v|= v>>1;
+        v|= v>>2;
+        v|= v>>4;
+        v|= v>>8;
+        v|= v>>16;
+        v|= v>>32;
+        v= (v>>1)+1;
+        return DEBRUIJN_IDX64[v*0x218A392CD3D5DBF>>58&0x3F];
+    }
+#endif
+}
+#endif
+
+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 +1,91 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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__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);
+#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);
+#endif

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

@@ -0,0 +1,104 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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 +1,99 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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_X86_64,
+	FLAC__CPUINFO_TYPE_UNKNOWN
+} FLAC__CPUInfo_Type;
+
+#if defined FLAC__CPU_IA32
+typedef struct {
+	FLAC__bool cmov;
+	FLAC__bool mmx;
+	FLAC__bool sse;
+	FLAC__bool sse2;
+
+	FLAC__bool sse3;
+	FLAC__bool ssse3;
+	FLAC__bool sse41;
+	FLAC__bool sse42;
+	FLAC__bool avx;
+	FLAC__bool avx2;
+	FLAC__bool fma;
+} FLAC__CPUInfo_IA32;
+#elif defined FLAC__CPU_X86_64
+typedef struct {
+	FLAC__bool sse3;
+	FLAC__bool ssse3;
+	FLAC__bool sse41;
+	FLAC__bool sse42;
+	FLAC__bool avx;
+	FLAC__bool avx2;
+	FLAC__bool fma;
+} FLAC__CPUInfo_x86;
+#endif
+
+typedef struct {
+	FLAC__bool use_asm;
+	FLAC__CPUInfo_Type type;
+#if defined FLAC__CPU_IA32
+	FLAC__CPUInfo_IA32 ia32;
+#elif defined FLAC__CPU_X86_64
+	FLAC__CPUInfo_x86 x86;
+#endif
+} FLAC__CPUInfo;
+
+void FLAC__cpu_info(FLAC__CPUInfo *info);
+
+#ifndef FLAC__NO_ASM
+# if defined FLAC__CPU_IA32 && defined 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);
+# endif
+# if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
+FLAC__uint32 FLAC__cpu_have_cpuid_x86(void);
+void         FLAC__cpu_info_x86(FLAC__uint32 level, FLAC__uint32 *eax, FLAC__uint32 *ebx, FLAC__uint32 *ecx, FLAC__uint32 *edx);
+FLAC__uint32 FLAC__cpu_xgetbv_x86(void);
+# endif
+#endif
+
+#endif

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

@@ -0,0 +1,62 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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 const 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 +1,107 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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 "cpu.h"
+#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]);
+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]);
+# ifndef FLAC__NO_ASM
+#  if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
+#   ifdef FLAC__SSE2_SUPPORTED
+unsigned FLAC__fixed_compute_best_predictor_intrin_sse2(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER + 1]);
+unsigned FLAC__fixed_compute_best_predictor_wide_intrin_sse2(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER + 1]);
+#   endif
+#   ifdef FLAC__SSSE3_SUPPORTED
+unsigned FLAC__fixed_compute_best_predictor_intrin_ssse3(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
+unsigned FLAC__fixed_compute_best_predictor_wide_intrin_ssse3(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER + 1]);
+#   endif
+#  endif
+#  if defined FLAC__CPU_IA32 && defined 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
+#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 +1,98 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2004-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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 +1,45 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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 +1,246 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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 "cpu.h"
+#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_sse_lag_16(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+#    endif
+#  endif
+#  if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
+#    ifdef FLAC__SSE_SUPPORTED
+void FLAC__lpc_compute_autocorrelation_intrin_sse_lag_4(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+void FLAC__lpc_compute_autocorrelation_intrin_sse_lag_8(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+void FLAC__lpc_compute_autocorrelation_intrin_sse_lag_12(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
+void FLAC__lpc_compute_autocorrelation_intrin_sse_lag_16(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[]);
+void FLAC__lpc_compute_residual_from_qlp_coefficients_wide_asm_ia32(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+#    endif
+#  endif
+#  if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
+#    ifdef FLAC__SSE2_SUPPORTED
+void FLAC__lpc_compute_residual_from_qlp_coefficients_16_intrin_sse2(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_intrin_sse2(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+#    endif
+#    ifdef FLAC__SSE4_1_SUPPORTED
+void FLAC__lpc_compute_residual_from_qlp_coefficients_intrin_sse41(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_intrin_sse41(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
+#    endif
+#    ifdef FLAC__AVX2_SUPPORTED
+void FLAC__lpc_compute_residual_from_qlp_coefficients_16_intrin_avx2(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_intrin_avx2(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_intrin_avx2(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[]);
+void FLAC__lpc_restore_signal_wide_asm_ia32(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
+#    endif /* FLAC__HAS_NASM */
+#  endif /* FLAC__CPU_IA32 */
+#  if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
+#    ifdef FLAC__SSE2_SUPPORTED
+void FLAC__lpc_restore_signal_16_intrin_sse2(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
+#    endif
+#    ifdef FLAC__SSE4_1_SUPPORTED
+void FLAC__lpc_restore_signal_wide_intrin_sse41(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
+#    endif
+#  endif
+#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 +1,50 @@
+#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 union {
+	FLAC__byte *p8;
+	FLAC__int16 *p16;
+	FLAC__int32 *p32;
+} FLAC__multibyte;
+
+typedef struct {
+	FLAC__uint32 in[16];
+	FLAC__uint32 buf[4];
+	FLAC__uint32 bytes[2];
+	FLAC__multibyte 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 +1,58 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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(size_t elements, FLAC__int32 **unaligned_pointer, FLAC__int32 **aligned_pointer);
+FLAC__bool FLAC__memory_alloc_aligned_uint32_array(size_t elements, FLAC__uint32 **unaligned_pointer, FLAC__uint32 **aligned_pointer);
+FLAC__bool FLAC__memory_alloc_aligned_uint64_array(size_t elements, FLAC__uint64 **unaligned_pointer, FLAC__uint64 **aligned_pointer);
+FLAC__bool FLAC__memory_alloc_aligned_unsigned_array(size_t elements, unsigned **unaligned_pointer, unsigned **aligned_pointer);
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+FLAC__bool FLAC__memory_alloc_aligned_real_array(size_t elements, FLAC__real **unaligned_pointer, FLAC__real **aligned_pointer);
+#endif
+void *safe_malloc_mul_2op_p(size_t size1, size_t size2);
+
+#endif

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

@@ -0,0 +1,46 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2002-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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.h

@@ -0,0 +1,67 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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_H
+#define FLAC__PRIVATE__STREAM_ENCODER_H
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+/*
+ * This is used to avoid overflow with unusual signals in 32-bit
+ * accumulator in the *precompute_partition_info_sums_* functions.
+ */
+#define FLAC__MAX_EXTRA_RESIDUAL_BPS 4
+
+#if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
+#include "cpu.h"
+#include "../../../format.h"
+
+#ifdef FLAC__SSE2_SUPPORTED
+extern void FLAC__precompute_partition_info_sums_intrin_sse2(const FLAC__int32 residual[], FLAC__uint64 abs_residual_partition_sums[],
+			unsigned residual_samples, unsigned predictor_order, unsigned min_partition_order, unsigned max_partition_order, unsigned bps);
+#endif
+
+#ifdef FLAC__SSSE3_SUPPORTED
+extern void FLAC__precompute_partition_info_sums_intrin_ssse3(const FLAC__int32 residual[], FLAC__uint64 abs_residual_partition_sums[],
+			unsigned residual_samples, unsigned predictor_order, unsigned min_partition_order, unsigned max_partition_order, unsigned bps);
+#endif
+
+#ifdef FLAC__AVX2_SUPPORTED
+extern void FLAC__precompute_partition_info_sums_intrin_avx2(const FLAC__int32 residual[], FLAC__uint64 abs_residual_partition_sums[],
+			unsigned residual_samples, unsigned predictor_order, unsigned min_partition_order, unsigned max_partition_order, unsigned bps);
+#endif
+
+#endif
+
+#endif

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

@@ -0,0 +1,46 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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 +1,74 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2006-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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_partial_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p, const FLAC__real start, const FLAC__real end);
+void FLAC__window_punchout_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p, const FLAC__real start, const FLAC__real end);
+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 +1,39 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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 +1,60 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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 "../private/ogg_decoder_aspect.h"
+#endif
+
+typedef struct FLAC__StreamDecoderProtected {
+	FLAC__StreamDecoderState state;
+	FLAC__StreamDecoderInitStatus initstate;
+	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 +1,118 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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_PARTIAL_TUKEY,
+	FLAC__APODIZATION_PUNCHOUT_TUKEY,
+	FLAC__APODIZATION_WELCH
+} FLAC__ApodizationFunction;
+
+typedef struct {
+	FLAC__ApodizationFunction type;
+	union {
+		struct {
+			FLAC__real stddev;
+		} gauss;
+		struct {
+			FLAC__real p;
+		} tukey;
+		struct {
+			FLAC__real p;
+			FLAC__real start;
+			FLAC__real end;
+		} multiple_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 +1,518 @@
+#ifdef 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"
+#include "../endswap.h"
+
+/*
+ * 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])
+{
+	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.p8= 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);
+	if (0 != ctx->internal_buf.p8) {
+		free(ctx->internal_buf.p8);
+		ctx->internal_buf.p8= 0;
+		ctx->capacity = 0;
+	}
+	memset(ctx, 0, sizeof(*ctx));	/* In case it's sensitive */
+}
+
+/*
+ * Convert the incoming audio signal to a byte stream
+ */
+static void format_input_(FLAC__multibyte *mbuf, const FLAC__int32 * const signal[], unsigned channels, unsigned samples, unsigned bytes_per_sample)
+{
+	FLAC__byte *buf_ = mbuf->p8;
+	FLAC__int16 *buf16 = mbuf->p16;
+	FLAC__int32 *buf32 = mbuf->p32;
+	FLAC__int32 a_word;
+	unsigned channel, sample;
+
+	/* Storage in the output buffer, buf, is little endian. */
+
+#define BYTES_CHANNEL_SELECTOR(bytes, channels)   (bytes * 100 + channels)
+
+	/* First do the most commonly used combinations. */
+	switch (BYTES_CHANNEL_SELECTOR (bytes_per_sample, channels)) {
+		/* One byte per sample. */
+		case (BYTES_CHANNEL_SELECTOR (1, 1)):
+			for (sample = 0; sample < samples; sample++)
+				*buf_++ = signal[0][sample];
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (1, 2)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf_++ = signal[0][sample];
+				*buf_++ = signal[1][sample];
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (1, 4)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf_++ = signal[0][sample];
+				*buf_++ = signal[1][sample];
+				*buf_++ = signal[2][sample];
+				*buf_++ = signal[3][sample];
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (1, 6)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf_++ = signal[0][sample];
+				*buf_++ = signal[1][sample];
+				*buf_++ = signal[2][sample];
+				*buf_++ = signal[3][sample];
+				*buf_++ = signal[4][sample];
+				*buf_++ = signal[5][sample];
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (1, 8)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf_++ = signal[0][sample];
+				*buf_++ = signal[1][sample];
+				*buf_++ = signal[2][sample];
+				*buf_++ = signal[3][sample];
+				*buf_++ = signal[4][sample];
+				*buf_++ = signal[5][sample];
+				*buf_++ = signal[6][sample];
+				*buf_++ = signal[7][sample];
+			}
+			return;
+
+		/* Two bytes per sample. */
+		case (BYTES_CHANNEL_SELECTOR (2, 1)):
+			for (sample = 0; sample < samples; sample++)
+				*buf16++ = H2LE_16(signal[0][sample]);
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (2, 2)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf16++ = H2LE_16(signal[0][sample]);
+				*buf16++ = H2LE_16(signal[1][sample]);
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (2, 4)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf16++ = H2LE_16(signal[0][sample]);
+				*buf16++ = H2LE_16(signal[1][sample]);
+				*buf16++ = H2LE_16(signal[2][sample]);
+				*buf16++ = H2LE_16(signal[3][sample]);
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (2, 6)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf16++ = H2LE_16(signal[0][sample]);
+				*buf16++ = H2LE_16(signal[1][sample]);
+				*buf16++ = H2LE_16(signal[2][sample]);
+				*buf16++ = H2LE_16(signal[3][sample]);
+				*buf16++ = H2LE_16(signal[4][sample]);
+				*buf16++ = H2LE_16(signal[5][sample]);
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (2, 8)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf16++ = H2LE_16(signal[0][sample]);
+				*buf16++ = H2LE_16(signal[1][sample]);
+				*buf16++ = H2LE_16(signal[2][sample]);
+				*buf16++ = H2LE_16(signal[3][sample]);
+				*buf16++ = H2LE_16(signal[4][sample]);
+				*buf16++ = H2LE_16(signal[5][sample]);
+				*buf16++ = H2LE_16(signal[6][sample]);
+				*buf16++ = H2LE_16(signal[7][sample]);
+			}
+			return;
+
+		/* Three bytes per sample. */
+		case (BYTES_CHANNEL_SELECTOR (3, 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;
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (3, 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;
+			}
+			return;
+
+		/* Four bytes per sample. */
+		case (BYTES_CHANNEL_SELECTOR (4, 1)):
+			for (sample = 0; sample < samples; sample++)
+				*buf32++ = H2LE_32(signal[0][sample]);
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (4, 2)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf32++ = H2LE_32(signal[0][sample]);
+				*buf32++ = H2LE_32(signal[1][sample]);
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (4, 4)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf32++ = H2LE_32(signal[0][sample]);
+				*buf32++ = H2LE_32(signal[1][sample]);
+				*buf32++ = H2LE_32(signal[2][sample]);
+				*buf32++ = H2LE_32(signal[3][sample]);
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (4, 6)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf32++ = H2LE_32(signal[0][sample]);
+				*buf32++ = H2LE_32(signal[1][sample]);
+				*buf32++ = H2LE_32(signal[2][sample]);
+				*buf32++ = H2LE_32(signal[3][sample]);
+				*buf32++ = H2LE_32(signal[4][sample]);
+				*buf32++ = H2LE_32(signal[5][sample]);
+			}
+			return;
+
+		case (BYTES_CHANNEL_SELECTOR (4, 8)):
+			for (sample = 0; sample < samples; sample++) {
+				*buf32++ = H2LE_32(signal[0][sample]);
+				*buf32++ = H2LE_32(signal[1][sample]);
+				*buf32++ = H2LE_32(signal[2][sample]);
+				*buf32++ = H2LE_32(signal[3][sample]);
+				*buf32++ = H2LE_32(signal[4][sample]);
+				*buf32++ = H2LE_32(signal[5][sample]);
+				*buf32++ = H2LE_32(signal[6][sample]);
+				*buf32++ = H2LE_32(signal[7][sample]);
+			}
+			return;
+
+		default:
+			break;
+	}
+
+	/* General version. */
+	switch (bytes_per_sample) {
+		case 1:
+			for (sample = 0; sample < samples; sample++)
+				for (channel = 0; channel < channels; channel++)
+					*buf_++ = signal[channel][sample];
+			return;
+
+		case 2:
+			for (sample = 0; sample < samples; sample++)
+				for (channel = 0; channel < channels; channel++)
+					*buf16++ = H2LE_16(signal[channel][sample]);
+			return;
+
+		case 3:
+			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;
+				}
+			return;
+
+		case 4:
+			for (sample = 0; sample < samples; sample++)
+				for (channel = 0; channel < channels; channel++)
+					*buf32++ = H2LE_32(signal[channel][sample]);
+			return;
+
+		default:
+			break;
+	}
+}
+
+/*
+ * 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.p8, bytes_needed);
+		if (0 == tmp) {
+			free(ctx->internal_buf.p8);
+			if (0 == (ctx->internal_buf.p8= (FLAC__byte*) safe_malloc_(bytes_needed)))
+				return false;
+		}
+		else
+			ctx->internal_buf.p8= tmp;
+		ctx->capacity = bytes_needed;
+	}
+
+	format_input_(&ctx->internal_buf, signal, channels, samples, bytes_per_sample);
+
+	FLAC__MD5Update(ctx, ctx->internal_buf.p8, bytes_needed);
+
+	return true;
+}

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

@@ -0,0 +1,218 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#ifdef HAVE_STDINT_H
+#include <stdint.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, /*+*/31L);
+	*aligned_address = (void*)(((uintptr_t)x + 31L) & -32L);
+#else
+	x = safe_malloc_(bytes);
+	*aligned_address = x;
+#endif
+	return x;
+}
+
+FLAC__bool FLAC__memory_alloc_aligned_int32_array(size_t 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);
+
+	if(elements > SIZE_MAX / sizeof(*pu)) /* overflow check */
+		return false;
+
+	pu = (FLAC__int32*) 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_uint32_array(size_t 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);
+
+	if(elements > SIZE_MAX / sizeof(*pu)) /* overflow check */
+		return false;
+
+	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(size_t 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);
+
+	if(elements > SIZE_MAX / sizeof(*pu)) /* overflow check */
+		return false;
+
+	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(size_t 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);
+
+	if(elements > SIZE_MAX / sizeof(*pu)) /* overflow check */
+		return false;
+
+	pu = (unsigned int*) 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(size_t 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);
+
+	if(elements > SIZE_MAX / sizeof(*pu)) /* overflow check */
+		return false;
+
+	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
+
+void *safe_malloc_mul_2op_p(size_t size1, size_t size2)
+{
+	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);
+}

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

@@ -0,0 +1,549 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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.
+ */
+
+#ifdef 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"
+
+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 *indx = track->indices + j;
+
+					if(!FLAC__bitwriter_write_raw_uint64(bw, indx->offset, FLAC__STREAM_METADATA_CUESHEET_INDEX_OFFSET_LEN))
+						return false;
+					if(!FLAC__bitwriter_write_raw_uint32(bw, indx->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 +1,281 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2006-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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.
+ */
+
+#ifdef 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
+
+
+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 / (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)));
+}
+
+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 + 1.0f);
+		for (; n <= L; n++)
+			window[n-1] = (float)(2 * (L - n + 1)) / ((float)L + 1.0f);
+	}
+}
+
+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_partial_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p, const FLAC__real start, const FLAC__real end)
+{
+	const FLAC__int32 start_n = (FLAC__int32)(start * L);
+	const FLAC__int32 end_n = (FLAC__int32)(end * L);
+	const FLAC__int32 N = end_n - start_n;
+	FLAC__int32 Np, n, i;
+
+	if (p <= 0.0f)
+		FLAC__window_partial_tukey(window, L, 0.05f, start, end);
+	else if (p >= 1.0f)
+		FLAC__window_partial_tukey(window, L, 0.95f, start, end);
+	else {
+
+		Np = (FLAC__int32)(p / 2.0f * N);
+
+		for (n = 0; n < start_n && n < L; n++)
+			window[n] = 0.0f;
+		for (i = 1; n < (start_n+Np) && n < L; n++, i++)
+			window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Np));
+		for (; n < (end_n-Np) && n < L; n++)
+			window[n] = 1.0f;
+		for (i = Np; n < end_n && n < L; n++, i--)
+			window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Np));
+		for (; n < L; n++)
+			window[n] = 0.0f;
+	}
+}
+
+void FLAC__window_punchout_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p, const FLAC__real start, const FLAC__real end)
+{
+	const FLAC__int32 start_n = (FLAC__int32)(start * L);
+	const FLAC__int32 end_n = (FLAC__int32)(end * L);
+	FLAC__int32 Ns, Ne, n, i;
+
+	if (p <= 0.0f)
+		FLAC__window_punchout_tukey(window, L, 0.05f, start, end);
+	else if (p >= 1.0f)
+		FLAC__window_punchout_tukey(window, L, 0.95f, start, end);
+	else {
+
+		Ns = (FLAC__int32)(p / 2.0f * start_n);
+		Ne = (FLAC__int32)(p / 2.0f * (L - end_n));
+
+		for (n = 0, i = 1; n < Ns && n < L; n++, i++)
+			window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Ns));
+		for (; n < start_n-Ns && n < L; n++)
+			window[n] = 1.0f;
+		for (i = Ns; n < start_n && n < L; n++, i--)
+			window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Ns));
+		for (; n < end_n && n < L; n++)
+			window[n] = 0.0f;
+		for (i = 1; n < end_n+Ne && n < L; n++, i++)
+			window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Ne));
+		for (; n < L - (Ne) && n < L; n++)
+			window[n] = 1.0f;
+		for (i = Ne; n < L; n++, i--)
+			window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Ne));
+	}
+}
+
+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 +1,86 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000-2009  Josh Coalson
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
+ *
+ * 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) && _MSC_VER < 1600
+
+/* Microsoft Visual Studio earlier than the 2010 version did not provide
+ * the 1999 ISO C Standard header file <stdint.h>.
+ */
+
+typedef __int8 FLAC__int8;
+typedef unsigned __int8 FLAC__uint8;
+
+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;
+
+#else
+
+/* For MSVC 2010 and everything else which provides <stdint.h>. */
+
+#include <stdint.h>
+
+typedef int8_t FLAC__int8;
+typedef uint8_t FLAC__uint8;
+
+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/flac/win_utf8_io.h

@@ -0,0 +1,64 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2013-2014  Xiph.Org Foundation
+ *
+ * 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.
+ */
+
+#ifdef _WIN32
+
+#ifndef flac__win_utf8_io_h
+#define flac__win_utf8_io_h
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+int get_utf8_argv(int *argc, char ***argv);
+
+int printf_utf8(const char *format, ...);
+int fprintf_utf8(FILE *stream, const char *format, ...);
+int vfprintf_utf8(FILE *stream, const char *format, va_list argptr);
+
+FILE *fopen_utf8(const char *filename, const char *mode);
+int stat_utf8(const char *path, struct stat *buffer);
+int _stat64_utf8(const char *path, struct __stat64 *buffer);
+int chmod_utf8(const char *filename, int pmode);
+int utime_utf8(const char *filename, struct utimbuf *times);
+int unlink_utf8(const char *filename);
+int rename_utf8(const char *oldname, const char *newname);
+size_t strlen_utf8(const char *str);
+int win_get_console_width(void);
+int print_console(FILE *stream, const wchar_t *text, size_t len);
+HANDLE WINAPI CreateFile_utf8(const char *lpFileName, DWORD dwDesiredAccess, DWORD dwShareMode, LPSECURITY_ATTRIBUTES lpSecurityAttributes, DWORD dwCreationDisposition, DWORD dwFlagsAndAttributes, HANDLE hTemplateFile);
+
+#ifdef __cplusplus
+} /* extern "C" */
+#endif
+
+#endif
+#endif

source/modules/juce_audio_formats/codecs/juce_AiffAudioFormat.h

@@ -0,0 +1,92 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2017 - ROLI Ltd.
+
+   JUCE is an open source library subject to commercial or open-source
+   licensing.
+
+   By using JUCE, you agree to the terms of both the JUCE 5 End-User License
+   Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
+   27th April 2017).
+
+   End User License Agreement: www.juce.com/juce-5-licence
+   Privacy Policy: www.juce.com/juce-5-privacy-policy
+
+   Or: You may also use this code under the terms of the GPL v3 (see
+   www.gnu.org/licenses).
+
+   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
+   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
+   DISCLAIMED.
+
+  ==============================================================================
+*/
+
+namespace juce
+{
+
+//==============================================================================
+/**
+    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;
+    MemoryMappedAudioFormatReader* createMemoryMappedReader (FileInputStream*) 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)
+};
+
+} // namespace juce

source/modules/juce_audio_formats/codecs/juce_CoreAudioFormat.cpp

@@ -0,0 +1,840 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2017 - ROLI Ltd.
+
+   JUCE is an open source library subject to commercial or open-source
+   licensing.
+
+   By using JUCE, you agree to the terms of both the JUCE 5 End-User License
+   Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
+   27th April 2017).
+
+   End User License Agreement: www.juce.com/juce-5-licence
+   Privacy Policy: www.juce.com/juce-5-privacy-policy
+
+   Or: You may also use this code under the terms of the GPL v3 (see
+   www.gnu.org/licenses).
+
+   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
+   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
+   DISCLAIMED.
+
+  ==============================================================================
+*/
+
+#if JUCE_MAC || JUCE_IOS
+
+#include "../../juce_audio_basics/native/juce_mac_CoreAudioLayouts.h"
+
+namespace juce
+{
+
+//==============================================================================
+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;
+    };
+
+    //==============================================================================
+    static StringPairArray parseUserDefinedChunk (InputStream& input, int64 size)
+    {
+        StringPairArray infoStrings;
+        const int64 originalPosition = input.getPosition();
+
+        uint8 uuid[16];
+        input.read (uuid, sizeof (uuid));
+
+        if (memcmp (uuid, "\x29\x81\x92\x73\xB5\xBF\x4A\xEF\xB7\x8D\x62\xD1\xEF\x90\xBB\x2C", 16) == 0)
+        {
+            const uint32 numEntries = (uint32) input.readIntBigEndian();
+
+            for (uint32 i = 0; i < numEntries && input.getPosition() < originalPosition + size; ++i)
+            {
+                String keyName = input.readString();
+                infoStrings.set (keyName, input.readString());
+            }
+        }
+
+        input.setPosition (originalPosition + size);
+        return infoStrings;
+    }
+
+    //==============================================================================
+    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"))
+                {
+                    metadataValues.addArray (parseUserDefinedChunk (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, 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);
+
+                HeapBlock<AudioChannelLayout> caLayout;
+                bool hasLayout = false;
+                UInt32 sizeOfLayout = 0, isWritable = 0;
+
+                status = AudioFileGetPropertyInfo (audioFileID, kAudioFilePropertyChannelLayout, &sizeOfLayout, &isWritable);
+
+                if (status == noErr)
+                {
+                    caLayout.malloc (1, static_cast<size_t> (sizeOfLayout));
+
+                    status = AudioFileGetProperty (audioFileID, kAudioFilePropertyChannelLayout,
+                                                   &sizeOfLayout, caLayout.get());
+
+                    if (status == noErr)
+                    {
+                        auto fileLayout = CoreAudioLayouts::fromCoreAudio (*caLayout.get());
+
+                        if (fileLayout.size() == static_cast<int> (numChannels))
+                        {
+                            hasLayout = true;
+                            channelSet = fileLayout;
+                        }
+                    }
+                }
+
+                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;
+                    channelMap.malloc (numChannels);
+
+                    if (hasLayout && caLayout != nullptr)
+                    {
+                        auto caOrder = CoreAudioLayouts::getCoreAudioLayoutChannels (*caLayout);
+
+                        for (int i = 0; i < static_cast<int> (numChannels); ++i)
+                        {
+                            auto idx = channelSet.getChannelIndexForType (caOrder.getReference (i));
+                            jassert (isPositiveAndBelow (idx, static_cast<int> (numChannels)));
+
+                            channelMap[i] = idx;
+                        }
+                    }
+                    else
+                    {
+                        for (int i = 0; i < static_cast<int> (numChannels); ++i)
+                            channelMap[i] = i;
+                    }
+
+                    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;)
+            {
+                int* dest = destSamples[(i < (int) numChannels ? channelMap[i] : i)];
+
+                if (dest != nullptr)
+                {
+                    if (i < (int) numChannels)
+                        memcpy (dest + startOffsetInDestBuffer, bufferList->mBuffers[i].mData, numBytes);
+                    else
+                        zeromem (dest + startOffsetInDestBuffer, numBytes);
+                }
+            }
+
+            startOffsetInDestBuffer += numThisTime;
+            numSamples -= numThisTime;
+            lastReadPosition += numThisTime;
+        }
+
+        return true;
+    }
+
+    AudioChannelSet getChannelLayout() override
+    {
+        if (channelSet.size() == static_cast<int> (numChannels)) return channelSet;
+
+        return AudioFormatReader::getChannelLayout();
+    }
+
+    bool ok;
+
+private:
+    AudioFileID audioFileID;
+    ExtAudioFileRef audioFileRef;
+    AudioChannelSet channelSet;
+    AudioStreamBasicDescription destinationAudioFormat;
+    MemoryBlock audioDataBlock;
+    HeapBlock<AudioBufferList> bufferList;
+    int64 lastReadPosition;
+    HeapBlock<int> channelMap;
+
+    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 (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;
+}
+
+//==============================================================================
+// Unit tests for Core Audio layout conversions
+//==============================================================================
+#if JUCE_UNIT_TESTS
+
+#define DEFINE_CHANNEL_LAYOUT_DFL_ENTRY(x) CoreAudioChannelLayoutTag { x, #x, AudioChannelSet() }
+#define DEFINE_CHANNEL_LAYOUT_TAG_ENTRY(x, y) CoreAudioChannelLayoutTag { x, #x, y }
+
+class CoreAudioLayoutsUnitTest  : public UnitTest
+{
+public:
+    CoreAudioLayoutsUnitTest() : UnitTest ("Core Audio Layout <-> JUCE channel layout conversion", "Audio") {}
+
+    void runTest() override
+    {
+        auto& knownTags = getAllKnownLayoutTags();
+
+        {
+            // Check that all known tags defined in CoreAudio SDK version 10.12.4 are known to JUCE
+            // Include all defined tags even if there are duplicates as Apple will sometimes change
+            // definitions
+            beginTest ("All CA tags handled");
+
+            for (auto tagEntry : knownTags)
+            {
+                auto labels = CoreAudioLayouts::fromCoreAudio (tagEntry.tag);
+
+                expect (! labels.isDiscreteLayout(), String ("Tag \"") + String (tagEntry.name) + "\" is not handled by JUCE");
+            }
+        }
+
+        {
+            beginTest ("Number of speakers");
+
+            for (auto tagEntry : knownTags)
+            {
+                auto labels = CoreAudioLayouts::getSpeakerLayoutForCoreAudioTag (tagEntry.tag);
+
+                expect (labels.size() == (tagEntry.tag & 0xffff), String ("Tag \"") + String (tagEntry.name) + "\" has incorrect channel count");
+            }
+        }
+
+        {
+            beginTest ("No duplicate speaker");
+
+            for (auto tagEntry : knownTags)
+            {
+                auto labels = CoreAudioLayouts::getSpeakerLayoutForCoreAudioTag (tagEntry.tag);
+                labels.sort();
+
+                for (int i = 0; i < (labels.size() - 1); ++i)
+                    expect (labels.getReference (i) != labels.getReference (i + 1),
+                            String ("Tag \"") + String (tagEntry.name) + "\" has the same speaker twice");
+            }
+        }
+
+        {
+            beginTest ("CA speaker list and juce layouts are consistent");
+
+            for (auto tagEntry : knownTags)
+                expect (AudioChannelSet::channelSetWithChannels (CoreAudioLayouts::getSpeakerLayoutForCoreAudioTag (tagEntry.tag))
+                            == CoreAudioLayouts::fromCoreAudio (tagEntry.tag),
+                        String ("Tag \"") + String (tagEntry.name) + "\" is not converted consistantly by JUCE");
+        }
+
+        {
+            beginTest ("AudioChannelSet documentation is correct");
+
+            for (auto tagEntry : knownTags)
+            {
+                if (tagEntry.equivalentChannelSet.isDisabled())
+                    continue;
+
+                expect (CoreAudioLayouts::fromCoreAudio (tagEntry.tag) == tagEntry.equivalentChannelSet,
+                        String ("Documentation for tag \"") + String (tagEntry.name) + "\" is incorrect");
+            }
+        }
+
+        {
+            beginTest ("CA tag reverse conversion");
+
+            for (auto tagEntry : knownTags)
+            {
+                if (tagEntry.equivalentChannelSet.isDisabled())
+                    continue;
+
+                expect (CoreAudioLayouts::toCoreAudio (tagEntry.equivalentChannelSet) == tagEntry.tag,
+                        String ("Incorrect reverse conversion for tag \"") + String (tagEntry.name) + "\"");
+            }
+        }
+    }
+
+private:
+    struct CoreAudioChannelLayoutTag
+    {
+        AudioChannelLayoutTag tag;
+        const char* name;
+        AudioChannelSet equivalentChannelSet; /* referred to this in the AudioChannelSet documentation */
+    };
+
+    //==============================================================================
+    const Array<CoreAudioChannelLayoutTag>& getAllKnownLayoutTags() const
+    {
+        static CoreAudioChannelLayoutTag tags[] = {
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_Mono,   AudioChannelSet::mono()),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_Stereo, AudioChannelSet::stereo()),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_StereoHeadphones),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_MatrixStereo),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_MidSide),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_XY),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_Binaural),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_Ambisonic_B_Format, AudioChannelSet::ambisonic()),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_Quadraphonic, AudioChannelSet::quadraphonic()),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_Pentagonal, AudioChannelSet::pentagonal()),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_Hexagonal, AudioChannelSet::hexagonal()),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_Octagonal, AudioChannelSet::octagonal()),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_Cube),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_MPEG_1_0),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_MPEG_2_0),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_MPEG_3_0_A, AudioChannelSet::createLCR()),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_MPEG_3_0_B),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_MPEG_4_0_A, AudioChannelSet::createLCRS()),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_MPEG_4_0_B),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_MPEG_5_0_A, AudioChannelSet::create5point0()),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_MPEG_5_0_B),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_MPEG_5_0_C),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_MPEG_5_0_D),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_MPEG_5_1_A, AudioChannelSet::create5point1()),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_MPEG_5_1_B),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_MPEG_5_1_C),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_MPEG_5_1_D),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_MPEG_6_1_A, AudioChannelSet::create6point1()),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_MPEG_7_1_A, AudioChannelSet::create7point1SDDS()),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_MPEG_7_1_B),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_MPEG_7_1_C, AudioChannelSet::create7point1()),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_Emagic_Default_7_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_SMPTE_DTV),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_ITU_1_0),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_ITU_2_0),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_ITU_2_1, AudioChannelSet::createLRS()),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_ITU_2_2),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_ITU_3_0),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_ITU_3_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_ITU_3_2),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_ITU_3_2_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_ITU_3_4_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_0),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_2),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_3),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_4),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_5),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_6),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_7),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_8),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_9),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_10),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_11),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_12),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_13),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_14),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_15),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_16),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_17),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_18),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_19),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DVD_20),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AudioUnit_4),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AudioUnit_5),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AudioUnit_6),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AudioUnit_8),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AudioUnit_5_0),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_AudioUnit_6_0, AudioChannelSet::create6point0()),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_AudioUnit_7_0, AudioChannelSet::create7point0()),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_AudioUnit_7_0_Front, AudioChannelSet::create7point0SDDS()),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AudioUnit_5_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AudioUnit_6_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AudioUnit_7_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AudioUnit_7_1_Front),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AAC_3_0),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AAC_Quadraphonic),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AAC_4_0),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AAC_5_0),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AAC_5_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AAC_6_0),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AAC_6_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AAC_7_0),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AAC_7_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AAC_7_1_B),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AAC_7_1_C),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AAC_Octagonal),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_TMH_10_2_std),
+            // DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_TMH_10_2_full), no indicatoin on how to handle this tag
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AC3_1_0_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AC3_3_0),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AC3_3_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AC3_3_0_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AC3_2_1_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_AC3_3_1_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_EAC_6_0_A),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_EAC_7_0_A),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_EAC3_6_1_A),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_EAC3_6_1_B),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_EAC3_6_1_C),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_EAC3_7_1_A),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_EAC3_7_1_B),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_EAC3_7_1_C),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_EAC3_7_1_D),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_EAC3_7_1_E),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_EAC3_7_1_F),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_EAC3_7_1_G),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_EAC3_7_1_H),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DTS_3_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DTS_4_1),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_DTS_6_0_A, AudioChannelSet::create6point0Music()),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DTS_6_0_B),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DTS_6_0_C),
+            DEFINE_CHANNEL_LAYOUT_TAG_ENTRY (kAudioChannelLayoutTag_DTS_6_1_A, AudioChannelSet::create6point1Music()),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DTS_6_1_B),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DTS_6_1_C),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DTS_7_0),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DTS_7_1),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DTS_8_0_A),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DTS_8_0_B),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DTS_8_1_A),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DTS_8_1_B),
+            DEFINE_CHANNEL_LAYOUT_DFL_ENTRY (kAudioChannelLayoutTag_DTS_6_1_D)
+        };
+        static Array<CoreAudioChannelLayoutTag> knownTags (tags, sizeof (tags) / sizeof (CoreAudioChannelLayoutTag));
+
+        return knownTags;
+    }
+};
+
+static CoreAudioLayoutsUnitTest coreAudioLayoutsUnitTest;
+
+#endif
+
+} // namespace juce
+
+#endif

source/modules/juce_audio_formats/codecs/juce_CoreAudioFormat.h

@@ -0,0 +1,84 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2017 - ROLI Ltd.
+
+   JUCE is an open source library subject to commercial or open-source
+   licensing.
+
+   By using JUCE, you agree to the terms of both the JUCE 5 End-User License
+   Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
+   27th April 2017).
+
+   End User License Agreement: www.juce.com/juce-5-licence
+   Privacy Policy: www.juce.com/juce-5-privacy-policy
+
+   Or: You may also use this code under the terms of the GPL v3 (see
+   www.gnu.org/licenses).
+
+   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
+   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
+   DISCLAIMED.
+
+  ==============================================================================
+*/
+
+namespace juce
+{
+
+#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
+
+} // namespace juce

source/modules/juce_audio_formats/codecs/juce_FlacAudioFormat.cpp

@@ -0,0 +1,633 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2017 - ROLI Ltd.
+
+   JUCE is an open source library subject to commercial or open-source
+   licensing.
+
+   By using JUCE, you agree to the terms of both the JUCE 5 End-User License
+   Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
+   27th April 2017).
+
+   End User License Agreement: www.juce.com/juce-5-licence
+   Privacy Policy: www.juce.com/juce-5-privacy-policy
+
+   Or: You may also use this code under the terms of the GPL v3 (see
+   www.gnu.org/licenses).
+
+   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
+   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
+   DISCLAIMED.
+
+  ==============================================================================
+*/
+
+namespace juce
+{
+
+#if JUCE_USE_FLAC
+
+}
+
+#if defined _WIN32 && !defined __CYGWIN__
+ #include <io.h>
+#else
+ #include <unistd.h>
+#endif
+
+#if defined _MSC_VER || defined __BORLANDC__ || defined __MINGW32__
+ #include <sys/types.h> /* for off_t */
+#endif
+
+#if HAVE_INTTYPES_H
+ #define __STDC_FORMAT_MACROS
+ #include <inttypes.h>
+#endif
+
+#if defined _MSC_VER || defined __MINGW32__ || defined __CYGWIN__ || defined __EMX__
+ #include <io.h> /* for _setmode(), chmod() */
+ #include <fcntl.h> /* for _O_BINARY */
+#else
+ #include <unistd.h> /* for chown(), unlink() */
+#endif
+
+#if defined _MSC_VER || defined __BORLANDC__ || defined __MINGW32__
+ #if defined __BORLANDC__
+  #include <utime.h> /* for utime() */
+ #else
+  #include <sys/utime.h> /* for utime() */
+ #endif
+#else
+ #include <sys/types.h> /* some flavors of BSD (like OS X) require this to get time_t */
+ #include <utime.h> /* for utime() */
+#endif
+
+#if defined _MSC_VER
+ #if _MSC_VER >= 1600
+  #include <stdint.h>
+ #else
+  #include <limits.h>
+ #endif
+#endif
+
+#ifdef _WIN32
+ #include <stdio.h>
+ #include <sys/stat.h>
+ #include <stdarg.h>
+ #include <windows.h>
+#endif
+
+#ifdef DEBUG
+ #include <assert.h>
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+
+namespace juce
+{
+
+namespace FlacNamespace
+{
+#if JUCE_INCLUDE_FLAC_CODE || ! defined (JUCE_INCLUDE_FLAC_CODE)
+
+ #undef VERSION
+ #define VERSION "1.3.1"
+
+ #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)
+ #else
+  #define HAVE_LROUND 1
+ #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"
+  #pragma clang diagnostic ignored "-Wdeprecated-register"
+ #elif JUCE_GCC && (__GNUC__ >= 7)
+  #pragma GCC diagnostic push
+  #pragma GCC diagnostic ignored "-Wimplicit-fallthrough"
+ #endif
+
+ #if JUCE_INTEL
+  #if JUCE_32BIT
+   #define FLAC__CPU_IA32 1
+  #endif
+  #if JUCE_64BIT
+   #define FLAC__CPU_X86_64 1
+  #endif
+  #define FLAC__HAS_X86INTRIN 1
+ #endif
+
+ #undef __STDC_LIMIT_MACROS
+ #define __STDC_LIMIT_MACROS 1
+ #define flac_max jmax
+ #define flac_min jmin
+ #undef DEBUG // (some flac code dumps debug trace if the app defines this macro)
+ #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
+ #elif JUCE_GCC && (__GNUC__ >= 7)
+  #pragma GCC diagnostic pop
+ #endif
+}
+
+#undef max
+#undef min
+
+//==============================================================================
+static const char* const flacFormatName = "FLAC file";
+
+
+//==============================================================================
+class FlacReader  : public AudioFormatReader
+{
+public:
+    FlacReader (InputStream* const in)
+        : AudioFormatReader (in, flacFormatName),
+          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.getReadPointer (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.getWritePointer(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, flacFormatName, rate, numChans, bits),
+          streamStartPos (output != nullptr ? jmax (output->getPosition(), 0ll) : 0ll)
+    {
+        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.get() + 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.get());
+        }
+
+        return FLAC__stream_encoder_process (encoder, (const FLAC__int32**) samplesToWrite, (unsigned) 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 (streamStartPos + 4);
+        ignoreUnused (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;
+    int64 streamStartPos;
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (FlacWriter)
+};
+
+
+//==============================================================================
+FlacAudioFormat::FlacAudioFormat()
+    : AudioFormat (flacFormatName, ".flac")
+{
+}
+
+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 (out != nullptr && 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()
+{
+    static const char* options[] = { "0 (Fastest)", "1", "2", "3", "4", "5 (Default)","6", "7", "8 (Highest quality)", 0 };
+    return StringArray (options);
+}
+
+#endif
+
+} // namespace juce

source/modules/juce_audio_formats/codecs/juce_FlacAudioFormat.h

@@ -0,0 +1,72 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2017 - ROLI Ltd.
+
+   JUCE is an open source library subject to commercial or open-source
+   licensing.
+
+   By using JUCE, you agree to the terms of both the JUCE 5 End-User License
+   Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
+   27th April 2017).
+
+   End User License Agreement: www.juce.com/juce-5-licence
+   Privacy Policy: www.juce.com/juce-5-privacy-policy
+
+   Or: You may also use this code under the terms of the GPL v3 (see
+   www.gnu.org/licenses).
+
+   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
+   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
+   DISCLAIMED.
+
+  ==============================================================================
+*/
+
+namespace juce
+{
+
+#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
+
+} // namespace juce

source/modules/juce_audio_formats/codecs/juce_LAMEEncoderAudioFormat.cpp

@@ -0,0 +1,232 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2017 - ROLI Ltd.
+
+   JUCE is an open source library subject to commercial or open-source
+   licensing.
+
+   By using JUCE, you agree to the terms of both the JUCE 5 End-User License
+   Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
+   27th April 2017).
+
+   End User License Agreement: www.juce.com/juce-5-licence
+   Privacy Policy: www.juce.com/juce-5-privacy-policy
+
+   Or: You may also use this code under the terms of the GPL v3 (see
+   www.gnu.org/licenses).
+
+   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
+   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
+   DISCLAIMED.
+
+  ==============================================================================
+*/
+
+namespace juce
+{
+
+#if JUCE_USE_LAME_AUDIO_FORMAT
+
+class LAMEEncoderAudioFormat::Writer   : public AudioFormatWriter
+{
+public:
+    Writer (OutputStream* destStream, const String& formatName,
+            const File& appFile, int vbr, int cbr,
+            double sampleRate, unsigned int numberOfChannels,
+            int bitsPerSample, const StringPairArray& metadata)
+        : AudioFormatWriter (destStream, formatName, sampleRate,
+                             numberOfChannels, (unsigned int) 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 (appFile.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()));
+
+        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& processArgs) const
+    {
+        ChildProcess cp;
+
+        if (cp.start (processArgs))
+        {
+            const String childOutput (cp.readAllProcessOutput());
+            DBG (childOutput); ignoreUnused (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)
+};
+
+//==============================================================================
+LAMEEncoderAudioFormat::LAMEEncoderAudioFormat (const File& lameApplication)
+    : AudioFormat ("MP3 file", ".mp3"),
+      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()
+{
+    static 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)
+{
+    if (streamToWriteTo == nullptr)
+        return nullptr;
+
+    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
+
+} // namespace juce

source/modules/juce_audio_formats/codecs/juce_LAMEEncoderAudioFormat.h

@@ -0,0 +1,78 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2017 - ROLI Ltd.
+
+   JUCE is an open source library subject to commercial or open-source
+   licensing.
+
+   By using JUCE, you agree to the terms of both the JUCE 5 End-User License
+   Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
+   27th April 2017).
+
+   End User License Agreement: www.juce.com/juce-5-licence
+   Privacy Policy: www.juce.com/juce-5-privacy-policy
+
+   Or: You may also use this code under the terms of the GPL v3 (see
+   www.gnu.org/licenses).
+
+   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
+   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
+   DISCLAIMED.
+
+  ==============================================================================
+*/
+
+namespace juce
+{
+
+#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
+
+} // namespace juce

source/modules/juce_audio_formats/codecs/juce_MP3AudioFormat.h

@@ -0,0 +1,71 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2017 - ROLI Ltd.
+
+   JUCE is an open source library subject to commercial or open-source
+   licensing.
+
+   By using JUCE, you agree to the terms of both the JUCE 5 End-User License
+   Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
+   27th April 2017).
+
+   End User License Agreement: www.juce.com/juce-5-licence
+   Privacy Policy: www.juce.com/juce-5-privacy-policy
+
+   Or: You may also use this code under the terms of the GPL v3 (see
+   www.gnu.org/licenses).
+
+   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
+   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
+   DISCLAIMED.
+
+  ==============================================================================
+*/
+
+namespace juce
+{
+
+#if JUCE_USE_MP3AUDIOFORMAT || DOXYGEN
+
+//==============================================================================
+/**
+    Software-based MP3 decoding format (doesn't currently provide an encoder).
+
+    IMPORTANT DISCLAIMER: By choosing to enable the JUCE_USE_MP3AUDIOFORMAT flag and
+    to compile the MP3 code into your software, you do so AT YOUR OWN RISK! By doing so,
+    you are agreeing that ROLI Ltd. 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 the JUCE_USE_MP3AUDIOFORMAT setting.
+*/
+class MP3AudioFormat  : public AudioFormat
+{
+public:
+    //==============================================================================
+    MP3AudioFormat();
+    ~MP3AudioFormat();
+
+    //==============================================================================
+    Array<int> getPossibleSampleRates() override;
+    Array<int> getPossibleBitDepths() override;
+    bool canDoStereo() override;
+    bool canDoMono() override;
+    bool isCompressed() override;
+    StringArray getQualityOptions() override;
+
+    //==============================================================================
+    AudioFormatReader* createReaderFor (InputStream*, bool deleteStreamIfOpeningFails) override;
+
+    AudioFormatWriter* createWriterFor (OutputStream*, double sampleRateToUse,
+                                        unsigned int numberOfChannels, int bitsPerSample,
+                                        const StringPairArray& metadataValues, int qualityOptionIndex) override;
+};
+
+#endif
+
+} // namespace juce

source/modules/juce_audio_formats/codecs/juce_OggVorbisAudioFormat.cpp

@@ -0,0 +1,550 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2017 - ROLI Ltd.
+
+   JUCE is an open source library subject to commercial or open-source
+   licensing.
+
+   By using JUCE, you agree to the terms of both the JUCE 5 End-User License
+   Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
+   27th April 2017).
+
+   End User License Agreement: www.juce.com/juce-5-licence
+   Privacy Policy: www.juce.com/juce-5-privacy-policy
+
+   Or: You may also use this code under the terms of the GPL v3 (see
+   www.gnu.org/licenses).
+
+   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
+   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
+   DISCLAIMED.
+
+  ==============================================================================
+*/
+
+namespace juce
+{
+
+#if JUCE_USE_OGGVORBIS
+
+#if JUCE_MAC && ! defined (__MACOSX__)
+ #define __MACOSX__ 1
+#endif
+
+namespace OggVorbisNamespace
+{
+#if JUCE_INCLUDE_OGGVORBIS_CODE || ! defined (JUCE_INCLUDE_OGGVORBIS_CODE)
+ #if JUCE_MSVC
+  #pragma warning (push)
+  #pragma warning (disable: 4267 4127 4244 4996 4100 4701 4702 4013 4133 4206 4305 4189 4706 4995 4365 4456 4457 4459)
+ #elif JUCE_CLANG
+  #pragma clang diagnostic push
+  #pragma clang diagnostic ignored "-Wconversion"
+  #pragma clang diagnostic ignored "-Wshadow"
+  #pragma clang diagnostic ignored "-Wdeprecated-register"
+ #elif JUCE_GCC
+  #pragma GCC diagnostic push
+  #pragma GCC diagnostic ignored "-Wshadow"
+  #if (__GNUC__ >= 6)
+   #pragma GCC diagnostic ignored "-Wmisleading-indentation"
+   #if (__GNUC__ >= 7)
+    #pragma GCC diagnostic ignored "-Wimplicit-fallthrough"
+   #endif
+  #endif
+ #endif
+
+ #include "oggvorbis/vorbisenc.h"
+ #include "oggvorbis/codec.h"
+ #include "oggvorbis/vorbisfile.h"
+
+ #include "oggvorbis/bitwise.c"
+ #include "oggvorbis/framing.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/analysis.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/bitrate.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/block.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/codebook.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/envelope.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/floor0.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/floor1.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/info.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/lpc.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/lsp.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/mapping0.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/mdct.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/psy.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/registry.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/res0.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/sharedbook.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/smallft.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/synthesis.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/vorbisenc.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/vorbisfile.c"
+ #include "oggvorbis/libvorbis-1.3.2/lib/window.c"
+
+ #if JUCE_MSVC
+  #pragma warning (pop)
+ #elif JUCE_CLANG
+  #pragma clang diagnostic pop
+ #elif JUCE_GCC
+  #pragma GCC diagnostic pop
+ #endif
+#else
+ #include <vorbis/vorbisenc.h>
+ #include <vorbis/codec.h>
+ #include <vorbis/vorbisfile.h>
+#endif
+}
+
+#undef max
+#undef min
+
+//==============================================================================
+static const char* const oggFormatName = "Ogg-Vorbis file";
+
+const char* const OggVorbisAudioFormat::encoderName = "encoder";
+const char* const OggVorbisAudioFormat::id3title = "id3title";
+const char* const OggVorbisAudioFormat::id3artist = "id3artist";
+const char* const OggVorbisAudioFormat::id3album = "id3album";
+const char* const OggVorbisAudioFormat::id3comment = "id3comment";
+const char* const OggVorbisAudioFormat::id3date = "id3date";
+const char* const OggVorbisAudioFormat::id3genre = "id3genre";
+const char* const OggVorbisAudioFormat::id3trackNumber = "id3trackNumber";
+
+
+//==============================================================================
+class OggReader : public AudioFormatReader
+{
+public:
+    OggReader (InputStream* const inp)
+        : AudioFormatReader (inp, oggFormatName),
+          reservoirStart (0),
+          samplesInReservoir (0)
+    {
+        using namespace OggVorbisNamespace;
+        sampleRate = 0;
+        usesFloatingPointData = true;
+
+        callbacks.read_func  = &oggReadCallback;
+        callbacks.seek_func  = &oggSeekCallback;
+        callbacks.close_func = &oggCloseCallback;
+        callbacks.tell_func  = &oggTellCallback;
+
+        const int err = ov_open_callbacks (input, &ovFile, 0, 0, callbacks);
+
+        if (err == 0)
+        {
+            vorbis_info* info = ov_info (&ovFile, -1);
+
+            vorbis_comment* const comment = ov_comment (&ovFile, -1);
+            addMetadataItem (comment, "ENCODER",     OggVorbisAudioFormat::encoderName);
+            addMetadataItem (comment, "TITLE",       OggVorbisAudioFormat::id3title);
+            addMetadataItem (comment, "ARTIST",      OggVorbisAudioFormat::id3artist);
+            addMetadataItem (comment, "ALBUM",       OggVorbisAudioFormat::id3album);
+            addMetadataItem (comment, "COMMENT",     OggVorbisAudioFormat::id3comment);
+            addMetadataItem (comment, "DATE",        OggVorbisAudioFormat::id3date);
+            addMetadataItem (comment, "GENRE",       OggVorbisAudioFormat::id3genre);
+            addMetadataItem (comment, "TRACKNUMBER", OggVorbisAudioFormat::id3trackNumber);
+
+            lengthInSamples = (uint32) ov_pcm_total (&ovFile, -1);
+            numChannels = (unsigned int) info->channels;
+            bitsPerSample = 16;
+            sampleRate = info->rate;
+
+            reservoir.setSize ((int) numChannels, (int) jmin (lengthInSamples, (int64) 4096));
+        }
+    }
+
+    ~OggReader()
+    {
+        OggVorbisNamespace::ov_clear (&ovFile);
+    }
+
+    void addMetadataItem (OggVorbisNamespace::vorbis_comment* comment, const char* name, const char* metadataName)
+    {
+        if (const char* value = vorbis_comment_query (comment, name, 0))
+            metadataValues.set (metadataName, value);
+    }
+
+    //==============================================================================
+    bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                      int64 startSampleInFile, int numSamples) override
+    {
+        while (numSamples > 0)
+        {
+            const int numAvailable = (int) (reservoirStart + samplesInReservoir - startSampleInFile);
+
+            if (startSampleInFile >= reservoirStart && numAvailable > 0)
+            {
+                // got a few samples overlapping, so use them before seeking..
+
+                const int numToUse = jmin (numSamples, numAvailable);
+
+                for (int i = jmin (numDestChannels, reservoir.getNumChannels()); --i >= 0;)
+                    if (destSamples[i] != nullptr)
+                        memcpy (destSamples[i] + startOffsetInDestBuffer,
+                                reservoir.getReadPointer (i, (int) (startSampleInFile - reservoirStart)),
+                                sizeof (float) * (size_t) numToUse);
+
+                startSampleInFile += numToUse;
+                numSamples -= numToUse;
+                startOffsetInDestBuffer += numToUse;
+
+                if (numSamples == 0)
+                    break;
+            }
+
+            if (startSampleInFile < reservoirStart
+                || startSampleInFile + numSamples > reservoirStart + samplesInReservoir)
+            {
+                // buffer miss, so refill the reservoir
+                int bitStream = 0;
+
+                reservoirStart = jmax (0, (int) startSampleInFile);
+                samplesInReservoir = reservoir.getNumSamples();
+
+                if (reservoirStart != (int) OggVorbisNamespace::ov_pcm_tell (&ovFile))
+                    OggVorbisNamespace::ov_pcm_seek (&ovFile, reservoirStart);
+
+                int offset = 0;
+                int numToRead = samplesInReservoir;
+
+                while (numToRead > 0)
+                {
+                    float** dataIn = nullptr;
+
+                    const long samps = OggVorbisNamespace::ov_read_float (&ovFile, &dataIn, numToRead, &bitStream);
+                    if (samps <= 0)
+                        break;
+
+                    jassert (samps <= numToRead);
+
+                    for (int i = jmin ((int) numChannels, reservoir.getNumChannels()); --i >= 0;)
+                        memcpy (reservoir.getWritePointer (i, offset), dataIn[i], sizeof (float) * (size_t) samps);
+
+                    numToRead -= samps;
+                    offset += samps;
+                }
+
+                if (numToRead > 0)
+                    reservoir.clear (offset, numToRead);
+            }
+        }
+
+        if (numSamples > 0)
+        {
+            for (int i = numDestChannels; --i >= 0;)
+                if (destSamples[i] != nullptr)
+                    zeromem (destSamples[i] + startOffsetInDestBuffer, sizeof (int) * (size_t) numSamples);
+        }
+
+        return true;
+    }
+
+    //==============================================================================
+    static size_t oggReadCallback (void* ptr, size_t size, size_t nmemb, void* datasource)
+    {
+        return (size_t) (static_cast<InputStream*> (datasource)->read (ptr, (int) (size * nmemb))) / size;
+    }
+
+    static int oggSeekCallback (void* datasource, OggVorbisNamespace::ogg_int64_t offset, int whence)
+    {
+        InputStream* const in = static_cast<InputStream*> (datasource);
+
+        if (whence == SEEK_CUR)
+            offset += in->getPosition();
+        else if (whence == SEEK_END)
+            offset += in->getTotalLength();
+
+        in->setPosition (offset);
+        return 0;
+    }
+
+    static int oggCloseCallback (void*)
+    {
+        return 0;
+    }
+
+    static long oggTellCallback (void* datasource)
+    {
+        return (long) static_cast<InputStream*> (datasource)->getPosition();
+    }
+
+private:
+    OggVorbisNamespace::OggVorbis_File ovFile;
+    OggVorbisNamespace::ov_callbacks callbacks;
+    AudioSampleBuffer reservoir;
+    int reservoirStart, samplesInReservoir;
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (OggReader)
+};
+
+//==============================================================================
+class OggWriter  : public AudioFormatWriter
+{
+public:
+    OggWriter (OutputStream* const out,
+               const double sampleRate_,
+               const unsigned int numChannels_,
+               const unsigned int bitsPerSample_,
+               const int qualityIndex,
+               const StringPairArray& metadata)
+        : AudioFormatWriter (out, oggFormatName, sampleRate_, numChannels_, bitsPerSample_),
+          ok (false)
+    {
+        using namespace OggVorbisNamespace;
+
+        vorbis_info_init (&vi);
+
+        if (vorbis_encode_init_vbr (&vi, (int) numChannels_, (int) sampleRate_,
+                                    jlimit (0.0f, 1.0f, qualityIndex * 0.1f)) == 0)
+        {
+            vorbis_comment_init (&vc);
+
+            addMetadata (metadata, OggVorbisAudioFormat::encoderName,    "ENCODER");
+            addMetadata (metadata, OggVorbisAudioFormat::id3title,       "TITLE");
+            addMetadata (metadata, OggVorbisAudioFormat::id3artist,      "ARTIST");
+            addMetadata (metadata, OggVorbisAudioFormat::id3album,       "ALBUM");
+            addMetadata (metadata, OggVorbisAudioFormat::id3comment,     "COMMENT");
+            addMetadata (metadata, OggVorbisAudioFormat::id3date,        "DATE");
+            addMetadata (metadata, OggVorbisAudioFormat::id3genre,       "GENRE");
+            addMetadata (metadata, OggVorbisAudioFormat::id3trackNumber, "TRACKNUMBER");
+
+            vorbis_analysis_init (&vd, &vi);
+            vorbis_block_init (&vd, &vb);
+
+            ogg_stream_init (&os, Random::getSystemRandom().nextInt());
+
+            ogg_packet header;
+            ogg_packet header_comm;
+            ogg_packet header_code;
+
+            vorbis_analysis_headerout (&vd, &vc, &header, &header_comm, &header_code);
+
+            ogg_stream_packetin (&os, &header);
+            ogg_stream_packetin (&os, &header_comm);
+            ogg_stream_packetin (&os, &header_code);
+
+            for (;;)
+            {
+                if (ogg_stream_flush (&os, &og) == 0)
+                    break;
+
+                output->write (og.header, (size_t) og.header_len);
+                output->write (og.body,   (size_t) og.body_len);
+            }
+
+            ok = true;
+        }
+    }
+
+    ~OggWriter()
+    {
+        using namespace OggVorbisNamespace;
+        if (ok)
+        {
+            // write a zero-length packet to show ogg that we're finished..
+            writeSamples (0);
+
+            ogg_stream_clear (&os);
+            vorbis_block_clear (&vb);
+            vorbis_dsp_clear (&vd);
+            vorbis_comment_clear (&vc);
+
+            vorbis_info_clear (&vi);
+            output->flush();
+        }
+        else
+        {
+            vorbis_info_clear (&vi);
+            output = nullptr; // to stop the base class deleting this, as it needs to be returned
+                              // to the caller of createWriter()
+        }
+    }
+
+    //==============================================================================
+    bool write (const int** samplesToWrite, int numSamples) override
+    {
+        if (ok)
+        {
+            using namespace OggVorbisNamespace;
+
+            if (numSamples > 0)
+            {
+                const double gain = 1.0 / 0x80000000u;
+                float** const vorbisBuffer = vorbis_analysis_buffer (&vd, numSamples);
+
+                for (int i = (int) numChannels; --i >= 0;)
+                {
+                    float* const dst = vorbisBuffer[i];
+                    const int* const src = samplesToWrite [i];
+
+                    if (src != nullptr && dst != nullptr)
+                    {
+                        for (int j = 0; j < numSamples; ++j)
+                            dst[j] = (float) (src[j] * gain);
+                    }
+                }
+            }
+
+            writeSamples (numSamples);
+        }
+
+        return ok;
+    }
+
+    void writeSamples (int numSamples)
+    {
+        using namespace OggVorbisNamespace;
+
+        vorbis_analysis_wrote (&vd, numSamples);
+
+        while (vorbis_analysis_blockout (&vd, &vb) == 1)
+        {
+            vorbis_analysis (&vb, 0);
+            vorbis_bitrate_addblock (&vb);
+
+            while (vorbis_bitrate_flushpacket (&vd, &op))
+            {
+                ogg_stream_packetin (&os, &op);
+
+                for (;;)
+                {
+                    if (ogg_stream_pageout (&os, &og) == 0)
+                        break;
+
+                    output->write (og.header, (size_t) og.header_len);
+                    output->write (og.body,   (size_t) og.body_len);
+
+                    if (ogg_page_eos (&og))
+                        break;
+                }
+            }
+        }
+    }
+
+    bool ok;
+
+private:
+    OggVorbisNamespace::ogg_stream_state os;
+    OggVorbisNamespace::ogg_page og;
+    OggVorbisNamespace::ogg_packet op;
+    OggVorbisNamespace::vorbis_info vi;
+    OggVorbisNamespace::vorbis_comment vc;
+    OggVorbisNamespace::vorbis_dsp_state vd;
+    OggVorbisNamespace::vorbis_block vb;
+
+    void addMetadata (const StringPairArray& metadata, const char* name, const char* vorbisName)
+    {
+        const String s (metadata [name]);
+
+        if (s.isNotEmpty())
+            vorbis_comment_add_tag (&vc, vorbisName, const_cast<char*> (s.toRawUTF8()));
+    }
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (OggWriter)
+};
+
+
+//==============================================================================
+OggVorbisAudioFormat::OggVorbisAudioFormat()  : AudioFormat (oggFormatName, ".ogg")
+{
+}
+
+OggVorbisAudioFormat::~OggVorbisAudioFormat()
+{
+}
+
+Array<int> OggVorbisAudioFormat::getPossibleSampleRates()
+{
+    const int rates[] = { 8000, 11025, 12000, 16000, 22050, 32000,
+                          44100, 48000, 88200, 96000, 176400, 192000 };
+
+    return Array<int> (rates, numElementsInArray (rates));
+}
+
+Array<int> OggVorbisAudioFormat::getPossibleBitDepths()
+{
+    const int depths[] = { 32 };
+
+    return Array<int> (depths, numElementsInArray (depths));
+}
+
+bool OggVorbisAudioFormat::canDoStereo()    { return true; }
+bool OggVorbisAudioFormat::canDoMono()      { return true; }
+bool OggVorbisAudioFormat::isCompressed()   { return true; }
+
+AudioFormatReader* OggVorbisAudioFormat::createReaderFor (InputStream* in, const bool deleteStreamIfOpeningFails)
+{
+    ScopedPointer<OggReader> r (new OggReader (in));
+
+    if (r->sampleRate > 0)
+        return r.release();
+
+    if (! deleteStreamIfOpeningFails)
+        r->input = nullptr;
+
+    return nullptr;
+}
+
+AudioFormatWriter* OggVorbisAudioFormat::createWriterFor (OutputStream* out,
+                                                          double sampleRate,
+                                                          unsigned int numChannels,
+                                                          int bitsPerSample,
+                                                          const StringPairArray& metadataValues,
+                                                          int qualityOptionIndex)
+{
+    if (out == nullptr)
+        return nullptr;
+
+    ScopedPointer<OggWriter> w (new OggWriter (out, sampleRate, numChannels,
+                                               (unsigned int) bitsPerSample,
+                                               qualityOptionIndex, metadataValues));
+
+    return w->ok ? w.release() : nullptr;
+}
+
+StringArray OggVorbisAudioFormat::getQualityOptions()
+{
+    static const char* options[] = { "64 kbps", "80 kbps", "96 kbps", "112 kbps", "128 kbps", "160 kbps",
+                                     "192 kbps", "224 kbps", "256 kbps", "320 kbps", "500 kbps", 0 };
+    return StringArray (options);
+}
+
+int OggVorbisAudioFormat::estimateOggFileQuality (const File& source)
+{
+    if (FileInputStream* const in = source.createInputStream())
+    {
+        ScopedPointer<AudioFormatReader> r (createReaderFor (in, true));
+
+        if (r != nullptr)
+        {
+            const double lengthSecs = r->lengthInSamples / r->sampleRate;
+            const int approxBitsPerSecond = (int) (source.getSize() * 8 / lengthSecs);
+
+            const StringArray qualities (getQualityOptions());
+            int bestIndex = 0;
+            int bestDiff = 10000;
+
+            for (int i = qualities.size(); --i >= 0;)
+            {
+                const int diff = std::abs (qualities[i].getIntValue() - approxBitsPerSecond);
+
+                if (diff < bestDiff)
+                {
+                    bestDiff = diff;
+                    bestIndex = i;
+                }
+            }
+
+            return bestIndex;
+        }
+    }
+
+    return 0;
+}
+
+#endif
+
+} // namespace juce

source/modules/juce_audio_formats/codecs/juce_OggVorbisAudioFormat.h

@@ -0,0 +1,100 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2017 - ROLI Ltd.
+
+   JUCE is an open source library subject to commercial or open-source
+   licensing.
+
+   By using JUCE, you agree to the terms of both the JUCE 5 End-User License
+   Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
+   27th April 2017).
+
+   End User License Agreement: www.juce.com/juce-5-licence
+   Privacy Policy: www.juce.com/juce-5-privacy-policy
+
+   Or: You may also use this code under the terms of the GPL v3 (see
+   www.gnu.org/licenses).
+
+   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
+   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
+   DISCLAIMED.
+
+  ==============================================================================
+*/
+
+namespace juce
+{
+
+#if JUCE_USE_OGGVORBIS || defined (DOXYGEN)
+
+//==============================================================================
+/**
+    Reads and writes the Ogg-Vorbis audio format.
+
+    To compile this, you'll need to set the JUCE_USE_OGGVORBIS flag.
+
+    @see AudioFormat,
+*/
+class JUCE_API  OggVorbisAudioFormat  : public AudioFormat
+{
+public:
+    //==============================================================================
+    OggVorbisAudioFormat();
+    ~OggVorbisAudioFormat();
+
+    //==============================================================================
+    Array<int> getPossibleSampleRates() override;
+    Array<int> getPossibleBitDepths() override;
+    bool canDoStereo() override;
+    bool canDoMono() override;
+    bool isCompressed() override;
+    StringArray getQualityOptions() override;
+
+    //==============================================================================
+    /** Tries to estimate the quality level of an ogg file based on its size.
+
+        If it can't read the file for some reason, this will just return 1 (medium quality),
+        otherwise it will return the approximate quality setting that would have been used
+        to create the file.
+
+        @see getQualityOptions
+    */
+    int estimateOggFileQuality (const File& source);
+
+    //==============================================================================
+    /** Metadata property name used by the Ogg writer - if you set a string for this
+        value, it will be written into the ogg file as the name of the encoder app.
+
+        @see createWriterFor
+    */
+    static const char* const encoderName;
+
+    static const char* const id3title;          /**< Metadata key for setting an ID3 title. */
+    static const char* const id3artist;         /**< Metadata key for setting an ID3 artist name. */
+    static const char* const id3album;          /**< Metadata key for setting an ID3 album. */
+    static const char* const id3comment;        /**< Metadata key for setting an ID3 comment. */
+    static const char* const id3date;           /**< Metadata key for setting an ID3 date. */
+    static const char* const id3genre;          /**< Metadata key for setting an ID3 genre. */
+    static const char* const id3trackNumber;    /**< Metadata key for setting an ID3 track number. */
+
+    //==============================================================================
+    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 (OggVorbisAudioFormat)
+};
+
+
+#endif
+
+} // namespace juce

source/modules/juce_audio_formats/codecs/juce_WavAudioFormat.h

@@ -0,0 +1,225 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2017 - ROLI Ltd.
+
+   JUCE is an open source library subject to commercial or open-source
+   licensing.
+
+   By using JUCE, you agree to the terms of both the JUCE 5 End-User License
+   Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
+   27th April 2017).
+
+   End User License Agreement: www.juce.com/juce-5-licence
+   Privacy Policy: www.juce.com/juce-5-privacy-policy
+
+   Or: You may also use this code under the terms of the GPL v3 (see
+   www.gnu.org/licenses).
+
+   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
+   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
+   DISCLAIMED.
+
+  ==============================================================================
+*/
+
+namespace juce
+{
+
+//==============================================================================
+/**
+    Reads and Writes WAV format audio files.
+
+    @see AudioFormat
+*/
+class JUCE_API  WavAudioFormat  : public AudioFormat
+{
+public:
+    //==============================================================================
+    /** Creates a format object. */
+    WavAudioFormat();
+
+    /** Destructor. */
+    ~WavAudioFormat();
+
+    //==============================================================================
+    // BWAV chunk properties:
+
+    static const char* const bwavDescription;       /**< Metadata property name used in BWAV chunks. */
+    static const char* const bwavOriginator;        /**< Metadata property name used in BWAV chunks. */
+    static const char* const bwavOriginatorRef;     /**< Metadata property name used in BWAV chunks. */
+    static const char* const bwavOriginationDate;   /**< Metadata property name used in BWAV chunks. The format should be: yyyy-mm-dd */
+    static const char* const bwavOriginationTime;   /**< Metadata property name used in BWAV chunks. The format should be: format is: hh-mm-ss */
+    static const char* const bwavCodingHistory;     /**< Metadata property name used in BWAV chunks. */
+
+    /** Metadata property name used in BWAV chunks.
+        This is the number of samples from the start of an edit that the
+        file is supposed to begin at. Seems like an obvious mistake to
+        only allow a file to occur in an edit once, but that's the way
+        it is..
+
+        @see AudioFormatReader::metadataValues, createWriterFor
+    */
+    static const char* const bwavTimeReference;
+
+    /** Utility function to fill out the appropriate metadata for a BWAV file.
+
+        This just makes it easier than using the property names directly, and it
+        fills out the time and date in the right format.
+    */
+    static StringPairArray createBWAVMetadata (const String& description,
+                                               const String& originator,
+                                               const String& originatorRef,
+                                               Time dateAndTime,
+                                               int64 timeReferenceSamples,
+                                               const String& codingHistory);
+
+    //==============================================================================
+    // 'acid' chunk properties:
+
+    static const char* const acidOneShot;           /**< Metadata property name used in acid chunks. */
+    static const char* const acidRootSet;           /**< Metadata property name used in acid chunks. */
+    static const char* const acidStretch;           /**< Metadata property name used in acid chunks. */
+    static const char* const acidDiskBased;         /**< Metadata property name used in acid chunks. */
+    static const char* const acidizerFlag;          /**< Metadata property name used in acid chunks. */
+    static const char* const acidRootNote;          /**< Metadata property name used in acid chunks. */
+    static const char* const acidBeats;             /**< Metadata property name used in acid chunks. */
+    static const char* const acidDenominator;       /**< Metadata property name used in acid chunks. */
+    static const char* const acidNumerator;         /**< Metadata property name used in acid chunks. */
+    static const char* const acidTempo;             /**< Metadata property name used in acid chunks. */
+
+    //==============================================================================
+    // INFO chunk properties:
+
+    static const char* const riffInfoArchivalLocation;      /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoArtist;                /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoBaseURL;               /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoCinematographer;       /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoComment;               /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoComment2;              /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoComments;              /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoCommissioned;          /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoCopyright;             /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoCostumeDesigner;       /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoCountry;               /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoCropped;               /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoDateCreated;           /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoDateTimeOriginal;      /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoDefaultAudioStream;    /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoDimension;             /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoDirectory;             /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoDistributedBy;         /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoDotsPerInch;           /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoEditedBy;              /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoEighthLanguage;        /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoEncodedBy;             /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoEndTimecode;           /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoEngineer;              /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoFifthLanguage;         /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoFirstLanguage;         /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoFourthLanguage;        /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoGenre;                 /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoKeywords;              /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoLanguage;              /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoLength;                /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoLightness;             /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoLocation;              /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoLogoIconURL;           /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoLogoURL;               /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoMedium;                /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoMoreInfoBannerImage;   /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoMoreInfoBannerURL;     /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoMoreInfoText;          /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoMoreInfoURL;           /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoMusicBy;               /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoNinthLanguage;         /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoNumberOfParts;         /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoOrganisation;          /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoPart;                  /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoProducedBy;            /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoProductName;           /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoProductionDesigner;    /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoProductionStudio;      /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoRate;                  /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoRated;                 /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoRating;                /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoRippedBy;              /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoSecondaryGenre;        /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoSecondLanguage;        /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoSeventhLanguage;       /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoSharpness;             /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoSixthLanguage;         /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoSoftware;              /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoSoundSchemeTitle;      /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoSource;                /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoSourceFrom;            /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoStarring_ISTR;         /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoStarring_STAR;         /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoStartTimecode;         /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoStatistics;            /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoSubject;               /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoTapeName;              /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoTechnician;            /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoThirdLanguage;         /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoTimeCode;              /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoTitle;                 /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoTrackNo;               /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoTrackNumber;           /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoURL;                   /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoVegasVersionMajor;     /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoVegasVersionMinor;     /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoVersion;               /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoWatermarkURL;          /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoWrittenBy;             /**< Metadata property name used in INFO chunks. */
+    static const char* const riffInfoYear;                  /**< Metadata property name used in INFO chunks. */
+
+    //==============================================================================
+    /** Metadata property name used when reading an ISRC code from an AXML chunk. */
+    static const char* const ISRC;
+
+    /** Metadata property name used when reading a WAV file with a Tracktion chunk. */
+    static const char* const tracktionLoopInfo;
+
+    //==============================================================================
+    Array<int> getPossibleSampleRates() override;
+    Array<int> getPossibleBitDepths() override;
+    bool canDoStereo() override;
+    bool canDoMono() override;
+    bool isChannelLayoutSupported (const AudioChannelSet& channelSet) override;
+
+    //==============================================================================
+    AudioFormatReader* createReaderFor (InputStream* sourceStream,
+                                        bool deleteStreamIfOpeningFails) override;
+
+    MemoryMappedAudioFormatReader* createMemoryMappedReader (const File&)      override;
+    MemoryMappedAudioFormatReader* createMemoryMappedReader (FileInputStream*) override;
+
+    AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
+                                        double sampleRateToUse,
+                                        unsigned int numberOfChannels,
+                                        int bitsPerSample,
+                                        const StringPairArray& metadataValues,
+                                        int qualityOptionIndex) override;
+
+    AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
+                                        double sampleRateToUse,
+                                        const AudioChannelSet& channelLayout,
+                                        int bitsPerSample,
+                                        const StringPairArray& metadataValues,
+                                        int qualityOptionIndex) override;
+
+    //==============================================================================
+    /** Utility function to replace the metadata in a wav file with a new set of values.
+
+        If possible, this cheats by overwriting just the metadata region of the file, rather
+        than by copying the whole file again.
+    */
+    bool replaceMetadataInFile (const File& wavFile, const StringPairArray& newMetadata);
+
+
+private:
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (WavAudioFormat)
+};
+
+} // namespace juce

source/modules/juce_audio_formats/codecs/juce_WindowsMediaAudioFormat.cpp

@@ -0,0 +1,360 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2017 - ROLI Ltd.
+
+   JUCE is an open source library subject to commercial or open-source
+   licensing.
+
+   By using JUCE, you agree to the terms of both the JUCE 5 End-User License
+   Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
+   27th April 2017).
+
+   End User License Agreement: www.juce.com/juce-5-licence
+   Privacy Policy: www.juce.com/juce-5-privacy-policy
+
+   Or: You may also use this code under the terms of the GPL v3 (see
+   www.gnu.org/licenses).
+
+   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
+   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
+   DISCLAIMED.
+
+  ==============================================================================
+*/
+
+namespace juce
+{
+
+namespace WindowsMediaCodec
+{
+
+class JuceIStream   : public ComBaseClassHelper <IStream>
+{
+public:
+    JuceIStream (InputStream& in) noexcept
+        : ComBaseClassHelper <IStream> (0), source (in)
+    {
+    }
+
+    JUCE_COMRESULT Commit (DWORD)                        { return S_OK; }
+    JUCE_COMRESULT Write (const void*, ULONG, ULONG*)    { return E_NOTIMPL; }
+    JUCE_COMRESULT Clone (IStream**)                     { return E_NOTIMPL; }
+    JUCE_COMRESULT SetSize (ULARGE_INTEGER)              { return E_NOTIMPL; }
+    JUCE_COMRESULT Revert()                              { return E_NOTIMPL; }
+    JUCE_COMRESULT LockRegion (ULARGE_INTEGER, ULARGE_INTEGER, DWORD)    { return E_NOTIMPL; }
+    JUCE_COMRESULT UnlockRegion (ULARGE_INTEGER, ULARGE_INTEGER, DWORD)  { return E_NOTIMPL; }
+
+    JUCE_COMRESULT Read (void* dest, ULONG numBytes, ULONG* bytesRead)
+    {
+        const int numRead = source.read (dest, numBytes);
+
+        if (bytesRead != nullptr)
+            *bytesRead = numRead;
+
+        return (numRead == (int) numBytes) ? S_OK : S_FALSE;
+    }
+
+    JUCE_COMRESULT Seek (LARGE_INTEGER position, DWORD origin, ULARGE_INTEGER* resultPosition)
+    {
+        int64 newPos = (int64) position.QuadPart;
+
+        if (origin == STREAM_SEEK_CUR)
+        {
+            newPos += source.getPosition();
+        }
+        else if (origin == STREAM_SEEK_END)
+        {
+            const int64 len = source.getTotalLength();
+            if (len < 0)
+                return E_NOTIMPL;
+
+            newPos += len;
+        }
+
+        if (resultPosition != nullptr)
+            resultPosition->QuadPart = newPos;
+
+        return source.setPosition (newPos) ? S_OK : E_NOTIMPL;
+    }
+
+    JUCE_COMRESULT CopyTo (IStream* destStream, ULARGE_INTEGER numBytesToDo,
+                           ULARGE_INTEGER* bytesRead, ULARGE_INTEGER* bytesWritten)
+    {
+        uint64 totalCopied = 0;
+        int64 numBytes = numBytesToDo.QuadPart;
+
+        while (numBytes > 0 && ! source.isExhausted())
+        {
+            char buffer [1024];
+
+            const int numToCopy = (int) jmin ((int64) sizeof (buffer), (int64) numBytes);
+            const int numRead = source.read (buffer, numToCopy);
+
+            if (numRead <= 0)
+                break;
+
+            destStream->Write (buffer, numRead, nullptr);
+            totalCopied += numRead;
+        }
+
+        if (bytesRead != nullptr)      bytesRead->QuadPart = totalCopied;
+        if (bytesWritten != nullptr)   bytesWritten->QuadPart = totalCopied;
+
+        return S_OK;
+    }
+
+    JUCE_COMRESULT Stat (STATSTG* stat, DWORD)
+    {
+        if (stat == nullptr)
+            return STG_E_INVALIDPOINTER;
+
+        zerostruct (*stat);
+        stat->type = STGTY_STREAM;
+        stat->cbSize.QuadPart = jmax ((int64) 0, source.getTotalLength());
+        return S_OK;
+    }
+
+private:
+    InputStream& source;
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (JuceIStream)
+};
+
+//==============================================================================
+static const char* wmFormatName = "Windows Media";
+static const char* const extensions[] = { ".mp3", ".wmv", ".asf", ".wm", ".wma", 0 };
+
+//==============================================================================
+class WMAudioReader   : public AudioFormatReader
+{
+public:
+    WMAudioReader (InputStream* const input_)
+        : AudioFormatReader (input_, TRANS (wmFormatName)),
+          wmvCoreLib ("Wmvcore.dll")
+    {
+        JUCE_LOAD_WINAPI_FUNCTION (wmvCoreLib, WMCreateSyncReader, wmCreateSyncReader,
+                                   HRESULT, (IUnknown*, DWORD, IWMSyncReader**))
+
+        if (wmCreateSyncReader != nullptr)
+        {
+            checkCoInitialiseCalled();
+
+            HRESULT hr = wmCreateSyncReader (nullptr, WMT_RIGHT_PLAYBACK, wmSyncReader.resetAndGetPointerAddress());
+
+            if (SUCCEEDED (hr))
+                hr = wmSyncReader->OpenStream (new JuceIStream (*input));
+
+            if (SUCCEEDED (hr))
+            {
+                WORD streamNum = 1;
+                hr = wmSyncReader->GetStreamNumberForOutput (0, &streamNum);
+                hr = wmSyncReader->SetReadStreamSamples (streamNum, false);
+
+                scanFileForDetails();
+            }
+        }
+    }
+
+    ~WMAudioReader()
+    {
+        if (wmSyncReader != nullptr)
+            wmSyncReader->Close();
+    }
+
+    bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                      int64 startSampleInFile, int numSamples) override
+    {
+        if (sampleRate <= 0)
+            return false;
+
+        checkCoInitialiseCalled();
+
+        clearSamplesBeyondAvailableLength (destSamples, numDestChannels, startOffsetInDestBuffer,
+                                           startSampleInFile, numSamples, lengthInSamples);
+
+        const int stride = numChannels * sizeof (int16);
+
+        while (numSamples > 0)
+        {
+            if (! bufferedRange.contains (startSampleInFile))
+            {
+                const bool hasJumped = (startSampleInFile != bufferedRange.getEnd());
+
+                if (hasJumped)
+                    wmSyncReader->SetRange ((QWORD) (startSampleInFile * 10000000 / (int64) sampleRate), 0);
+
+                ComSmartPtr<INSSBuffer> sampleBuffer;
+                QWORD sampleTime, duration;
+                DWORD flags, outputNum;
+                WORD streamNum;
+
+                HRESULT hr = wmSyncReader->GetNextSample (1, sampleBuffer.resetAndGetPointerAddress(),
+                                                          &sampleTime, &duration, &flags, &outputNum, &streamNum);
+
+                if (sampleBuffer != nullptr)
+                {
+                    BYTE* rawData = nullptr;
+                    DWORD dataLength = 0;
+                    hr = sampleBuffer->GetBufferAndLength (&rawData, &dataLength);
+
+                    if (dataLength == 0)
+                        return false;
+
+                    if (hasJumped)
+                        bufferedRange.setStart ((int64) ((sampleTime * (int64) sampleRate) / 10000000));
+                    else
+                        bufferedRange.setStart (bufferedRange.getEnd()); // (because the positions returned often aren't continguous)
+
+                    bufferedRange.setLength ((int64) (dataLength / stride));
+
+                    buffer.ensureSize ((int) dataLength);
+                    memcpy (buffer.getData(), rawData, (size_t) dataLength);
+                }
+                else if (hr == NS_E_NO_MORE_SAMPLES)
+                {
+                    bufferedRange.setStart (startSampleInFile);
+                    bufferedRange.setLength (256);
+                    buffer.ensureSize (256 * stride);
+                    buffer.fillWith (0);
+                }
+                else
+                {
+                    return false;
+                }
+            }
+
+            const int offsetInBuffer = (int) (startSampleInFile - bufferedRange.getStart());
+            const int16* const rawData = static_cast<const int16*> (addBytesToPointer (buffer.getData(), offsetInBuffer * stride));
+            const int numToDo = jmin (numSamples, (int) (bufferedRange.getLength() - offsetInBuffer));
+
+            for (int i = 0; i < numDestChannels; ++i)
+            {
+                jassert (destSamples[i] != nullptr);
+
+                const int srcChan = jmin (i, (int) numChannels - 1);
+                const int16* src = rawData + srcChan;
+                int* const dst = destSamples[i] + startOffsetInDestBuffer;
+
+                for (int j = 0; j < numToDo; ++j)
+                {
+                    dst[j] = ((uint32) *src) << 16;
+                    src += numChannels;
+                }
+            }
+
+            startSampleInFile += numToDo;
+            startOffsetInDestBuffer += numToDo;
+            numSamples -= numToDo;
+        }
+
+        return true;
+    }
+
+private:
+    DynamicLibrary wmvCoreLib;
+    ComSmartPtr<IWMSyncReader> wmSyncReader;
+    MemoryBlock buffer;
+    Range<int64> bufferedRange;
+
+    void checkCoInitialiseCalled()
+    {
+        CoInitialize (0);
+    }
+
+    void scanFileForDetails()
+    {
+        ComSmartPtr<IWMHeaderInfo> wmHeaderInfo;
+        HRESULT hr = wmSyncReader.QueryInterface (wmHeaderInfo);
+
+        if (SUCCEEDED (hr))
+        {
+            QWORD lengthInNanoseconds = 0;
+            WORD lengthOfLength = sizeof (lengthInNanoseconds);
+            WORD streamNum = 0;
+            WMT_ATTR_DATATYPE wmAttrDataType;
+            hr = wmHeaderInfo->GetAttributeByName (&streamNum, L"Duration", &wmAttrDataType,
+                                                   (BYTE*) &lengthInNanoseconds, &lengthOfLength);
+
+            ComSmartPtr<IWMProfile> wmProfile;
+            hr = wmSyncReader.QueryInterface (wmProfile);
+
+            if (SUCCEEDED (hr))
+            {
+                ComSmartPtr<IWMStreamConfig> wmStreamConfig;
+                hr = wmProfile->GetStream (0, wmStreamConfig.resetAndGetPointerAddress());
+
+                if (SUCCEEDED (hr))
+                {
+                    ComSmartPtr<IWMMediaProps> wmMediaProperties;
+                    hr = wmStreamConfig.QueryInterface (wmMediaProperties);
+
+                    if (SUCCEEDED (hr))
+                    {
+                        DWORD sizeMediaType;
+                        hr = wmMediaProperties->GetMediaType (0, &sizeMediaType);
+
+                        HeapBlock<WM_MEDIA_TYPE> mediaType;
+                        mediaType.malloc (sizeMediaType, 1);
+                        hr = wmMediaProperties->GetMediaType (mediaType, &sizeMediaType);
+
+                        if (mediaType->majortype == WMMEDIATYPE_Audio)
+                        {
+                            const WAVEFORMATEX* const inputFormat = reinterpret_cast<WAVEFORMATEX*> (mediaType->pbFormat);
+
+                            sampleRate = inputFormat->nSamplesPerSec;
+                            numChannels = inputFormat->nChannels;
+                            bitsPerSample = inputFormat->wBitsPerSample != 0 ? inputFormat->wBitsPerSample : 16;
+                            lengthInSamples = (lengthInNanoseconds * (int) sampleRate) / 10000000;
+                        }
+                    }
+                }
+            }
+        }
+    }
+
+    JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (WMAudioReader)
+};
+
+}
+
+//==============================================================================
+WindowsMediaAudioFormat::WindowsMediaAudioFormat()
+    : AudioFormat (TRANS (WindowsMediaCodec::wmFormatName),
+                   StringArray (WindowsMediaCodec::extensions))
+{
+}
+
+WindowsMediaAudioFormat::~WindowsMediaAudioFormat() {}
+
+Array<int> WindowsMediaAudioFormat::getPossibleSampleRates()    { return Array<int>(); }
+Array<int> WindowsMediaAudioFormat::getPossibleBitDepths()      { return Array<int>(); }
+
+bool WindowsMediaAudioFormat::canDoStereo()     { return true; }
+bool WindowsMediaAudioFormat::canDoMono()       { return true; }
+bool WindowsMediaAudioFormat::isCompressed()    { return true; }
+
+//==============================================================================
+AudioFormatReader* WindowsMediaAudioFormat::createReaderFor (InputStream* sourceStream, bool deleteStreamIfOpeningFails)
+{
+    ScopedPointer<WindowsMediaCodec::WMAudioReader> r (new WindowsMediaCodec::WMAudioReader (sourceStream));
+
+    if (r->sampleRate > 0)
+        return r.release();
+
+    if (! deleteStreamIfOpeningFails)
+        r->input = nullptr;
+
+    return nullptr;
+}
+
+AudioFormatWriter* WindowsMediaAudioFormat::createWriterFor (OutputStream* /*streamToWriteTo*/, double /*sampleRateToUse*/,
+                                                             unsigned int /*numberOfChannels*/, int /*bitsPerSample*/,
+                                                             const StringPairArray& /*metadataValues*/, int /*qualityOptionIndex*/)
+{
+    jassertfalse; // not yet implemented!
+    return nullptr;
+}
+
+} // namespace juce

source/modules/juce_audio_formats/codecs/juce_WindowsMediaAudioFormat.h

@@ -0,0 +1,60 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library.
+   Copyright (c) 2017 - ROLI Ltd.
+
+   JUCE is an open source library subject to commercial or open-source
+   licensing.
+
+   By using JUCE, you agree to the terms of both the JUCE 5 End-User License
+   Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
+   27th April 2017).
+
+   End User License Agreement: www.juce.com/juce-5-licence
+   Privacy Policy: www.juce.com/juce-5-privacy-policy
+
+   Or: You may also use this code under the terms of the GPL v3 (see
+   www.gnu.org/licenses).
+
+   JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
+   EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
+   DISCLAIMED.
+
+  ==============================================================================
+*/
+
+namespace juce
+{
+
+#if JUCE_WINDOWS || DOXYGEN
+
+//==============================================================================
+/**
+    Audio format which uses the Windows Media codecs (Windows only).
+*/
+class WindowsMediaAudioFormat  : public AudioFormat
+{
+public:
+    //==============================================================================
+    WindowsMediaAudioFormat();
+    ~WindowsMediaAudioFormat();
+
+    //==============================================================================
+    Array<int> getPossibleSampleRates() override;
+    Array<int> getPossibleBitDepths() override;
+    bool canDoStereo() override;
+    bool canDoMono() override;
+    bool isCompressed() override;
+
+    //==============================================================================
+    AudioFormatReader* createReaderFor (InputStream*, bool deleteStreamIfOpeningFails) override;
+
+    AudioFormatWriter* createWriterFor (OutputStream*, double sampleRateToUse,
+                                        unsigned int numberOfChannels, int bitsPerSample,
+                                        const StringPairArray& metadataValues, int qualityOptionIndex) override;
+};
+
+#endif
+
+}

source/modules/juce_audio_formats/codecs/oggvorbis/Ogg Vorbis Licence.txt

@@ -0,0 +1,47 @@
+=====================================================================
+
+I've incorporated Ogg-Vorbis 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 Ogg-Vorbis code - just tweaked a few include 
+paths to make it build smoothly, and added some headers to allow you to exclude 
+it from the build.
+
+=====================================================================
+
+The following license is the BSD-style license that comes with the
+Ogg-Vorbis distribution, and which applies just to the header files I've
+included in this directory. For more info, and to get the rest of the
+distribution, visit the Ogg-Vorbis homepage: www.vorbis.com
+
+=====================================================================
+
+Copyright (c) 2002-2004 Xiph.org Foundation
+
+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/oggvorbis/bitwise.c

@@ -0,0 +1,788 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2002             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+  function: packing variable sized words into an octet stream
+  last mod: $Id: bitwise.c,v 1.1 2007/06/07 17:48:18 jules_rms Exp $
+
+ ********************************************************************/
+
+#ifdef JUCE_MSVC
+ #pragma warning (disable: 4456 4457 4459)
+#endif
+
+/* We're 'LSb' endian; if we write a word but read individual bits,
+   then we'll read the lsb first */
+
+#include <string.h>
+#include <stdlib.h>
+#include "ogg.h"
+
+#define BUFFER_INCREMENT 256
+
+static const unsigned long mask[]=
+{0x00000000,0x00000001,0x00000003,0x00000007,0x0000000f,
+ 0x0000001f,0x0000003f,0x0000007f,0x000000ff,0x000001ff,
+ 0x000003ff,0x000007ff,0x00000fff,0x00001fff,0x00003fff,
+ 0x00007fff,0x0000ffff,0x0001ffff,0x0003ffff,0x0007ffff,
+ 0x000fffff,0x001fffff,0x003fffff,0x007fffff,0x00ffffff,
+ 0x01ffffff,0x03ffffff,0x07ffffff,0x0fffffff,0x1fffffff,
+ 0x3fffffff,0x7fffffff,0xffffffff };
+
+static const unsigned int mask8B[]=
+{0x00,0x80,0xc0,0xe0,0xf0,0xf8,0xfc,0xfe,0xff};
+
+void oggpack_writeinit(oggpack_buffer *b){
+  memset(b,0,sizeof(*b));
+  b->ptr=b->buffer=(unsigned char*) _ogg_malloc(BUFFER_INCREMENT);
+  b->buffer[0]='\0';
+  b->storage=BUFFER_INCREMENT;
+}
+
+void oggpackB_writeinit(oggpack_buffer *b){
+  oggpack_writeinit(b);
+}
+
+void oggpack_writetrunc(oggpack_buffer *b,long bits){
+  long bytes=bits>>3;
+  bits-=bytes*8;
+  b->ptr=b->buffer+bytes;
+  b->endbit=bits;
+  b->endbyte=bytes;
+  *b->ptr&=mask[bits];
+}
+
+void oggpackB_writetrunc(oggpack_buffer *b,long bits){
+  long bytes=bits>>3;
+  bits-=bytes*8;
+  b->ptr=b->buffer+bytes;
+  b->endbit=bits;
+  b->endbyte=bytes;
+  *b->ptr&=mask8B[bits];
+}
+
+/* Takes only up to 32 bits. */
+void oggpack_write(oggpack_buffer *b,unsigned long value,int bits){
+  if(b->endbyte+4>=b->storage){
+    b->buffer=(unsigned char*) _ogg_realloc(b->buffer,b->storage+BUFFER_INCREMENT);
+    b->storage+=BUFFER_INCREMENT;
+    b->ptr=b->buffer+b->endbyte;
+  }
+
+  value&=mask[bits];
+  bits+=b->endbit;
+
+  b->ptr[0]|=value<<b->endbit;
+
+  if(bits>=8){
+    b->ptr[1]=(unsigned char)(value>>(8-b->endbit));
+    if(bits>=16){
+      b->ptr[2]=(unsigned char)(value>>(16-b->endbit));
+      if(bits>=24){
+	b->ptr[3]=(unsigned char)(value>>(24-b->endbit));
+	if(bits>=32){
+	  if(b->endbit)
+	    b->ptr[4]=(unsigned char)(value>>(32-b->endbit));
+	  else
+	    b->ptr[4]=0;
+	}
+      }
+    }
+  }
+
+  b->endbyte+=bits/8;
+  b->ptr+=bits/8;
+  b->endbit=bits&7;
+}
+
+/* Takes only up to 32 bits. */
+void oggpackB_write(oggpack_buffer *b,unsigned long value,int bits){
+  if(b->endbyte+4>=b->storage){
+    b->buffer=(unsigned char*) _ogg_realloc(b->buffer,b->storage+BUFFER_INCREMENT);
+    b->storage+=BUFFER_INCREMENT;
+    b->ptr=b->buffer+b->endbyte;
+  }
+
+  value=(value&mask[bits])<<(32-bits);
+  bits+=b->endbit;
+
+  b->ptr[0]|=value>>(24+b->endbit);
+
+  if(bits>=8){
+    b->ptr[1]=(unsigned char)(value>>(16+b->endbit));
+    if(bits>=16){
+      b->ptr[2]=(unsigned char)(value>>(8+b->endbit));
+      if(bits>=24){
+	b->ptr[3]=(unsigned char)(value>>(b->endbit));
+	if(bits>=32){
+	  if(b->endbit)
+	    b->ptr[4]=(unsigned char)(value<<(8-b->endbit));
+	  else
+	    b->ptr[4]=0;
+	}
+      }
+    }
+  }
+
+  b->endbyte+=bits/8;
+  b->ptr+=bits/8;
+  b->endbit=bits&7;
+}
+
+void oggpack_writealign(oggpack_buffer *b){
+  int bits=8-b->endbit;
+  if(bits<8)
+    oggpack_write(b,0,bits);
+}
+
+void oggpackB_writealign(oggpack_buffer *b){
+  int bits=8-b->endbit;
+  if(bits<8)
+    oggpackB_write(b,0,bits);
+}
+
+static void oggpack_writecopy_helper(oggpack_buffer *b,
+				     void *source,
+				     long bits,
+				     void (*w)(oggpack_buffer *,
+					       unsigned long,
+					       int),
+				     int msb){
+  unsigned char *ptr=(unsigned char *)source;
+
+  long bytes=bits/8;
+  bits-=bytes*8;
+
+  if(b->endbit){
+    int i;
+    /* unaligned copy.  Do it the hard way. */
+    for(i=0;i<bytes;i++)
+      w(b,(unsigned long)(ptr[i]),8);
+  }else{
+    /* aligned block copy */
+    if(b->endbyte+bytes+1>=b->storage){
+      b->storage=b->endbyte+bytes+BUFFER_INCREMENT;
+      b->buffer=(unsigned char*) _ogg_realloc(b->buffer,b->storage);
+      b->ptr=b->buffer+b->endbyte;
+    }
+
+    memmove(b->ptr,source,bytes);
+    b->ptr+=bytes;
+    b->endbyte+=bytes;
+    *b->ptr=0;
+
+  }
+  if(bits){
+    if(msb)
+      w(b,(unsigned long)(ptr[bytes]>>(8-bits)),bits);
+    else
+      w(b,(unsigned long)(ptr[bytes]),bits);
+  }
+}
+
+void oggpack_writecopy(oggpack_buffer *b,void *source,long bits){
+  oggpack_writecopy_helper(b,source,bits,oggpack_write,0);
+}
+
+void oggpackB_writecopy(oggpack_buffer *b,void *source,long bits){
+  oggpack_writecopy_helper(b,source,bits,oggpackB_write,1);
+}
+
+void oggpack_reset(oggpack_buffer *b){
+  b->ptr=b->buffer;
+  b->buffer[0]=0;
+  b->endbit=b->endbyte=0;
+}
+
+void oggpackB_reset(oggpack_buffer *b){
+  oggpack_reset(b);
+}
+
+void oggpack_writeclear(oggpack_buffer *b){
+  _ogg_free(b->buffer);
+  memset(b,0,sizeof(*b));
+}
+
+void oggpackB_writeclear(oggpack_buffer *b){
+  oggpack_writeclear(b);
+}
+
+void oggpack_readinit(oggpack_buffer *b,unsigned char *buf,int bytes){
+  memset(b,0,sizeof(*b));
+  b->buffer=b->ptr=buf;
+  b->storage=bytes;
+}
+
+void oggpackB_readinit(oggpack_buffer *b,unsigned char *buf,int bytes){
+  oggpack_readinit(b,buf,bytes);
+}
+
+/* Read in bits without advancing the bitptr; bits <= 32 */
+long oggpack_look(oggpack_buffer *b,int bits){
+  unsigned long ret;
+  unsigned long m=mask[bits];
+
+  bits+=b->endbit;
+
+  if(b->endbyte+4>=b->storage){
+    /* not the main path */
+    if(b->endbyte*8+bits>b->storage*8)return(-1);
+  }
+
+  ret=b->ptr[0]>>b->endbit;
+  if(bits>8){
+    ret|=b->ptr[1]<<(8-b->endbit);
+    if(bits>16){
+      ret|=b->ptr[2]<<(16-b->endbit);
+      if(bits>24){
+	ret|=b->ptr[3]<<(24-b->endbit);
+	if(bits>32 && b->endbit)
+	  ret|=b->ptr[4]<<(32-b->endbit);
+      }
+    }
+  }
+  return(m&ret);
+}
+
+/* Read in bits without advancing the bitptr; bits <= 32 */
+long oggpackB_look(oggpack_buffer *b,int bits){
+  unsigned long ret;
+  int m=32-bits;
+
+  bits+=b->endbit;
+
+  if(b->endbyte+4>=b->storage){
+    /* not the main path */
+    if(b->endbyte*8+bits>b->storage*8)return(-1);
+  }
+
+  ret=b->ptr[0]<<(24+b->endbit);
+  if(bits>8){
+    ret|=b->ptr[1]<<(16+b->endbit);
+    if(bits>16){
+      ret|=b->ptr[2]<<(8+b->endbit);
+      if(bits>24){
+	ret|=b->ptr[3]<<(b->endbit);
+	if(bits>32 && b->endbit)
+	  ret|=b->ptr[4]>>(8-b->endbit);
+      }
+    }
+  }
+  return ((ret&0xffffffff)>>(m>>1))>>((m+1)>>1);
+}
+
+long oggpack_look1(oggpack_buffer *b){
+  if(b->endbyte>=b->storage)return(-1);
+  return((b->ptr[0]>>b->endbit)&1);
+}
+
+long oggpackB_look1(oggpack_buffer *b){
+  if(b->endbyte>=b->storage)return(-1);
+  return((b->ptr[0]>>(7-b->endbit))&1);
+}
+
+void oggpack_adv(oggpack_buffer *b,int bits){
+  bits+=b->endbit;
+  b->ptr+=bits/8;
+  b->endbyte+=bits/8;
+  b->endbit=bits&7;
+}
+
+void oggpackB_adv(oggpack_buffer *b,int bits){
+  oggpack_adv(b,bits);
+}
+
+void oggpack_adv1(oggpack_buffer *b){
+  if(++(b->endbit)>7){
+    b->endbit=0;
+    b->ptr++;
+    b->endbyte++;
+  }
+}
+
+void oggpackB_adv1(oggpack_buffer *b){
+  oggpack_adv1(b);
+}
+
+/* bits <= 32 */
+long oggpack_read(oggpack_buffer *b,int bits){
+  long ret;
+  unsigned long m=mask[bits];
+
+  bits+=b->endbit;
+
+  if(b->endbyte+4>=b->storage){
+    /* not the main path */
+    ret=-1L;
+    if(b->endbyte*8+bits>b->storage*8)goto overflow;
+  }
+
+  ret=b->ptr[0]>>b->endbit;
+  if(bits>8){
+    ret|=b->ptr[1]<<(8-b->endbit);
+    if(bits>16){
+      ret|=b->ptr[2]<<(16-b->endbit);
+      if(bits>24){
+	ret|=b->ptr[3]<<(24-b->endbit);
+	if(bits>32 && b->endbit){
+	  ret|=b->ptr[4]<<(32-b->endbit);
+	}
+      }
+    }
+  }
+  ret&=m;
+
+ overflow:
+
+  b->ptr+=bits/8;
+  b->endbyte+=bits/8;
+  b->endbit=bits&7;
+  return(ret);
+}
+
+/* bits <= 32 */
+long oggpackB_read(oggpack_buffer *b,int bits){
+  long ret;
+  long m=32-bits;
+
+  bits+=b->endbit;
+
+  if(b->endbyte+4>=b->storage){
+    /* not the main path */
+    ret=-1L;
+    if(b->endbyte*8+bits>b->storage*8)goto overflow;
+  }
+
+  ret=b->ptr[0]<<(24+b->endbit);
+  if(bits>8){
+    ret|=b->ptr[1]<<(16+b->endbit);
+    if(bits>16){
+      ret|=b->ptr[2]<<(8+b->endbit);
+      if(bits>24){
+	ret|=b->ptr[3]<<(b->endbit);
+	if(bits>32 && b->endbit)
+	  ret|=b->ptr[4]>>(8-b->endbit);
+      }
+    }
+  }
+  ret=((ret&0xffffffffUL)>>(m>>1))>>((m+1)>>1);
+
+ overflow:
+
+  b->ptr+=bits/8;
+  b->endbyte+=bits/8;
+  b->endbit=bits&7;
+  return(ret);
+}
+
+long oggpack_read1(oggpack_buffer *b){
+  long ret;
+
+  if(b->endbyte>=b->storage){
+    /* not the main path */
+    ret=-1L;
+    goto overflow;
+  }
+
+  ret=(b->ptr[0]>>b->endbit)&1;
+
+ overflow:
+
+  b->endbit++;
+  if(b->endbit>7){
+    b->endbit=0;
+    b->ptr++;
+    b->endbyte++;
+  }
+  return(ret);
+}
+
+long oggpackB_read1(oggpack_buffer *b){
+  long ret;
+
+  if(b->endbyte>=b->storage){
+    /* not the main path */
+    ret=-1L;
+    goto overflow;
+  }
+
+  ret=(b->ptr[0]>>(7-b->endbit))&1;
+
+ overflow:
+
+  b->endbit++;
+  if(b->endbit>7){
+    b->endbit=0;
+    b->ptr++;
+    b->endbyte++;
+  }
+  return(ret);
+}
+
+long oggpack_bytes(oggpack_buffer *b){
+  return(b->endbyte+(b->endbit+7)/8);
+}
+
+long oggpack_bits(oggpack_buffer *b){
+  return(b->endbyte*8+b->endbit);
+}
+
+long oggpackB_bytes(oggpack_buffer *b){
+  return oggpack_bytes(b);
+}
+
+long oggpackB_bits(oggpack_buffer *b){
+  return oggpack_bits(b);
+}
+
+unsigned char *oggpack_get_buffer(oggpack_buffer *b){
+  return(b->buffer);
+}
+
+unsigned char *oggpackB_get_buffer(oggpack_buffer *b){
+  return oggpack_get_buffer(b);
+}
+
+/* Self test of the bitwise routines; everything else is based on
+   them, so they damned well better be solid. */
+
+#ifdef _V_SELFTEST
+#include <stdio.h>
+
+static int ilog(unsigned int v){
+  int ret=0;
+  while(v){
+    ret++;
+    v>>=1;
+  }
+  return(ret);
+}
+
+oggpack_buffer o;
+oggpack_buffer r;
+
+void report(char *in){
+  fprintf(stderr,"%s",in);
+  exit(1);
+}
+
+void cliptest(unsigned long *b,int vals,int bits,int *comp,int compsize){
+  long bytes,i;
+  unsigned char *buffer;
+
+  oggpack_reset(&o);
+  for(i=0;i<vals;i++)
+    oggpack_write(&o,b[i],bits?bits:ilog(b[i]));
+  buffer=oggpack_get_buffer(&o);
+  bytes=oggpack_bytes(&o);
+  if(bytes!=compsize)report("wrong number of bytes!\n");
+  for(i=0;i<bytes;i++)if(buffer[i]!=comp[i]){
+    for(i=0;i<bytes;i++)fprintf(stderr,"%x %x\n",(int)buffer[i],(int)comp[i]);
+    report("wrote incorrect value!\n");
+  }
+  oggpack_readinit(&r,buffer,bytes);
+  for(i=0;i<vals;i++){
+    int tbit=bits?bits:ilog(b[i]);
+    if(oggpack_look(&r,tbit)==-1)
+      report("out of data!\n");
+    if(oggpack_look(&r,tbit)!=(b[i]&mask[tbit]))
+      report("looked at incorrect value!\n");
+    if(tbit==1)
+      if(oggpack_look1(&r)!=(b[i]&mask[tbit]))
+	report("looked at single bit incorrect value!\n");
+    if(tbit==1){
+      if(oggpack_read1(&r)!=(b[i]&mask[tbit]))
+	report("read incorrect single bit value!\n");
+    }else{
+    if(oggpack_read(&r,tbit)!=(b[i]&mask[tbit]))
+      report("read incorrect value!\n");
+    }
+  }
+  if(oggpack_bytes(&r)!=bytes)report("leftover bytes after read!\n");
+}
+
+void cliptestB(unsigned long *b,int vals,int bits,int *comp,int compsize){
+  long bytes,i;
+  unsigned char *buffer;
+
+  oggpackB_reset(&o);
+  for(i=0;i<vals;i++)
+    oggpackB_write(&o,b[i],bits?bits:ilog(b[i]));
+  buffer=oggpackB_get_buffer(&o);
+  bytes=oggpackB_bytes(&o);
+  if(bytes!=compsize)report("wrong number of bytes!\n");
+  for(i=0;i<bytes;i++)if(buffer[i]!=comp[i]){
+    for(i=0;i<bytes;i++)fprintf(stderr,"%x %x\n",(int)buffer[i],(int)comp[i]);
+    report("wrote incorrect value!\n");
+  }
+  oggpackB_readinit(&r,buffer,bytes);
+  for(i=0;i<vals;i++){
+    int tbit=bits?bits:ilog(b[i]);
+    if(oggpackB_look(&r,tbit)==-1)
+      report("out of data!\n");
+    if(oggpackB_look(&r,tbit)!=(b[i]&mask[tbit]))
+      report("looked at incorrect value!\n");
+    if(tbit==1)
+      if(oggpackB_look1(&r)!=(b[i]&mask[tbit]))
+	report("looked at single bit incorrect value!\n");
+    if(tbit==1){
+      if(oggpackB_read1(&r)!=(b[i]&mask[tbit]))
+	report("read incorrect single bit value!\n");
+    }else{
+    if(oggpackB_read(&r,tbit)!=(b[i]&mask[tbit]))
+      report("read incorrect value!\n");
+    }
+  }
+  if(oggpackB_bytes(&r)!=bytes)report("leftover bytes after read!\n");
+}
+
+int main(void){
+  unsigned char *buffer;
+  long bytes,i;
+  static unsigned long testbuffer1[]=
+    {18,12,103948,4325,543,76,432,52,3,65,4,56,32,42,34,21,1,23,32,546,456,7,
+       567,56,8,8,55,3,52,342,341,4,265,7,67,86,2199,21,7,1,5,1,4};
+  int test1size=43;
+
+  static unsigned long testbuffer2[]=
+    {216531625L,1237861823,56732452,131,3212421,12325343,34547562,12313212,
+       1233432,534,5,346435231,14436467,7869299,76326614,167548585,
+       85525151,0,12321,1,349528352};
+  int test2size=21;
+
+  static unsigned long testbuffer3[]=
+    {1,0,14,0,1,0,12,0,1,0,0,0,1,1,0,1,0,1,0,1,0,1,0,1,0,1,0,0,1,1,1,1,1,0,0,1,
+       0,1,30,1,1,1,0,0,1,0,0,0,12,0,11,0,1,0,0,1};
+  int test3size=56;
+
+  static unsigned long large[]=
+    {2136531625L,2137861823,56732452,131,3212421,12325343,34547562,12313212,
+       1233432,534,5,2146435231,14436467,7869299,76326614,167548585,
+       85525151,0,12321,1,2146528352};
+
+  int onesize=33;
+  static int one[33]={146,25,44,151,195,15,153,176,233,131,196,65,85,172,47,40,
+                    34,242,223,136,35,222,211,86,171,50,225,135,214,75,172,
+                    223,4};
+  static int oneB[33]={150,101,131,33,203,15,204,216,105,193,156,65,84,85,222,
+		       8,139,145,227,126,34,55,244,171,85,100,39,195,173,18,
+		       245,251,128};
+
+  int twosize=6;
+  static int two[6]={61,255,255,251,231,29};
+  static int twoB[6]={247,63,255,253,249,120};
+
+  int threesize=54;
+  static int three[54]={169,2,232,252,91,132,156,36,89,13,123,176,144,32,254,
+                      142,224,85,59,121,144,79,124,23,67,90,90,216,79,23,83,
+                      58,135,196,61,55,129,183,54,101,100,170,37,127,126,10,
+                      100,52,4,14,18,86,77,1};
+  static int threeB[54]={206,128,42,153,57,8,183,251,13,89,36,30,32,144,183,
+			 130,59,240,121,59,85,223,19,228,180,134,33,107,74,98,
+			 233,253,196,135,63,2,110,114,50,155,90,127,37,170,104,
+			 200,20,254,4,58,106,176,144,0};
+
+  int foursize=38;
+  static int four[38]={18,6,163,252,97,194,104,131,32,1,7,82,137,42,129,11,72,
+                     132,60,220,112,8,196,109,64,179,86,9,137,195,208,122,169,
+                     28,2,133,0,1};
+  static int fourB[38]={36,48,102,83,243,24,52,7,4,35,132,10,145,21,2,93,2,41,
+			1,219,184,16,33,184,54,149,170,132,18,30,29,98,229,67,
+			129,10,4,32};
+
+  int fivesize=45;
+  static int five[45]={169,2,126,139,144,172,30,4,80,72,240,59,130,218,73,62,
+                     241,24,210,44,4,20,0,248,116,49,135,100,110,130,181,169,
+                     84,75,159,2,1,0,132,192,8,0,0,18,22};
+  static int fiveB[45]={1,84,145,111,245,100,128,8,56,36,40,71,126,78,213,226,
+			124,105,12,0,133,128,0,162,233,242,67,152,77,205,77,
+			172,150,169,129,79,128,0,6,4,32,0,27,9,0};
+
+  int sixsize=7;
+  static int six[7]={17,177,170,242,169,19,148};
+  static int sixB[7]={136,141,85,79,149,200,41};
+
+  /* Test read/write together */
+  /* Later we test against pregenerated bitstreams */
+  oggpack_writeinit(&o);
+
+  fprintf(stderr,"\nSmall preclipped packing (LSb): ");
+  cliptest(testbuffer1,test1size,0,one,onesize);
+  fprintf(stderr,"ok.");
+
+  fprintf(stderr,"\nNull bit call (LSb): ");
+  cliptest(testbuffer3,test3size,0,two,twosize);
+  fprintf(stderr,"ok.");
+
+  fprintf(stderr,"\nLarge preclipped packing (LSb): ");
+  cliptest(testbuffer2,test2size,0,three,threesize);
+  fprintf(stderr,"ok.");
+
+  fprintf(stderr,"\n32 bit preclipped packing (LSb): ");
+  oggpack_reset(&o);
+  for(i=0;i<test2size;i++)
+    oggpack_write(&o,large[i],32);
+  buffer=oggpack_get_buffer(&o);
+  bytes=oggpack_bytes(&o);
+  oggpack_readinit(&r,buffer,bytes);
+  for(i=0;i<test2size;i++){
+    if(oggpack_look(&r,32)==-1)report("out of data. failed!");
+    if(oggpack_look(&r,32)!=large[i]){
+      fprintf(stderr,"%ld != %ld (%lx!=%lx):",oggpack_look(&r,32),large[i],
+	      oggpack_look(&r,32),large[i]);
+      report("read incorrect value!\n");
+    }
+    oggpack_adv(&r,32);
+  }
+  if(oggpack_bytes(&r)!=bytes)report("leftover bytes after read!\n");
+  fprintf(stderr,"ok.");
+
+  fprintf(stderr,"\nSmall unclipped packing (LSb): ");
+  cliptest(testbuffer1,test1size,7,four,foursize);
+  fprintf(stderr,"ok.");
+
+  fprintf(stderr,"\nLarge unclipped packing (LSb): ");
+  cliptest(testbuffer2,test2size,17,five,fivesize);
+  fprintf(stderr,"ok.");
+
+  fprintf(stderr,"\nSingle bit unclipped packing (LSb): ");
+  cliptest(testbuffer3,test3size,1,six,sixsize);
+  fprintf(stderr,"ok.");
+
+  fprintf(stderr,"\nTesting read past end (LSb): ");
+  oggpack_readinit(&r,"\0\0\0\0\0\0\0\0",8);
+  for(i=0;i<64;i++){
+    if(oggpack_read(&r,1)!=0){
+      fprintf(stderr,"failed; got -1 prematurely.\n");
+      exit(1);
+    }
+  }
+  if(oggpack_look(&r,1)!=-1 ||
+     oggpack_read(&r,1)!=-1){
+      fprintf(stderr,"failed; read past end without -1.\n");
+      exit(1);
+  }
+  oggpack_readinit(&r,"\0\0\0\0\0\0\0\0",8);
+  if(oggpack_read(&r,30)!=0 || oggpack_read(&r,16)!=0){
+      fprintf(stderr,"failed 2; got -1 prematurely.\n");
+      exit(1);
+  }
+
+  if(oggpack_look(&r,18)!=0 ||
+     oggpack_look(&r,18)!=0){
+    fprintf(stderr,"failed 3; got -1 prematurely.\n");
+      exit(1);
+  }
+  if(oggpack_look(&r,19)!=-1 ||
+     oggpack_look(&r,19)!=-1){
+    fprintf(stderr,"failed; read past end without -1.\n");
+      exit(1);
+  }
+  if(oggpack_look(&r,32)!=-1 ||
+     oggpack_look(&r,32)!=-1){
+    fprintf(stderr,"failed; read past end without -1.\n");
+      exit(1);
+  }
+  oggpack_writeclear(&o);
+  fprintf(stderr,"ok.\n");
+
+  /********** lazy, cut-n-paste retest with MSb packing ***********/
+
+  /* Test read/write together */
+  /* Later we test against pregenerated bitstreams */
+  oggpackB_writeinit(&o);
+
+  fprintf(stderr,"\nSmall preclipped packing (MSb): ");
+  cliptestB(testbuffer1,test1size,0,oneB,onesize);
+  fprintf(stderr,"ok.");
+
+  fprintf(stderr,"\nNull bit call (MSb): ");
+  cliptestB(testbuffer3,test3size,0,twoB,twosize);
+  fprintf(stderr,"ok.");
+
+  fprintf(stderr,"\nLarge preclipped packing (MSb): ");
+  cliptestB(testbuffer2,test2size,0,threeB,threesize);
+  fprintf(stderr,"ok.");
+
+  fprintf(stderr,"\n32 bit preclipped packing (MSb): ");
+  oggpackB_reset(&o);
+  for(i=0;i<test2size;i++)
+    oggpackB_write(&o,large[i],32);
+  buffer=oggpackB_get_buffer(&o);
+  bytes=oggpackB_bytes(&o);
+  oggpackB_readinit(&r,buffer,bytes);
+  for(i=0;i<test2size;i++){
+    if(oggpackB_look(&r,32)==-1)report("out of data. failed!");
+    if(oggpackB_look(&r,32)!=large[i]){
+      fprintf(stderr,"%ld != %ld (%lx!=%lx):",oggpackB_look(&r,32),large[i],
+	      oggpackB_look(&r,32),large[i]);
+      report("read incorrect value!\n");
+    }
+    oggpackB_adv(&r,32);
+  }
+  if(oggpackB_bytes(&r)!=bytes)report("leftover bytes after read!\n");
+  fprintf(stderr,"ok.");
+
+  fprintf(stderr,"\nSmall unclipped packing (MSb): ");
+  cliptestB(testbuffer1,test1size,7,fourB,foursize);
+  fprintf(stderr,"ok.");
+
+  fprintf(stderr,"\nLarge unclipped packing (MSb): ");
+  cliptestB(testbuffer2,test2size,17,fiveB,fivesize);
+  fprintf(stderr,"ok.");
+
+  fprintf(stderr,"\nSingle bit unclipped packing (MSb): ");
+  cliptestB(testbuffer3,test3size,1,sixB,sixsize);
+  fprintf(stderr,"ok.");
+
+  fprintf(stderr,"\nTesting read past end (MSb): ");
+  oggpackB_readinit(&r,"\0\0\0\0\0\0\0\0",8);
+  for(i=0;i<64;i++){
+    if(oggpackB_read(&r,1)!=0){
+      fprintf(stderr,"failed; got -1 prematurely.\n");
+      exit(1);
+    }
+  }
+  if(oggpackB_look(&r,1)!=-1 ||
+     oggpackB_read(&r,1)!=-1){
+      fprintf(stderr,"failed; read past end without -1.\n");
+      exit(1);
+  }
+  oggpackB_readinit(&r,"\0\0\0\0\0\0\0\0",8);
+  if(oggpackB_read(&r,30)!=0 || oggpackB_read(&r,16)!=0){
+      fprintf(stderr,"failed 2; got -1 prematurely.\n");
+      exit(1);
+  }
+
+  if(oggpackB_look(&r,18)!=0 ||
+     oggpackB_look(&r,18)!=0){
+    fprintf(stderr,"failed 3; got -1 prematurely.\n");
+      exit(1);
+  }
+  if(oggpackB_look(&r,19)!=-1 ||
+     oggpackB_look(&r,19)!=-1){
+    fprintf(stderr,"failed; read past end without -1.\n");
+      exit(1);
+  }
+  if(oggpackB_look(&r,32)!=-1 ||
+     oggpackB_look(&r,32)!=-1){
+    fprintf(stderr,"failed; read past end without -1.\n");
+      exit(1);
+  }
+  oggpackB_writeclear(&o);
+  fprintf(stderr,"ok.\n\n");
+
+
+  return(0);
+}
+#endif  /* _V_SELFTEST */
+
+#undef BUFFER_INCREMENT

source/modules/juce_audio_formats/codecs/oggvorbis/codec.h

@@ -0,0 +1,242 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2001             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+
+ ********************************************************************
+
+ function: libvorbis codec headers
+ last mod: $Id: codec.h 17021 2010-03-24 09:29:41Z xiphmont $
+
+ ********************************************************************/
+
+#ifndef _vorbis_codec_h_
+#define _vorbis_codec_h_
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif /* __cplusplus */
+
+#include "ogg.h"
+
+typedef struct vorbis_info{
+  int version;
+  int channels;
+  long rate;
+
+  /* The below bitrate declarations are *hints*.
+     Combinations of the three values carry the following implications:
+
+     all three set to the same value:
+       implies a fixed rate bitstream
+     only nominal set:
+       implies a VBR stream that averages the nominal bitrate.  No hard
+       upper/lower limit
+     upper and or lower set:
+       implies a VBR bitstream that obeys the bitrate limits. nominal
+       may also be set to give a nominal rate.
+     none set:
+       the coder does not care to speculate.
+  */
+
+  long bitrate_upper;
+  long bitrate_nominal;
+  long bitrate_lower;
+  long bitrate_window;
+
+  void *codec_setup;
+} vorbis_info;
+
+/* vorbis_dsp_state buffers the current vorbis audio
+   analysis/synthesis state.  The DSP state belongs to a specific
+   logical bitstream ****************************************************/
+typedef struct vorbis_dsp_state{
+  int analysisp;
+  vorbis_info *vi;
+
+  float **pcm;
+  float **pcmret;
+  int      pcm_storage;
+  int      pcm_current;
+  int      pcm_returned;
+
+  int  preextrapolate;
+  int  eofflag;
+
+  long lW;
+  long W;
+  long nW;
+  long centerW;
+
+  ogg_int64_t granulepos;
+  ogg_int64_t sequence;
+
+  ogg_int64_t glue_bits;
+  ogg_int64_t time_bits;
+  ogg_int64_t floor_bits;
+  ogg_int64_t res_bits;
+
+  void       *backend_state;
+} vorbis_dsp_state;
+
+typedef struct vorbis_block{
+  /* necessary stream state for linking to the framing abstraction */
+  float  **pcm;       /* this is a pointer into local storage */
+  oggpack_buffer opb;
+
+  long  lW;
+  long  W;
+  long  nW;
+  int   pcmend;
+  int   mode;
+
+  int         eofflag;
+  ogg_int64_t granulepos;
+  ogg_int64_t sequence;
+  vorbis_dsp_state *vd; /* For read-only access of configuration */
+
+  /* local storage to avoid remallocing; it's up to the mapping to
+     structure it */
+  void               *localstore;
+  long                localtop;
+  long                localalloc;
+  long                totaluse;
+  struct alloc_chain *reap;
+
+  /* bitmetrics for the frame */
+  long glue_bits;
+  long time_bits;
+  long floor_bits;
+  long res_bits;
+
+  void *internal;
+
+} vorbis_block;
+
+/* vorbis_block is a single block of data to be processed as part of
+the analysis/synthesis stream; it belongs to a specific logical
+bitstream, but is independent from other vorbis_blocks belonging to
+that logical bitstream. *************************************************/
+
+struct alloc_chain{
+  void *ptr;
+  struct alloc_chain *next;
+};
+
+/* vorbis_info contains all the setup information specific to the
+   specific compression/decompression mode in progress (eg,
+   psychoacoustic settings, channel setup, options, codebook
+   etc). vorbis_info and substructures are in backends.h.
+*********************************************************************/
+
+/* the comments are not part of vorbis_info so that vorbis_info can be
+   static storage */
+typedef struct vorbis_comment{
+  /* unlimited user comment fields.  libvorbis writes 'libvorbis'
+     whatever vendor is set to in encode */
+  char **user_comments;
+  int   *comment_lengths;
+  int    comments;
+  char  *vendor;
+
+} vorbis_comment;
+
+
+/* libvorbis encodes in two abstraction layers; first we perform DSP
+   and produce a packet (see docs/analysis.txt).  The packet is then
+   coded into a framed OggSquish bitstream by the second layer (see
+   docs/framing.txt).  Decode is the reverse process; we sync/frame
+   the bitstream and extract individual packets, then decode the
+   packet back into PCM audio.
+
+   The extra framing/packetizing is used in streaming formats, such as
+   files.  Over the net (such as with UDP), the framing and
+   packetization aren't necessary as they're provided by the transport
+   and the streaming layer is not used */
+
+/* Vorbis PRIMITIVES: general ***************************************/
+
+extern void     vorbis_info_init(vorbis_info *vi);
+extern void     vorbis_info_clear(vorbis_info *vi);
+extern int      vorbis_info_blocksize(vorbis_info *vi,int zo);
+extern void     vorbis_comment_init(vorbis_comment *vc);
+extern void     vorbis_comment_add(vorbis_comment *vc, const char *comment);
+extern void     vorbis_comment_add_tag(vorbis_comment *vc,
+                                       const char *tag, const char *contents);
+extern char    *vorbis_comment_query(vorbis_comment *vc, const char *tag, int count);
+extern int      vorbis_comment_query_count(vorbis_comment *vc, const char *tag);
+extern void     vorbis_comment_clear(vorbis_comment *vc);
+
+extern int      vorbis_block_init(vorbis_dsp_state *v, vorbis_block *vb);
+extern int      vorbis_block_clear(vorbis_block *vb);
+extern void     vorbis_dsp_clear(vorbis_dsp_state *v);
+extern double   vorbis_granule_time(vorbis_dsp_state *v,
+                                    ogg_int64_t granulepos);
+
+extern const char *vorbis_version_string(void);
+
+/* Vorbis PRIMITIVES: analysis/DSP layer ****************************/
+
+extern int      vorbis_analysis_init(vorbis_dsp_state *v,vorbis_info *vi);
+extern int      vorbis_commentheader_out(vorbis_comment *vc, ogg_packet *op);
+extern int      vorbis_analysis_headerout(vorbis_dsp_state *v,
+                                          vorbis_comment *vc,
+                                          ogg_packet *op,
+                                          ogg_packet *op_comm,
+                                          ogg_packet *op_code);
+extern float  **vorbis_analysis_buffer(vorbis_dsp_state *v,int vals);
+extern int      vorbis_analysis_wrote(vorbis_dsp_state *v,int vals);
+extern int      vorbis_analysis_blockout(vorbis_dsp_state *v,vorbis_block *vb);
+extern int      vorbis_analysis(vorbis_block *vb,ogg_packet *op);
+
+extern int      vorbis_bitrate_addblock(vorbis_block *vb);
+extern int      vorbis_bitrate_flushpacket(vorbis_dsp_state *vd,
+                                           ogg_packet *op);
+
+/* Vorbis PRIMITIVES: synthesis layer *******************************/
+extern int      vorbis_synthesis_idheader(ogg_packet *op);
+extern int      vorbis_synthesis_headerin(vorbis_info *vi,vorbis_comment *vc,
+                                          ogg_packet *op);
+
+extern int      vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi);
+extern int      vorbis_synthesis_restart(vorbis_dsp_state *v);
+extern int      vorbis_synthesis(vorbis_block *vb,ogg_packet *op);
+extern int      vorbis_synthesis_trackonly(vorbis_block *vb,ogg_packet *op);
+extern int      vorbis_synthesis_blockin(vorbis_dsp_state *v,vorbis_block *vb);
+extern int      vorbis_synthesis_pcmout(vorbis_dsp_state *v,float ***pcm);
+extern int      vorbis_synthesis_lapout(vorbis_dsp_state *v,float ***pcm);
+extern int      vorbis_synthesis_read(vorbis_dsp_state *v,int samples);
+extern long     vorbis_packet_blocksize(vorbis_info *vi,ogg_packet *op);
+
+extern int      vorbis_synthesis_halfrate(vorbis_info *v,int flag);
+extern int      vorbis_synthesis_halfrate_p(vorbis_info *v);
+
+/* Vorbis ERRORS and return codes ***********************************/
+
+#define OV_FALSE      -1
+#define OV_EOF        -2
+#define OV_HOLE       -3
+
+#define OV_EREAD      -128
+#define OV_EFAULT     -129
+#define OV_EIMPL      -130
+#define OV_EINVAL     -131
+#define OV_ENOTVORBIS -132
+#define OV_EBADHEADER -133
+#define OV_EVERSION   -134
+#define OV_ENOTAUDIO  -135
+#define OV_EBADPACKET -136
+#define OV_EBADLINK   -137
+#define OV_ENOSEEK    -138
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+#endif

source/modules/juce_audio_formats/codecs/oggvorbis/config_types.h

@@ -0,0 +1,10 @@
+#ifndef __CONFIG_TYPES_H__
+#define __CONFIG_TYPES_H__
+
+typedef int16_t ogg_int16_t;
+typedef unsigned short ogg_uint16_t;
+typedef int32_t ogg_int32_t;
+typedef unsigned int ogg_uint32_t;
+typedef int64_t ogg_int64_t;
+
+#endif

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/AUTHORS

@@ -0,0 +1,3 @@
+Monty <monty@xiph.org>
+
+and the rest of the Xiph.org Foundation.

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/CHANGES

@@ -0,0 +1,126 @@
+libvorbis 1.3.2 (2010-11-01) -- "Xiph.Org libVorbis I 20101101 (Schaufenugget)"
+
+ * vorbis: additional proofing against invalid/malicious 
+   streams in floor, residue, and bos/eos packet trimming 
+   code (see SVN for details). 
+ * vorbis: Added programming documentation tree for the 
+   low-level calls
+ * vorbisfile: Correct handling of serial numbers array 
+   element [0] on non-seekable streams
+ * vorbisenc: Back out an [old] AoTuV HF weighting that was 
+   first enabled in 1.3.0; there are a few samples where I 
+   really don't like the effect it causes.
+ * vorbis: return correct timestamp for granule positions 
+   with high bit set.
+ * vorbisfile: the [undocumented] half-rate decode api made no 
+   attempt to keep the pcm offset tracking consistent in seeks. 
+   Fix and add a testing mode to seeking_example.c to torture 
+   test seeking in halfrate mode.  Also remove requirement that 
+   halfrate mode only work with seekable files.
+ * vorbisfile:  Fix a chaining bug in raw_seeks where seeking 
+   out of the current link would fail due to not 
+   reinitializing the decode machinery.  
+ * vorbisfile: improve seeking strategy. Reduces the 
+   necessary number of seek callbacks in an open or seek 
+   operation by well over 2/3.
+
+libvorbis 1.3.1 (2010-02-26) -- "Xiph.Org libVorbis I 20100325 (Everywhere)"
+
+ * tweak + minor arithmetic fix in floor1 fit
+ * revert noise norm to conservative 1.2.3 behavior pending 
+   more listening testing
+
+libvorbis 1.3.0 (2010-02-25) -- unreleased staging snapshot
+
+ * Optimized surround support for 5.1 encoding at 44.1/48kHz
+ * Added encoder control call to disable channel coupling
+ * Correct an overflow bug in very low-bitrate encoding on 32 bit 
+   machines that caused inflated bitrates
+ * Numerous API hardening, leak and build fixes 
+ * Correct bug in 22kHz compand setup that could cause a crash
+ * Correct bug in 16kHz codebooks that could cause unstable pure 
+   tones at high bitrates
+
+libvorbis 1.2.3 (2009-07-09) -- "Xiph.Org libVorbis I 20090709"
+
+ * correct a vorbisfile bug that prevented proper playback of
+   Vorbis files where all audio in a logical stream is in a
+   single page
+ * Additional decode setup hardening against malicious streams
+ * Add 'OV_EXCLUDE_STATIC_CALLBACKS' define for developers who 
+   wish to avoid unused symbol warnings from the static callbacks 
+   defined in vorbisfile.h
+
+libvorbis 1.2.2 (2009-06-24) -- "Xiph.Org libVorbis I 20090624"
+
+ * define VENDOR and ENCODER strings
+ * seek correctly in files bigger than 2 GB (Windows)
+ * fix regression from CVE-2008-1420; 1.0b1 files work again
+ * mark all tables as constant to reduce memory occupation
+ * additional decoder hardening against malicious streams
+ * substantially reduce amount of seeking performed by Vorbisfile
+ * Multichannel decode bugfix 
+ * build system updates
+ * minor specification clarifications/fixes
+
+libvorbis 1.2.1 (unreleased) -- "Xiph.Org libVorbis I 20080501"
+
+ * Improved robustness with corrupt streams.
+ * New ov_read_filter() vorbisfile call allows filtering decoded
+   audio as floats before converting to integer samples.
+ * Fix an encoder bug with multichannel streams.
+ * Replaced RTP payload format draft with RFC 5215.
+ * Bare bones self test under 'make check'.
+ * Fix a problem encoding some streams between 14 and 28 kHz.
+ * Fix a numerical instability in the edge extrapolation filter.
+ * Build system improvements.
+ * Specification correction.
+
+libvorbis 1.2.0 (2007-07-25) -- "Xiph.Org libVorbis I 20070622"
+
+ * new ov_fopen() convenience call that avoids the common
+   stdio conflicts with ov_open() and MSVC runtimes.
+ * libvorbisfile now handles multiplexed streams
+ * improve robustness to corrupt input streams
+ * fix a minor encoder bug
+ * updated RTP draft
+ * build system updates
+ * minor corrections to the specification
+
+libvorbis 1.1.2 (2005-11-27) -- "Xiph.Org libVorbis I 20050304"
+
+ * fix a serious encoder bug with gcc 4 optimized builds
+ * documentation and spec fixes
+ * updated VS2003 and XCode builds
+ * new draft RTP encapsulation spec
+
+libvorbis 1.1.1 (2005-06-27) -- "Xiph.Org libVorbis I 20050304"
+
+ * bug fix to the bitrate management encoder interface
+ * bug fix to properly set packetno field in the encoder
+ * new draft RTP encapsulation spec
+ * library API documentation improvements
+
+libvorbis 1.1.0 (2004-09-22) -- "Xiph.Org libVorbis I 20040629"
+
+ * merges tuning improvements from Aoyumi's aoTuV with fixups
+ * new managed bitrate (CBR) mode support
+ * new vorbis_encoder_ctl() interface
+ * extensive documentation updates
+ * application/ogg mimetype is now official
+ * autotools cleanup from Thomas Vander Stichele
+ * SymbianOS build support from Colin Ward at CSIRO
+ * various bugfixes
+ * various packaging improvements
+
+libvorbis 1.0.1 (2003-11-17) -- "Xiph.Org libVorbis I 20030909"
+
+ * numerous bug fixes
+ * specification corrections
+ * new crosslap and halfrate APIs for game use
+ * packaging and build updates
+
+libvorbis 1.0.0 (2002-07-19) -- "Xiph.Org libVorbis I 20020717"
+
+ * first stable release
+

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/COPYING

@@ -0,0 +1,28 @@
+Copyright (c) 2002-2008 Xiph.org Foundation
+
+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/oggvorbis/libvorbis-1.3.2/README

@@ -0,0 +1,134 @@
+********************************************************************
+*                                                                  *
+* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+*                                                                  *
+* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007             *
+* by the Xiph.org Foundation, http://www.xiph.org/                 *
+*                                                                  *
+********************************************************************
+
+Vorbis is a general purpose audio and music encoding format
+contemporary to MPEG-4's AAC and TwinVQ, the next generation beyond
+MPEG audio layer 3. Unlike the MPEG sponsored formats (and other
+proprietary formats such as RealAudio G2 and Windows' flavor of the
+month), the Vorbis CODEC specification belongs to the public domain.
+All the technical details are published and documented, and any
+software entity may make full use of the format without license 
+fee, royalty or patent concerns.
+
+This package contains:
+
+* libvorbis, a BSD-style license software implementation of
+  the Vorbis specification by the Xiph.Org Foundation 
+  (http://www.xiph.org/) 
+
+* libvorbisfile, a BSD-style license convenience library
+  built on Vorbis designed to simplify common uses
+
+* libvorbisenc, a BSD-style license library that provides a simple,
+  programmatic encoding setup interface 
+
+* example code making use of libogg, libvorbis, libvorbisfile and
+  libvorbisenc
+
+WHAT'S HERE:
+
+This source distribution includes libvorbis and an example
+encoder/player to demonstrate use of libvorbis as well as
+documentation on the Ogg Vorbis audio coding format.
+
+You'll need libogg (distributed separately) to compile this library.
+A more comprehensive set of utilities is available in the vorbis-tools
+package.
+
+Directory:
+
+./lib  		The source for the libraries, a BSD-license implementation
+		of the public domain Ogg Vorbis audio encoding format.
+
+./include       Library API headers
+
+./debian        Rules/spec files for building Debian .deb packages
+
+./doc           Vorbis documentation
+
+./examples	Example code illustrating programmatic use of libvorbis, 
+		libvorbisfile and libvorbisenc
+
+./mac 		Codewarrior project files and build tweaks for MacOS.
+
+./macosx 	Project files for MacOS X.
+
+./win32		Win32 projects files and build automation
+
+./vq 		Internal utilities for training/building new LSP/residue 
+		and auxiliary codebooks.
+
+CONTACT:
+
+The Ogg homepage is located at 'http://www.xiph.org/ogg/'.
+Vorbis's homepage is located at 'http://www.xiph.org/vorbis/'.
+Up to date technical documents, contact information, source code and
+pre-built utilities may be found there.
+
+The user website for Ogg Vorbis software and audio is http://vorbis.com/
+
+BUILDING FROM TRUNK:
+
+Development source is under subversion revision control at 
+https://svn.xiph.org/trunk/vorbis/. You will also need the 
+newest versions of autoconf, automake, libtool and pkg-config in
+order to compile Vorbis from development source. A configure script
+is provided for you in the source tarball distributions.
+
+  [update or checkout latest source]
+  ./autogen.sh
+  make
+
+and as root if desired:
+
+  make install
+
+This will install the Vorbis libraries (static and shared) into
+/usr/local/lib, includes into /usr/local/include and API manpages
+(once we write some) into /usr/local/man.
+
+Documentation building requires xsltproc and pdfxmltex.
+
+BUILDING FROM TARBALL DISTRIBUTIONS:
+
+  ./configure
+  make
+
+and optionally (as root):
+  make install
+
+BUILDING RPMS:
+
+after normal configuring:
+
+  make dist
+  rpm -ta libvorbis-<version>.tar.gz
+
+BUILDING ON MACOS 9:
+
+Vorbis on MacOS 9 is built using Metroworks CodeWarrior.  To build it, 
+first verify that the Ogg libraries are already built following the
+instructions in the Ogg module README.  Open vorbis/mac/libvorbis.mcp,
+switch to the "Targets" pane, select everything, and make the project.
+Do the same thing to build libvorbisenc.mcp, and libvorbisfile.mcp (in
+that order).  In vorbis/mac/Output you will now have both debug and final
+versions of Vorbis shared libraries to link your projects against.
+
+To build a project using Ogg Vorbis, add access paths to your
+CodeWarrior project for the ogg/include, ogg/mac/Output,
+vorbis/include, and vorbis/mac/Output folders.  Be sure that
+"interpret DOS and Unix paths" is turned on in your project; it can
+be found in the "access paths" pane in your project settings.  Now
+simply add the shared libraries you need to your project (OggLib and
+VorbisLib at least) and #include "ogg/ogg.h" and "vorbis/codec.h"
+wherever you need to access Ogg and Vorbis functionality.
+

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/analysis.c

@@ -0,0 +1,109 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: single-block PCM analysis mode dispatch
+ last mod: $Id: analysis.c 16226 2009-07-08 06:43:49Z xiphmont $
+
+ ********************************************************************/
+
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+#include "../../ogg.h"
+#include "../../codec.h"
+#include "codec_internal.h"
+#include "registry.h"
+#include "scales.h"
+#include "os.h"
+#include "misc.h"
+
+/* decides between modes, dispatches to the appropriate mapping. */
+int vorbis_analysis(vorbis_block *vb, ogg_packet *op){
+  int ret,i;
+  vorbis_block_internal *vbi=(vorbis_block_internal *)vb->internal;
+
+  vb->glue_bits=0;
+  vb->time_bits=0;
+  vb->floor_bits=0;
+  vb->res_bits=0;
+
+  /* first things first.  Make sure encode is ready */
+  for(i=0;i<PACKETBLOBS;i++)
+    oggpack_reset(vbi->packetblob[i]);
+
+  /* we only have one mapping type (0), and we let the mapping code
+     itself figure out what soft mode to use.  This allows easier
+     bitrate management */
+
+  if((ret=_mapping_P[0]->forward(vb)))
+    return(ret);
+
+  if(op){
+    if(vorbis_bitrate_managed(vb))
+      /* The app is using a bitmanaged mode... but not using the
+         bitrate management interface. */
+      return(OV_EINVAL);
+
+    op->packet=oggpack_get_buffer(&vb->opb);
+    op->bytes=oggpack_bytes(&vb->opb);
+    op->b_o_s=0;
+    op->e_o_s=vb->eofflag;
+    op->granulepos=vb->granulepos;
+    op->packetno=vb->sequence; /* for sake of completeness */
+  }
+  return(0);
+}
+
+#ifdef ANALYSIS
+int analysis_noisy=1;
+
+/* there was no great place to put this.... */
+void _analysis_output_always(char *base,int i,float *v,int n,int bark,int dB,ogg_int64_t off){
+  int j;
+  FILE *of;
+  char buffer[80];
+
+  sprintf(buffer,"%s_%d.m",base,i);
+  of=fopen(buffer,"w");
+
+  if(!of)perror("failed to open data dump file");
+
+  for(j=0;j<n;j++){
+    if(bark){
+      float b=toBARK((4000.f*j/n)+.25);
+      fprintf(of,"%f ",b);
+    }else
+      if(off!=0)
+        fprintf(of,"%f ",(double)(j+off)/8000.);
+      else
+        fprintf(of,"%f ",(double)j);
+
+    if(dB){
+      float val;
+      if(v[j]==0.)
+        val=-140.;
+      else
+        val=todB(v+j);
+      fprintf(of,"%f\n",val);
+    }else{
+      fprintf(of,"%f\n",v[j]);
+    }
+  }
+  fclose(of);
+}
+
+void _analysis_output(char *base,int i,float *v,int n,int bark,int dB,
+                      ogg_int64_t off){
+  if(analysis_noisy)_analysis_output_always(base,i,v,n,bark,dB,off);
+}
+
+#endif

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/backends.h

@@ -0,0 +1,144 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: libvorbis backend and mapping structures; needed for
+           static mode headers
+ last mod: $Id: backends.h 16962 2010-03-11 07:30:34Z xiphmont $
+
+ ********************************************************************/
+
+/* this is exposed up here because we need it for static modes.
+   Lookups for each backend aren't exposed because there's no reason
+   to do so */
+
+#ifndef _vorbis_backend_h_
+#define _vorbis_backend_h_
+
+#include "codec_internal.h"
+
+/* this would all be simpler/shorter with templates, but.... */
+/* Floor backend generic *****************************************/
+typedef struct{
+  void                   (*pack)  (vorbis_info_floor *,oggpack_buffer *);
+  vorbis_info_floor     *(*unpack)(vorbis_info *,oggpack_buffer *);
+  vorbis_look_floor     *(*look)  (vorbis_dsp_state *,vorbis_info_floor *);
+  void (*free_info) (vorbis_info_floor *);
+  void (*free_look) (vorbis_look_floor *);
+  void *(*inverse1)  (struct vorbis_block *,vorbis_look_floor *);
+  int   (*inverse2)  (struct vorbis_block *,vorbis_look_floor *,
+                     void *buffer,float *);
+} vorbis_func_floor;
+
+typedef struct{
+  int   order;
+  long  rate;
+  long  barkmap;
+
+  int   ampbits;
+  int   ampdB;
+
+  int   numbooks; /* <= 16 */
+  int   books[16];
+
+  float lessthan;     /* encode-only config setting hacks for libvorbis */
+  float greaterthan;  /* encode-only config setting hacks for libvorbis */
+
+} vorbis_info_floor0;
+
+
+#define VIF_POSIT 63
+#define VIF_CLASS 16
+#define VIF_PARTS 31
+typedef struct{
+  int   partitions;                /* 0 to 31 */
+  int   partitionclass[VIF_PARTS]; /* 0 to 15 */
+
+  int   class_dim[VIF_CLASS];        /* 1 to 8 */
+  int   class_subs[VIF_CLASS];       /* 0,1,2,3 (bits: 1<<n poss) */
+  int   class_book[VIF_CLASS];       /* subs ^ dim entries */
+  int   class_subbook[VIF_CLASS][8]; /* [VIF_CLASS][subs] */
+
+
+  int   mult;                      /* 1 2 3 or 4 */
+  int   postlist[VIF_POSIT+2];    /* first two implicit */
+
+
+  /* encode side analysis parameters */
+  float maxover;
+  float maxunder;
+  float maxerr;
+
+  float twofitweight;
+  float twofitatten;
+
+  int   n;
+
+} vorbis_info_floor1;
+
+/* Residue backend generic *****************************************/
+typedef struct{
+  void                 (*pack)  (vorbis_info_residue *,oggpack_buffer *);
+  vorbis_info_residue *(*unpack)(vorbis_info *,oggpack_buffer *);
+  vorbis_look_residue *(*look)  (vorbis_dsp_state *,
+                                 vorbis_info_residue *);
+  void (*free_info)    (vorbis_info_residue *);
+  void (*free_look)    (vorbis_look_residue *);
+  long **(*classx)      (struct vorbis_block *,vorbis_look_residue *,
+			int **,int *,int);
+  int  (*forward)      (oggpack_buffer *,struct vorbis_block *,
+                        vorbis_look_residue *,
+                        int **,int *,int,long **,int);
+  int  (*inverse)      (struct vorbis_block *,vorbis_look_residue *,
+                        float **,int *,int);
+} vorbis_func_residue;
+
+typedef struct vorbis_info_residue0{
+/* block-partitioned VQ coded straight residue */
+  long  begin;
+  long  end;
+
+  /* first stage (lossless partitioning) */
+  int    grouping;         /* group n vectors per partition */
+  int    partitions;       /* possible codebooks for a partition */
+  int    partvals;         /* partitions ^ groupbook dim */
+  int    groupbook;        /* huffbook for partitioning */
+  int    secondstages[64]; /* expanded out to pointers in lookup */
+  int    booklist[512];    /* list of second stage books */
+
+  /*const*/ int classmetric1[64];
+  /*const*/ int classmetric2[64];
+} vorbis_info_residue0;
+
+/* Mapping backend generic *****************************************/
+typedef struct{
+  void                 (*pack)  (vorbis_info *,vorbis_info_mapping *,
+                                 oggpack_buffer *);
+  vorbis_info_mapping *(*unpack)(vorbis_info *,oggpack_buffer *);
+  void (*free_info)    (vorbis_info_mapping *);
+  int  (*forward)      (struct vorbis_block *vb);
+  int  (*inverse)      (struct vorbis_block *vb,vorbis_info_mapping *);
+} vorbis_func_mapping;
+
+typedef struct vorbis_info_mapping0{
+  int   submaps;  /* <= 16 */
+  int   chmuxlist[256];   /* up to 256 channels in a Vorbis stream */
+
+  int   floorsubmap[16];   /* [mux] submap to floors */
+  int   residuesubmap[16]; /* [mux] submap to residue */
+
+  int   coupling_steps;
+  int   coupling_mag[256];
+  int   coupling_ang[256];
+
+} vorbis_info_mapping0;
+
+#endif

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/bitrate.c

@@ -0,0 +1,253 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: bitrate tracking and management
+ last mod: $Id: bitrate.c 16227 2009-07-08 06:58:46Z xiphmont $
+
+ ********************************************************************/
+
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+#include "../../ogg.h"
+#include "../../codec.h"
+#include "codec_internal.h"
+#include "os.h"
+#include "misc.h"
+#include "bitrate.h"
+
+/* compute bitrate tracking setup  */
+void vorbis_bitrate_init(vorbis_info *vi,bitrate_manager_state *bm){
+  codec_setup_info *ci=(codec_setup_info *)vi->codec_setup;
+  bitrate_manager_info *bi=&ci->bi;
+
+  memset(bm,0,sizeof(*bm));
+
+  if(bi && (bi->reservoir_bits>0)){
+    long ratesamples=vi->rate;
+    int  halfsamples=ci->blocksizes[0]>>1;
+
+    bm->short_per_long=ci->blocksizes[1]/ci->blocksizes[0];
+    bm->managed=1;
+
+    bm->avg_bitsper= (int) rint(1.*bi->avg_rate*halfsamples/ratesamples);
+    bm->min_bitsper= (int) rint(1.*bi->min_rate*halfsamples/ratesamples);
+    bm->max_bitsper= (int) rint(1.*bi->max_rate*halfsamples/ratesamples);
+
+    bm->avgfloat=PACKETBLOBS/2;
+
+    /* not a necessary fix, but one that leads to a more balanced
+       typical initialization */
+    {
+      long desired_fill = (long) (bi->reservoir_bits*bi->reservoir_bias);
+      bm->minmax_reservoir=desired_fill;
+      bm->avg_reservoir=desired_fill;
+    }
+
+  }
+}
+
+void vorbis_bitrate_clear(bitrate_manager_state *bm){
+  memset(bm,0,sizeof(*bm));
+  return;
+}
+
+int vorbis_bitrate_managed(vorbis_block *vb){
+  vorbis_dsp_state      *vd=vb->vd;
+  private_state         *b=(private_state*)vd->backend_state;
+  bitrate_manager_state *bm=&b->bms;
+
+  if(bm && bm->managed)return(1);
+  return(0);
+}
+
+/* finish taking in the block we just processed */
+int vorbis_bitrate_addblock(vorbis_block *vb){
+  vorbis_block_internal *vbi=(vorbis_block_internal*)vb->internal;
+  vorbis_dsp_state      *vd=vb->vd;
+  private_state         *b=(private_state*)vd->backend_state;
+  bitrate_manager_state *bm=&b->bms;
+  vorbis_info           *vi=vd->vi;
+  codec_setup_info      *ci=(codec_setup_info*)vi->codec_setup;
+  bitrate_manager_info  *bi=&ci->bi;
+
+  int  choice = (int) rint(bm->avgfloat);
+  long this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+  long min_target_bits=(vb->W?bm->min_bitsper*bm->short_per_long:bm->min_bitsper);
+  long max_target_bits=(vb->W?bm->max_bitsper*bm->short_per_long:bm->max_bitsper);
+  int  samples=ci->blocksizes[vb->W]>>1;
+  long desired_fill = (long) (bi->reservoir_bits*bi->reservoir_bias);
+  if(!bm->managed){
+    /* not a bitrate managed stream, but for API simplicity, we'll
+       buffer the packet to keep the code path clean */
+
+    if(bm->vb)return(-1); /* one has been submitted without
+                             being claimed */
+    bm->vb=vb;
+    return(0);
+  }
+
+  bm->vb=vb;
+
+  /* look ahead for avg floater */
+  if(bm->avg_bitsper>0){
+    double slew=0.;
+    long avg_target_bits=(vb->W?bm->avg_bitsper*bm->short_per_long:bm->avg_bitsper);
+    double slewlimit= 15./bi->slew_damp;
+
+    /* choosing a new floater:
+       if we're over target, we slew down
+       if we're under target, we slew up
+
+       choose slew as follows: look through packetblobs of this frame
+       and set slew as the first in the appropriate direction that
+       gives us the slew we want.  This may mean no slew if delta is
+       already favorable.
+
+       Then limit slew to slew max */
+
+    if(bm->avg_reservoir+(this_bits-avg_target_bits)>desired_fill){
+      while(choice>0 && this_bits>avg_target_bits &&
+            bm->avg_reservoir+(this_bits-avg_target_bits)>desired_fill){
+        choice--;
+        this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+      }
+    }else if(bm->avg_reservoir+(this_bits-avg_target_bits)<desired_fill){
+      while(choice+1<PACKETBLOBS && this_bits<avg_target_bits &&
+            bm->avg_reservoir+(this_bits-avg_target_bits)<desired_fill){
+        choice++;
+        this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+      }
+    }
+
+    slew=rint(choice-bm->avgfloat)/samples*vi->rate;
+    if(slew<-slewlimit)slew=-slewlimit;
+    if(slew>slewlimit)slew=slewlimit;
+    choice = (int) rint(bm->avgfloat+= slew/vi->rate*samples);
+    this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+  }
+
+
+
+  /* enforce min(if used) on the current floater (if used) */
+  if(bm->min_bitsper>0){
+    /* do we need to force the bitrate up? */
+    if(this_bits<min_target_bits){
+      while(bm->minmax_reservoir-(min_target_bits-this_bits)<0){
+        choice++;
+        if(choice>=PACKETBLOBS)break;
+        this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+      }
+    }
+  }
+
+  /* enforce max (if used) on the current floater (if used) */
+  if(bm->max_bitsper>0){
+    /* do we need to force the bitrate down? */
+    if(this_bits>max_target_bits){
+      while(bm->minmax_reservoir+(this_bits-max_target_bits)>bi->reservoir_bits){
+        choice--;
+        if(choice<0)break;
+        this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+      }
+    }
+  }
+
+  /* Choice of packetblobs now made based on floater, and min/max
+     requirements. Now boundary check extreme choices */
+
+  if(choice<0){
+    /* choosing a smaller packetblob is insufficient to trim bitrate.
+       frame will need to be truncated */
+    long maxsize=(max_target_bits+(bi->reservoir_bits-bm->minmax_reservoir))/8;
+    bm->choice=choice=0;
+
+    if(oggpack_bytes(vbi->packetblob[choice])>maxsize){
+
+      oggpack_writetrunc(vbi->packetblob[choice],maxsize*8);
+      this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+    }
+  }else{
+    long minsize=(min_target_bits-bm->minmax_reservoir+7)/8;
+    if(choice>=PACKETBLOBS)
+      choice=PACKETBLOBS-1;
+
+    bm->choice=choice;
+
+    /* prop up bitrate according to demand. pad this frame out with zeroes */
+    minsize-=oggpack_bytes(vbi->packetblob[choice]);
+    while(minsize-->0)oggpack_write(vbi->packetblob[choice],0,8);
+    this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
+
+  }
+
+  /* now we have the final packet and the final packet size.  Update statistics */
+  /* min and max reservoir */
+  if(bm->min_bitsper>0 || bm->max_bitsper>0){
+
+    if(max_target_bits>0 && this_bits>max_target_bits){
+      bm->minmax_reservoir+=(this_bits-max_target_bits);
+    }else if(min_target_bits>0 && this_bits<min_target_bits){
+      bm->minmax_reservoir+=(this_bits-min_target_bits);
+    }else{
+      /* inbetween; we want to take reservoir toward but not past desired_fill */
+      if(bm->minmax_reservoir>desired_fill){
+        if(max_target_bits>0){ /* logical bulletproofing against initialization state */
+          bm->minmax_reservoir+=(this_bits-max_target_bits);
+          if(bm->minmax_reservoir<desired_fill)bm->minmax_reservoir=desired_fill;
+        }else{
+          bm->minmax_reservoir=desired_fill;
+        }
+      }else{
+        if(min_target_bits>0){ /* logical bulletproofing against initialization state */
+          bm->minmax_reservoir+=(this_bits-min_target_bits);
+          if(bm->minmax_reservoir>desired_fill)bm->minmax_reservoir=desired_fill;
+        }else{
+          bm->minmax_reservoir=desired_fill;
+        }
+      }
+    }
+  }
+
+  /* avg reservoir */
+  if(bm->avg_bitsper>0){
+    long avg_target_bits=(vb->W?bm->avg_bitsper*bm->short_per_long:bm->avg_bitsper);
+    bm->avg_reservoir+=this_bits-avg_target_bits;
+  }
+
+  return(0);
+}
+
+int vorbis_bitrate_flushpacket(vorbis_dsp_state *vd,ogg_packet *op){
+  private_state         *b=(private_state*)vd->backend_state;
+  bitrate_manager_state *bm=&b->bms;
+  vorbis_block          *vb=bm->vb;
+  int                    choice=PACKETBLOBS/2;
+  if(!vb)return 0;
+
+  if(op){
+    vorbis_block_internal *vbi=(vorbis_block_internal*)vb->internal;
+
+    if(vorbis_bitrate_managed(vb))
+      choice=bm->choice;
+
+    op->packet=oggpack_get_buffer(vbi->packetblob[choice]);
+    op->bytes=oggpack_bytes(vbi->packetblob[choice]);
+    op->b_o_s=0;
+    op->e_o_s=vb->eofflag;
+    op->granulepos=vb->granulepos;
+    op->packetno=vb->sequence; /* for sake of completeness */
+  }
+
+  bm->vb=0;
+  return(1);
+}

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/bitrate.h

@@ -0,0 +1,59 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: bitrate tracking and management
+ last mod: $Id: bitrate.h 13293 2007-07-24 00:09:47Z xiphmont $
+
+ ********************************************************************/
+
+#ifndef _V_BITRATE_H_
+#define _V_BITRATE_H_
+
+#include "../../codec.h"
+#include "codec_internal.h"
+#include "os.h"
+
+/* encode side bitrate tracking */
+typedef struct bitrate_manager_state {
+  int            managed;
+
+  long           avg_reservoir;
+  long           minmax_reservoir;
+  long           avg_bitsper;
+  long           min_bitsper;
+  long           max_bitsper;
+
+  long           short_per_long;
+  double         avgfloat;
+
+  vorbis_block  *vb;
+  int            choice;
+} bitrate_manager_state;
+
+typedef struct bitrate_manager_info{
+  long           avg_rate;
+  long           min_rate;
+  long           max_rate;
+  long           reservoir_bits;
+  double         reservoir_bias;
+
+  double         slew_damp;
+
+} bitrate_manager_info;
+
+extern void vorbis_bitrate_init(vorbis_info *vi,bitrate_manager_state *bs);
+extern void vorbis_bitrate_clear(bitrate_manager_state *bs);
+extern int vorbis_bitrate_managed(vorbis_block *vb);
+extern int vorbis_bitrate_addblock(vorbis_block *vb);
+extern int vorbis_bitrate_flushpacket(vorbis_dsp_state *vd, ogg_packet *op);
+
+#endif

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/codebook.c

@@ -0,0 +1,479 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: basic codebook pack/unpack/code/decode operations
+ last mod: $Id: codebook.c 17553 2010-10-21 17:54:26Z tterribe $
+
+ ********************************************************************/
+
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+#include "../../ogg.h"
+#include "../../codec.h"
+#include "codebook.h"
+#include "scales.h"
+#include "misc.h"
+#include "os.h"
+
+/* packs the given codebook into the bitstream **************************/
+
+int vorbis_staticbook_pack(const static_codebook *c,oggpack_buffer *opb){
+  long i,j;
+  int ordered=0;
+
+  /* first the basic parameters */
+  oggpack_write(opb,0x564342,24);
+  oggpack_write(opb,c->dim,16);
+  oggpack_write(opb,c->entries,24);
+
+  /* pack the codewords.  There are two packings; length ordered and
+     length random.  Decide between the two now. */
+
+  for(i=1;i<c->entries;i++)
+    if(c->lengthlist[i-1]==0 || c->lengthlist[i]<c->lengthlist[i-1])break;
+  if(i==c->entries)ordered=1;
+
+  if(ordered){
+    /* length ordered.  We only need to say how many codewords of
+       each length.  The actual codewords are generated
+       deterministically */
+
+    long count=0;
+    oggpack_write(opb,1,1);  /* ordered */
+    oggpack_write(opb,c->lengthlist[0]-1,5); /* 1 to 32 */
+
+    for(i=1;i<c->entries;i++){
+      long thisx=c->lengthlist[i];
+      long last=c->lengthlist[i-1];
+      if(thisx>last){
+	for(j=last;j<thisx;j++){
+          oggpack_write(opb,i-count,_ilog(c->entries-count));
+          count=i;
+        }
+      }
+    }
+    oggpack_write(opb,i-count,_ilog(c->entries-count));
+
+  }else{
+    /* length random.  Again, we don't code the codeword itself, just
+       the length.  This time, though, we have to encode each length */
+    oggpack_write(opb,0,1);   /* unordered */
+
+    /* algortihmic mapping has use for 'unused entries', which we tag
+       here.  The algorithmic mapping happens as usual, but the unused
+       entry has no codeword. */
+    for(i=0;i<c->entries;i++)
+      if(c->lengthlist[i]==0)break;
+
+    if(i==c->entries){
+      oggpack_write(opb,0,1); /* no unused entries */
+      for(i=0;i<c->entries;i++)
+        oggpack_write(opb,c->lengthlist[i]-1,5);
+    }else{
+      oggpack_write(opb,1,1); /* we have unused entries; thus we tag */
+      for(i=0;i<c->entries;i++){
+        if(c->lengthlist[i]==0){
+          oggpack_write(opb,0,1);
+        }else{
+          oggpack_write(opb,1,1);
+          oggpack_write(opb,c->lengthlist[i]-1,5);
+        }
+      }
+    }
+  }
+
+  /* is the entry number the desired return value, or do we have a
+     mapping? If we have a mapping, what type? */
+  oggpack_write(opb,c->maptype,4);
+  switch(c->maptype){
+  case 0:
+    /* no mapping */
+    break;
+  case 1:case 2:
+    /* implicitly populated value mapping */
+    /* explicitly populated value mapping */
+
+    if(!c->quantlist){
+      /* no quantlist?  error */
+      return(-1);
+    }
+
+    /* values that define the dequantization */
+    oggpack_write(opb,c->q_min,32);
+    oggpack_write(opb,c->q_delta,32);
+    oggpack_write(opb,c->q_quant-1,4);
+    oggpack_write(opb,c->q_sequencep,1);
+
+    {
+      int quantvals;
+      switch(c->maptype){
+      case 1:
+        /* a single column of (c->entries/c->dim) quantized values for
+           building a full value list algorithmically (square lattice) */
+        quantvals=_book_maptype1_quantvals(c);
+        break;
+      case 2:
+        /* every value (c->entries*c->dim total) specified explicitly */
+        quantvals=c->entries*c->dim;
+        break;
+      default: /* NOT_REACHABLE */
+        quantvals=-1;
+      }
+
+      /* quantized values */
+      for(i=0;i<quantvals;i++)
+        oggpack_write(opb,labs(c->quantlist[i]),c->q_quant);
+
+    }
+    break;
+  default:
+    /* error case; we don't have any other map types now */
+    return(-1);
+  }
+
+  return(0);
+}
+
+/* unpacks a codebook from the packet buffer into the codebook struct,
+   readies the codebook auxiliary structures for decode *************/
+static_codebook *vorbis_staticbook_unpack(oggpack_buffer *opb){
+  long i,j;
+  static_codebook *s=(static_codebook*)_ogg_calloc(1,sizeof(*s));
+  s->allocedp=1;
+
+  /* make sure alignment is correct */
+  if(oggpack_read(opb,24)!=0x564342)goto _eofout;
+
+  /* first the basic parameters */
+  s->dim=oggpack_read(opb,16);
+  s->entries=oggpack_read(opb,24);
+  if(s->entries==-1)goto _eofout;
+
+  if(_ilog(s->dim)+_ilog(s->entries)>24)goto _eofout;
+
+  /* codeword ordering.... length ordered or unordered? */
+  switch((int)oggpack_read(opb,1)){
+  case 0:{
+    long unused;
+    /* allocated but unused entries? */
+    unused=oggpack_read(opb,1);
+    if((s->entries*(unused?1:5)+7)>>3>opb->storage-oggpack_bytes(opb))
+      goto _eofout;
+    /* unordered */
+    s->lengthlist=(long*)_ogg_malloc(sizeof(*s->lengthlist)*s->entries);
+
+    /* allocated but unused entries? */
+    if(unused){
+      /* yes, unused entries */
+
+      for(i=0;i<s->entries;i++){
+        if(oggpack_read(opb,1)){
+          long num=oggpack_read(opb,5);
+          if(num==-1)goto _eofout;
+          s->lengthlist[i]=num+1;
+        }else
+          s->lengthlist[i]=0;
+      }
+    }else{
+      /* all entries used; no tagging */
+      for(i=0;i<s->entries;i++){
+        long num=oggpack_read(opb,5);
+        if(num==-1)goto _eofout;
+        s->lengthlist[i]=num+1;
+      }
+    }
+
+    break;
+  }
+  case 1:
+    /* ordered */
+    {
+      long length=oggpack_read(opb,5)+1;
+      if(length==0)goto _eofout;
+      s->lengthlist=(long*)_ogg_malloc(sizeof(*s->lengthlist)*s->entries);
+
+      for(i=0;i<s->entries;){
+        long num=oggpack_read(opb,_ilog(s->entries-i));
+        if(num==-1)goto _eofout;
+        if(length>32 || num>s->entries-i ||
+           (num>0 && (num-1)>>(length-1)>1)){
+          goto _errout;
+        }
+        if(length>32)goto _errout;
+        for(j=0;j<num;j++,i++)
+          s->lengthlist[i]=length;
+        length++;
+      }
+    }
+    break;
+  default:
+    /* EOF */
+    goto _eofout;
+  }
+
+  /* Do we have a mapping to unpack? */
+  switch((s->maptype=oggpack_read(opb,4))){
+  case 0:
+    /* no mapping */
+    break;
+  case 1: case 2:
+    /* implicitly populated value mapping */
+    /* explicitly populated value mapping */
+
+    s->q_min=oggpack_read(opb,32);
+    s->q_delta=oggpack_read(opb,32);
+    s->q_quant=oggpack_read(opb,4)+1;
+    s->q_sequencep=oggpack_read(opb,1);
+    if(s->q_sequencep==-1)goto _eofout;
+
+    {
+      int quantvals=0;
+      switch(s->maptype){
+      case 1:
+        quantvals=(s->dim==0?0:_book_maptype1_quantvals(s));
+        break;
+      case 2:
+        quantvals=s->entries*s->dim;
+        break;
+      }
+
+      /* quantized values */
+      if(((quantvals * s->q_quant + 7) >> 3) > opb->storage-oggpack_bytes(opb))
+        goto _eofout;
+      s->quantlist=(long*)_ogg_malloc(sizeof(*s->quantlist)*quantvals);
+      for(i=0;i<quantvals;i++)
+        s->quantlist[i]=oggpack_read(opb,s->q_quant);
+
+      if(quantvals&&s->quantlist[quantvals-1]==-1)goto _eofout;
+    }
+    break;
+  default:
+    goto _errout;
+  }
+
+  /* all set */
+  return(s);
+
+ _errout:
+ _eofout:
+  vorbis_staticbook_destroy(s);
+  return(NULL);
+}
+
+/* returns the number of bits ************************************************/
+int vorbis_book_encode(codebook *book, int a, oggpack_buffer *b){
+  if(a<0 || a>=book->c->entries)return(0);
+  oggpack_write(b,book->codelist[a],book->c->lengthlist[a]);
+  return(book->c->lengthlist[a]);
+}
+
+/* the 'eliminate the decode tree' optimization actually requires the
+   codewords to be MSb first, not LSb.  This is an annoying inelegancy
+   (and one of the first places where carefully thought out design
+   turned out to be wrong; Vorbis II and future Ogg codecs should go
+   to an MSb bitpacker), but not actually the huge hit it appears to
+   be.  The first-stage decode table catches most words so that
+   bitreverse is not in the main execution path. */
+
+static ogg_uint32_t bitreverse(ogg_uint32_t x){
+  x=    ((x>>16)&0x0000ffff) | ((x<<16)&0xffff0000);
+  x=    ((x>> 8)&0x00ff00ff) | ((x<< 8)&0xff00ff00);
+  x=    ((x>> 4)&0x0f0f0f0f) | ((x<< 4)&0xf0f0f0f0);
+  x=    ((x>> 2)&0x33333333) | ((x<< 2)&0xcccccccc);
+  return((x>> 1)&0x55555555) | ((x<< 1)&0xaaaaaaaa);
+}
+
+STIN long decode_packed_entry_number(codebook *book, oggpack_buffer *b){
+  int  read=book->dec_maxlength;
+  long lo,hi;
+  long lok = oggpack_look(b,book->dec_firsttablen);
+
+  if (lok >= 0) {
+    long entry = book->dec_firsttable[lok];
+    if(entry&0x80000000UL){
+      lo=(entry>>15)&0x7fff;
+      hi=book->used_entries-(entry&0x7fff);
+    }else{
+      oggpack_adv(b, book->dec_codelengths[entry-1]);
+      return(entry-1);
+    }
+  }else{
+    lo=0;
+    hi=book->used_entries;
+  }
+
+  lok = oggpack_look(b, read);
+
+  while(lok<0 && read>1)
+    lok = oggpack_look(b, --read);
+  if(lok<0)return -1;
+
+  /* bisect search for the codeword in the ordered list */
+  {
+    ogg_uint32_t testword=bitreverse((ogg_uint32_t)lok);
+
+    while(hi-lo>1){
+      long p=(hi-lo)>>1;
+      long test=book->codelist[lo+p]>testword;
+      lo+=p&(test-1);
+      hi-=p&(-test);
+      }
+
+    if(book->dec_codelengths[lo]<=read){
+      oggpack_adv(b, book->dec_codelengths[lo]);
+      return(lo);
+    }
+  }
+
+  oggpack_adv(b, read);
+
+  return(-1);
+}
+
+/* Decode side is specced and easier, because we don't need to find
+   matches using different criteria; we simply read and map.  There are
+   two things we need to do 'depending':
+
+   We may need to support interleave.  We don't really, but it's
+   convenient to do it here rather than rebuild the vector later.
+
+   Cascades may be additive or multiplicitive; this is not inherent in
+   the codebook, but set in the code using the codebook.  Like
+   interleaving, it's easiest to do it here.
+   addmul==0 -> declarative (set the value)
+   addmul==1 -> additive
+   addmul==2 -> multiplicitive */
+
+/* returns the [original, not compacted] entry number or -1 on eof *********/
+long vorbis_book_decode(codebook *book, oggpack_buffer *b){
+  if(book->used_entries>0){
+    long packed_entry=decode_packed_entry_number(book,b);
+    if(packed_entry>=0)
+      return(book->dec_index[packed_entry]);
+  }
+
+  /* if there's no dec_index, the codebook unpacking isn't collapsed */
+  return(-1);
+}
+
+/* returns 0 on OK or -1 on eof *************************************/
+long vorbis_book_decodevs_add(codebook *book,float *a,oggpack_buffer *b,int n){
+  if(book->used_entries>0){
+    int step=n/book->dim;
+    long *entry = (long*)alloca(sizeof(*entry)*step);
+    float **t = (float**)alloca(sizeof(*t)*step);
+    int i,j,o;
+
+    for (i = 0; i < step; i++) {
+      entry[i]=decode_packed_entry_number(book,b);
+      if(entry[i]==-1)return(-1);
+      t[i] = book->valuelist+entry[i]*book->dim;
+    }
+    for(i=0,o=0;i<book->dim;i++,o+=step)
+      for (j=0;j<step;j++)
+        a[o+j]+=t[j][i];
+  }
+  return(0);
+}
+
+long vorbis_book_decodev_add(codebook *book,float *a,oggpack_buffer *b,int n){
+  if(book->used_entries>0){
+    int i,j,entry;
+    float *t;
+
+    if(book->dim>8){
+      for(i=0;i<n;){
+        entry = decode_packed_entry_number(book,b);
+        if(entry==-1)return(-1);
+        t     = book->valuelist+entry*book->dim;
+        for (j=0;j<book->dim;)
+          a[i++]+=t[j++];
+      }
+    }else{
+      for(i=0;i<n;){
+        entry = decode_packed_entry_number(book,b);
+        if(entry==-1)return(-1);
+        t     = book->valuelist+entry*book->dim;
+        j=0;
+        switch((int)book->dim){
+        case 8:
+          a[i++]+=t[j++];
+        case 7:
+          a[i++]+=t[j++];
+        case 6:
+          a[i++]+=t[j++];
+        case 5:
+          a[i++]+=t[j++];
+        case 4:
+          a[i++]+=t[j++];
+        case 3:
+          a[i++]+=t[j++];
+        case 2:
+          a[i++]+=t[j++];
+        case 1:
+          a[i++]+=t[j++];
+        case 0:
+          break;
+        }
+      }
+    }
+  }
+  return(0);
+}
+
+long vorbis_book_decodev_set(codebook *book,float *a,oggpack_buffer *b,int n){
+  if(book->used_entries>0){
+    int i,j,entry;
+    float *t;
+
+    for(i=0;i<n;){
+      entry = decode_packed_entry_number(book,b);
+      if(entry==-1)return(-1);
+      t     = book->valuelist+entry*book->dim;
+      for (j=0;j<book->dim;)
+        a[i++]=t[j++];
+    }
+  }else{
+    int i,j;
+
+    for(i=0;i<n;){
+      for (j=0;j<book->dim;)
+        a[i++]=0.f;
+    }
+  }
+  return(0);
+}
+
+long vorbis_book_decodevv_add(codebook *book,float **a,long offset,int ch,
+                              oggpack_buffer *b,int n){
+
+  long i,j,entry;
+  int chptr=0;
+  if(book->used_entries>0){
+    for(i=offset/ch;i<(offset+n)/ch;){
+      entry = decode_packed_entry_number(book,b);
+      if(entry==-1)return(-1);
+      {
+        const float *t = book->valuelist+entry*book->dim;
+        for (j=0;j<book->dim;j++){
+          a[chptr++][i]+=t[j];
+          if(chptr==ch){
+            chptr=0;
+            i++;
+          }
+        }
+      }
+    }
+  }
+  return(0);
+}

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/codebook.h

@@ -0,0 +1,119 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: basic shared codebook operations
+ last mod: $Id: codebook.h 17030 2010-03-25 06:52:55Z xiphmont $
+
+ ********************************************************************/
+
+#ifndef _V_CODEBOOK_H_
+#define _V_CODEBOOK_H_
+
+#include "../../ogg.h"
+
+/* This structure encapsulates huffman and VQ style encoding books; it
+   doesn't do anything specific to either.
+
+   valuelist/quantlist are nonNULL (and q_* significant) only if
+   there's entry->value mapping to be done.
+
+   If encode-side mapping must be done (and thus the entry needs to be
+   hunted), the auxiliary encode pointer will point to a decision
+   tree.  This is true of both VQ and huffman, but is mostly useful
+   with VQ.
+
+*/
+
+typedef struct static_codebook{
+  long   dim;            /* codebook dimensions (elements per vector) */
+  long   entries;        /* codebook entries */
+  long  *lengthlist;     /* codeword lengths in bits */
+
+  /* mapping ***************************************************************/
+  int    maptype;        /* 0=none
+                            1=implicitly populated values from map column
+                            2=listed arbitrary values */
+
+  /* The below does a linear, single monotonic sequence mapping. */
+  long     q_min;       /* packed 32 bit float; quant value 0 maps to minval */
+  long     q_delta;     /* packed 32 bit float; val 1 - val 0 == delta */
+  int      q_quant;     /* bits: 0 < quant <= 16 */
+  int      q_sequencep; /* bitflag */
+
+  long     *quantlist;  /* map == 1: (int)(entries^(1/dim)) element column map
+                           map == 2: list of dim*entries quantized entry vals
+                        */
+  int allocedp;
+} static_codebook;
+
+typedef struct codebook{
+  long dim;           /* codebook dimensions (elements per vector) */
+  long entries;       /* codebook entries */
+  long used_entries;  /* populated codebook entries */
+  const static_codebook *c;
+
+  /* for encode, the below are entry-ordered, fully populated */
+  /* for decode, the below are ordered by bitreversed codeword and only
+     used entries are populated */
+  float        *valuelist;  /* list of dim*entries actual entry values */
+  ogg_uint32_t *codelist;   /* list of bitstream codewords for each entry */
+
+  int          *dec_index;  /* only used if sparseness collapsed */
+  char         *dec_codelengths;
+  ogg_uint32_t *dec_firsttable;
+  int           dec_firsttablen;
+  int           dec_maxlength;
+
+  /* The current encoder uses only centered, integer-only lattice books. */
+  int           quantvals;
+  int           minval;
+  int           delta;
+} codebook;
+
+extern void vorbis_staticbook_destroy(static_codebook *b);
+extern int vorbis_book_init_encode(codebook *dest,const static_codebook *source);
+extern int vorbis_book_init_decode(codebook *dest,const static_codebook *source);
+extern void vorbis_book_clear(codebook *b);
+
+extern float *_book_unquantize(const static_codebook *b,int n,int *map);
+extern float *_book_logdist(const static_codebook *b,float *vals);
+extern float _float32_unpack(long val);
+extern long   _float32_pack(float val);
+extern int  _best(codebook *book, float *a, int step);
+extern int _ilog(unsigned int v);
+extern long _book_maptype1_quantvals(const static_codebook *b);
+
+extern int vorbis_book_besterror(codebook *book,float *a,int step,int addmul);
+extern long vorbis_book_codeword(codebook *book,int entry);
+extern long vorbis_book_codelen(codebook *book,int entry);
+
+
+
+extern int vorbis_staticbook_pack(const static_codebook *c,oggpack_buffer *b);
+extern static_codebook *vorbis_staticbook_unpack(oggpack_buffer *b);
+
+extern int vorbis_book_encode(codebook *book, int a, oggpack_buffer *b);
+
+extern long vorbis_book_decode(codebook *book, oggpack_buffer *b);
+extern long vorbis_book_decodevs_add(codebook *book, float *a,
+                                     oggpack_buffer *b,int n);
+extern long vorbis_book_decodev_set(codebook *book, float *a,
+                                    oggpack_buffer *b,int n);
+extern long vorbis_book_decodev_add(codebook *book, float *a,
+                                    oggpack_buffer *b,int n);
+extern long vorbis_book_decodevv_add(codebook *book, float **a,
+                                     long off,int ch,
+                                    oggpack_buffer *b,int n);
+
+
+
+#endif

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/codec_internal.h

@@ -0,0 +1,187 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: libvorbis codec headers
+ last mod: $Id: codec_internal.h 16227 2009-07-08 06:58:46Z xiphmont $
+
+ ********************************************************************/
+
+#ifndef _V_CODECI_H_
+#define _V_CODECI_H_
+
+#include "envelope.h"
+#include "codebook.h"
+
+#define BLOCKTYPE_IMPULSE    0
+#define BLOCKTYPE_PADDING    1
+#define BLOCKTYPE_TRANSITION 0
+#define BLOCKTYPE_LONG       1
+
+#define PACKETBLOBS 15
+
+typedef struct vorbis_block_internal{
+  float  **pcmdelay;  /* this is a pointer into local storage */
+  float  ampmax;
+  int    blocktype;
+
+  oggpack_buffer *packetblob[PACKETBLOBS]; /* initialized, must be freed;
+                                              blob [PACKETBLOBS/2] points to
+                                              the oggpack_buffer in the
+                                              main vorbis_block */
+} vorbis_block_internal;
+
+typedef void vorbis_look_floor;
+typedef void vorbis_look_residue;
+typedef void vorbis_look_transform;
+
+/* mode ************************************************************/
+typedef struct {
+  int blockflag;
+  int windowtype;
+  int transformtype;
+  int mapping;
+} vorbis_info_mode;
+
+typedef void vorbis_info_floor;
+typedef void vorbis_info_residue;
+typedef void vorbis_info_mapping;
+
+#include "psy.h"
+#include "bitrate.h"
+
+static int ilog(unsigned int v){
+  int ret=0;
+  while(v){
+    ret++;
+    v>>=1;
+  }
+  return(ret);
+}
+
+static int ilog2(unsigned int v){
+  int ret=0;
+  if(v)--v;
+  while(v){
+    ret++;
+    v>>=1;
+  }
+  return(ret);
+}
+
+
+typedef struct private_state {
+  /* local lookup storage */
+  envelope_lookup        *ve; /* envelope lookup */
+  int                     window[2];
+  vorbis_look_transform **transform[2];    /* block, type */
+  drft_lookup             fft_look[2];
+
+  int                     modebits;
+  vorbis_look_floor     **flr;
+  vorbis_look_residue   **residue;
+  vorbis_look_psy        *psy;
+  vorbis_look_psy_global *psy_g_look;
+
+  /* local storage, only used on the encoding side.  This way the
+     application does not need to worry about freeing some packets'
+     memory and not others'; packet storage is always tracked.
+     Cleared next call to a _dsp_ function */
+  unsigned char *header;
+  unsigned char *header1;
+  unsigned char *header2;
+
+  bitrate_manager_state bms;
+
+  ogg_int64_t sample_count;
+} private_state;
+
+/* codec_setup_info contains all the setup information specific to the
+   specific compression/decompression mode in progress (eg,
+   psychoacoustic settings, channel setup, options, codebook
+   etc).
+*********************************************************************/
+
+#include "highlevel.h"
+typedef struct codec_setup_info {
+
+  /* Vorbis supports only short and long blocks, but allows the
+     encoder to choose the sizes */
+
+  long blocksizes[2];
+
+  /* modes are the primary means of supporting on-the-fly different
+     blocksizes, different channel mappings (LR or M/A),
+     different residue backends, etc.  Each mode consists of a
+     blocksize flag and a mapping (along with the mapping setup */
+
+  int        modes;
+  int        maps;
+  int        floors;
+  int        residues;
+  int        books;
+  int        psys;     /* encode only */
+
+  vorbis_info_mode       *mode_param[64];
+  int                     map_type[64];
+  vorbis_info_mapping    *map_param[64];
+  int                     floor_type[64];
+  vorbis_info_floor      *floor_param[64];
+  int                     residue_type[64];
+  vorbis_info_residue    *residue_param[64];
+  static_codebook        *book_param[256];
+  codebook               *fullbooks;
+
+  vorbis_info_psy        *psy_param[4]; /* encode only */
+  vorbis_info_psy_global psy_g_param;
+
+  bitrate_manager_info   bi;
+  highlevel_encode_setup hi; /* used only by vorbisenc.c.  It's a
+                                highly redundant structure, but
+                                improves clarity of program flow. */
+  int         halfrate_flag; /* painless downsample for decode */
+} codec_setup_info;
+
+extern vorbis_look_psy_global *_vp_global_look(vorbis_info *vi);
+extern void _vp_global_free(vorbis_look_psy_global *look);
+
+
+
+typedef struct {
+  int sorted_index[VIF_POSIT+2];
+  int forward_index[VIF_POSIT+2];
+  int reverse_index[VIF_POSIT+2];
+
+  int hineighbor[VIF_POSIT];
+  int loneighbor[VIF_POSIT];
+  int posts;
+
+  int n;
+  int quant_q;
+  vorbis_info_floor1 *vi;
+
+  long phrasebits;
+  long postbits;
+  long frames;
+} vorbis_look_floor1;
+
+
+
+extern int *floor1_fit(vorbis_block *vb,vorbis_look_floor1 *look,
+                          const float *logmdct,   /* in */
+                          const float *logmask);
+extern int *floor1_interpolate_fit(vorbis_block *vb,vorbis_look_floor1 *look,
+                          int *A,int *B,
+                          int del);
+extern int floor1_encode(oggpack_buffer *opb,vorbis_block *vb,
+                  vorbis_look_floor1 *look,
+                  int *post,int *ilogmask);
+#endif

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/envelope.c

@@ -0,0 +1,375 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: PCM data envelope analysis
+ last mod: $Id: envelope.c 16227 2009-07-08 06:58:46Z xiphmont $
+
+ ********************************************************************/
+
+#include <stdlib.h>
+#include <string.h>
+#include <stdio.h>
+#include <math.h>
+#include "../../ogg.h"
+#include "../../codec.h"
+#include "codec_internal.h"
+
+#include "os.h"
+#include "scales.h"
+#include "envelope.h"
+#include "mdct.h"
+#include "misc.h"
+
+void _ve_envelope_init(envelope_lookup *e,vorbis_info *vi){
+  codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
+  vorbis_info_psy_global *gi=&ci->psy_g_param;
+  int ch=vi->channels;
+  int i,j;
+  int n=e->winlength=128;
+  e->searchstep=64; /* not random */
+
+  e->minenergy=gi->preecho_minenergy;
+  e->ch=ch;
+  e->storage=128;
+  e->cursor=ci->blocksizes[1]/2;
+  e->mdct_win=(float*)_ogg_calloc(n,sizeof(*e->mdct_win));
+  mdct_init(&e->mdct,n);
+
+  for(i=0;i<n;i++){
+    e->mdct_win[i]=sin(i/(n-1.)*M_PI);
+    e->mdct_win[i]*=e->mdct_win[i];
+  }
+
+  /* magic follows */
+  e->band[0].begin=2;  e->band[0].end=4;
+  e->band[1].begin=4;  e->band[1].end=5;
+  e->band[2].begin=6;  e->band[2].end=6;
+  e->band[3].begin=9;  e->band[3].end=8;
+  e->band[4].begin=13;  e->band[4].end=8;
+  e->band[5].begin=17;  e->band[5].end=8;
+  e->band[6].begin=22;  e->band[6].end=8;
+
+  for(j=0;j<VE_BANDS;j++){
+    n=e->band[j].end;
+    e->band[j].window=(float*)_ogg_malloc(n*sizeof(*e->band[0].window));
+    for(i=0;i<n;i++){
+      e->band[j].window[i]=sin((i+.5)/n*M_PI);
+      e->band[j].total+=e->band[j].window[i];
+    }
+    e->band[j].total=1./e->band[j].total;
+  }
+
+  e->filter=(envelope_filter_state*)_ogg_calloc(VE_BANDS*ch,sizeof(*e->filter));
+  e->mark=(int*)_ogg_calloc(e->storage,sizeof(*e->mark));
+
+}
+
+void _ve_envelope_clear(envelope_lookup *e){
+  int i;
+  mdct_clear(&e->mdct);
+  for(i=0;i<VE_BANDS;i++)
+    _ogg_free(e->band[i].window);
+  _ogg_free(e->mdct_win);
+  _ogg_free(e->filter);
+  _ogg_free(e->mark);
+  memset(e,0,sizeof(*e));
+}
+
+/* fairly straight threshhold-by-band based until we find something
+   that works better and isn't patented. */
+
+static int _ve_amp(envelope_lookup *ve,
+                   vorbis_info_psy_global *gi,
+                   float *data,
+                   envelope_band *bands,
+                   envelope_filter_state *filters){
+  long n=ve->winlength;
+  int ret=0;
+  long i,j;
+  float decay;
+
+  /* we want to have a 'minimum bar' for energy, else we're just
+     basing blocks on quantization noise that outweighs the signal
+     itself (for low power signals) */
+
+  float minV=ve->minenergy;
+  float *vec=(float*) alloca(n*sizeof(*vec));
+
+  /* stretch is used to gradually lengthen the number of windows
+     considered prevoius-to-potential-trigger */
+  int stretch=max(VE_MINSTRETCH,ve->stretch/2);
+  float penalty=gi->stretch_penalty-(ve->stretch/2-VE_MINSTRETCH);
+  if(penalty<0.f)penalty=0.f;
+  if(penalty>gi->stretch_penalty)penalty=gi->stretch_penalty;
+
+  /*_analysis_output_always("lpcm",seq2,data,n,0,0,
+    totalshift+pos*ve->searchstep);*/
+
+ /* window and transform */
+  for(i=0;i<n;i++)
+    vec[i]=data[i]*ve->mdct_win[i];
+  mdct_forward(&ve->mdct,vec,vec);
+
+  /*_analysis_output_always("mdct",seq2,vec,n/2,0,1,0); */
+
+  /* near-DC spreading function; this has nothing to do with
+     psychoacoustics, just sidelobe leakage and window size */
+  {
+    float temp=vec[0]*vec[0]+.7*vec[1]*vec[1]+.2*vec[2]*vec[2];
+    int ptr=filters->nearptr;
+
+    /* the accumulation is regularly refreshed from scratch to avoid
+       floating point creep */
+    if(ptr==0){
+      decay=filters->nearDC_acc=filters->nearDC_partialacc+temp;
+      filters->nearDC_partialacc=temp;
+    }else{
+      decay=filters->nearDC_acc+=temp;
+      filters->nearDC_partialacc+=temp;
+    }
+    filters->nearDC_acc-=filters->nearDC[ptr];
+    filters->nearDC[ptr]=temp;
+
+    decay*=(1./(VE_NEARDC+1));
+    filters->nearptr++;
+    if(filters->nearptr>=VE_NEARDC)filters->nearptr=0;
+    decay=todB(&decay)*.5-15.f;
+  }
+
+  /* perform spreading and limiting, also smooth the spectrum.  yes,
+     the MDCT results in all real coefficients, but it still *behaves*
+     like real/imaginary pairs */
+  for(i=0;i<n/2;i+=2){
+    float val=vec[i]*vec[i]+vec[i+1]*vec[i+1];
+    val=todB(&val)*.5f;
+    if(val<decay)val=decay;
+    if(val<minV)val=minV;
+    vec[i>>1]=val;
+    decay-=8.;
+  }
+
+  /*_analysis_output_always("spread",seq2++,vec,n/4,0,0,0);*/
+
+  /* perform preecho/postecho triggering by band */
+  for(j=0;j<VE_BANDS;j++){
+    float acc=0.;
+    float valmax,valmin;
+
+    /* accumulate amplitude */
+    for(i=0;i<bands[j].end;i++)
+      acc+=vec[i+bands[j].begin]*bands[j].window[i];
+
+    acc*=bands[j].total;
+
+    /* convert amplitude to delta */
+    {
+      int p,thisx=filters[j].ampptr;
+      float postmax,postmin,premax=-99999.f,premin=99999.f;
+
+      p=thisx;
+      p--;
+      if(p<0)p+=VE_AMP;
+      postmax=max(acc,filters[j].ampbuf[p]);
+      postmin=min(acc,filters[j].ampbuf[p]);
+
+      for(i=0;i<stretch;i++){
+        p--;
+        if(p<0)p+=VE_AMP;
+        premax=max(premax,filters[j].ampbuf[p]);
+        premin=min(premin,filters[j].ampbuf[p]);
+      }
+
+      valmin=postmin-premin;
+      valmax=postmax-premax;
+
+      /*filters[j].markers[pos]=valmax;*/
+      filters[j].ampbuf[thisx]=acc;
+      filters[j].ampptr++;
+      if(filters[j].ampptr>=VE_AMP)filters[j].ampptr=0;
+    }
+
+    /* look at min/max, decide trigger */
+    if(valmax>gi->preecho_thresh[j]+penalty){
+      ret|=1;
+      ret|=4;
+    }
+    if(valmin<gi->postecho_thresh[j]-penalty)ret|=2;
+  }
+
+  return(ret);
+}
+
+#if 0
+static int seq=0;
+static ogg_int64_t totalshift=-1024;
+#endif
+
+long _ve_envelope_search(vorbis_dsp_state *v){
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci=(codec_setup_info *)vi->codec_setup;
+  vorbis_info_psy_global *gi=&ci->psy_g_param;
+  envelope_lookup *ve=((private_state *)(v->backend_state))->ve;
+  long i,j;
+
+  int first=ve->current/ve->searchstep;
+  int last=v->pcm_current/ve->searchstep-VE_WIN;
+  if(first<0)first=0;
+
+  /* make sure we have enough storage to match the PCM */
+  if(last+VE_WIN+VE_POST>ve->storage){
+    ve->storage=last+VE_WIN+VE_POST; /* be sure */
+    ve->mark=(int*)_ogg_realloc(ve->mark,ve->storage*sizeof(*ve->mark));
+  }
+
+  for(j=first;j<last;j++){
+    int ret=0;
+
+    ve->stretch++;
+    if(ve->stretch>VE_MAXSTRETCH*2)
+      ve->stretch=VE_MAXSTRETCH*2;
+
+    for(i=0;i<ve->ch;i++){
+      float *pcm=v->pcm[i]+ve->searchstep*(j);
+      ret|=_ve_amp(ve,gi,pcm,ve->band,ve->filter+i*VE_BANDS);
+    }
+
+    ve->mark[j+VE_POST]=0;
+    if(ret&1){
+      ve->mark[j]=1;
+      ve->mark[j+1]=1;
+    }
+
+    if(ret&2){
+      ve->mark[j]=1;
+      if(j>0)ve->mark[j-1]=1;
+    }
+
+    if(ret&4)ve->stretch=-1;
+  }
+
+  ve->current=last*ve->searchstep;
+
+  {
+    long centerW=v->centerW;
+    long testW=
+      centerW+
+      ci->blocksizes[v->W]/4+
+      ci->blocksizes[1]/2+
+      ci->blocksizes[0]/4;
+
+    j=ve->cursor;
+
+    while(j<ve->current-(ve->searchstep)){/* account for postecho
+                                             working back one window */
+      if(j>=testW)return(1);
+
+      ve->cursor=j;
+
+      if(ve->mark[j/ve->searchstep]){
+        if(j>centerW){
+
+#if 0
+          if(j>ve->curmark){
+            float *marker=(float*)alloca(v->pcm_current*sizeof(*marker));
+            int l,m;
+            memset(marker,0,sizeof(*marker)*v->pcm_current);
+            fprintf(stderr,"mark! seq=%d, cursor:%fs time:%fs\n",
+                    seq,
+                    (totalshift+ve->cursor)/44100.,
+                    (totalshift+j)/44100.);
+            _analysis_output_always("pcmL",seq,v->pcm[0],v->pcm_current,0,0,totalshift);
+            _analysis_output_always("pcmR",seq,v->pcm[1],v->pcm_current,0,0,totalshift);
+
+            _analysis_output_always("markL",seq,v->pcm[0],j,0,0,totalshift);
+            _analysis_output_always("markR",seq,v->pcm[1],j,0,0,totalshift);
+
+            for(m=0;m<VE_BANDS;m++){
+              char buf[80];
+              sprintf(buf,"delL%d",m);
+              for(l=0;l<last;l++)marker[l*ve->searchstep]=ve->filter[m].markers[l]*.1;
+              _analysis_output_always(buf,seq,marker,v->pcm_current,0,0,totalshift);
+            }
+
+            for(m=0;m<VE_BANDS;m++){
+              char buf[80];
+              sprintf(buf,"delR%d",m);
+              for(l=0;l<last;l++)marker[l*ve->searchstep]=ve->filter[m+VE_BANDS].markers[l]*.1;
+              _analysis_output_always(buf,seq,marker,v->pcm_current,0,0,totalshift);
+            }
+
+            for(l=0;l<last;l++)marker[l*ve->searchstep]=ve->mark[l]*.4;
+            _analysis_output_always("mark",seq,marker,v->pcm_current,0,0,totalshift);
+
+
+            seq++;
+
+          }
+#endif
+
+          ve->curmark=j;
+          if(j>=testW)return(1);
+          return(0);
+        }
+      }
+      j+=ve->searchstep;
+    }
+  }
+
+  return(-1);
+}
+
+int _ve_envelope_mark(vorbis_dsp_state *v){
+  envelope_lookup *ve=((private_state *)(v->backend_state))->ve;
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
+  long centerW=v->centerW;
+  long beginW=centerW-ci->blocksizes[v->W]/4;
+  long endW=centerW+ci->blocksizes[v->W]/4;
+  if(v->W){
+    beginW-=ci->blocksizes[v->lW]/4;
+    endW+=ci->blocksizes[v->nW]/4;
+  }else{
+    beginW-=ci->blocksizes[0]/4;
+    endW+=ci->blocksizes[0]/4;
+  }
+
+  if(ve->curmark>=beginW && ve->curmark<endW)return(1);
+  {
+    long first=beginW/ve->searchstep;
+    long last=endW/ve->searchstep;
+    long i;
+    for(i=first;i<last;i++)
+      if(ve->mark[i])return(1);
+  }
+  return(0);
+}
+
+void _ve_envelope_shift(envelope_lookup *e,long shift){
+  int smallsize=e->current/e->searchstep+VE_POST; /* adjust for placing marks
+                                                     ahead of ve->current */
+  int smallshift=shift/e->searchstep;
+
+  memmove(e->mark,e->mark+smallshift,(smallsize-smallshift)*sizeof(*e->mark));
+
+#if 0
+  for(i=0;i<VE_BANDS*e->ch;i++)
+    memmove(e->filter[i].markers,
+            e->filter[i].markers+smallshift,
+            (1024-smallshift)*sizeof(*(*e->filter).markers));
+  totalshift+=shift;
+#endif
+
+  e->current-=shift;
+  if(e->curmark>=0)
+    e->curmark-=shift;
+  e->cursor-=shift;
+}

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/envelope.h

@@ -0,0 +1,80 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: PCM data envelope analysis and manipulation
+ last mod: $Id: envelope.h 16227 2009-07-08 06:58:46Z xiphmont $
+
+ ********************************************************************/
+
+#ifndef _V_ENVELOPE_
+#define _V_ENVELOPE_
+
+#include "mdct.h"
+
+#define VE_PRE    16
+#define VE_WIN    4
+#define VE_POST   2
+#define VE_AMP    (VE_PRE+VE_POST-1)
+
+#define VE_BANDS  7
+#define VE_NEARDC 15
+
+#define VE_MINSTRETCH 2   /* a bit less than short block */
+#define VE_MAXSTRETCH 12  /* one-third full block */
+
+typedef struct {
+  float ampbuf[VE_AMP];
+  int   ampptr;
+
+  float nearDC[VE_NEARDC];
+  float nearDC_acc;
+  float nearDC_partialacc;
+  int   nearptr;
+
+} envelope_filter_state;
+
+typedef struct {
+  int begin;
+  int end;
+  float *window;
+  float total;
+} envelope_band;
+
+typedef struct {
+  int ch;
+  int winlength;
+  int searchstep;
+  float minenergy;
+
+  mdct_lookup  mdct;
+  float       *mdct_win;
+
+  envelope_band          band[VE_BANDS];
+  envelope_filter_state *filter;
+  int   stretch;
+
+  int                   *mark;
+
+  long storage;
+  long current;
+  long curmark;
+  long cursor;
+} envelope_lookup;
+
+extern void _ve_envelope_init(envelope_lookup *e,vorbis_info *vi);
+extern void _ve_envelope_clear(envelope_lookup *e);
+extern long _ve_envelope_search(vorbis_dsp_state *v);
+extern void _ve_envelope_shift(envelope_lookup *e,long shift);
+extern int  _ve_envelope_mark(vorbis_dsp_state *v);
+
+
+#endif

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/floor0.c

@@ -0,0 +1,223 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: floor backend 0 implementation
+ last mod: $Id: floor0.c 17558 2010-10-22 00:24:41Z tterribe $
+
+ ********************************************************************/
+
+
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+#include "../../ogg.h"
+#include "../../codec.h"
+#include "codec_internal.h"
+#include "registry.h"
+#include "lpc.h"
+#include "lsp.h"
+#include "codebook.h"
+#include "scales.h"
+#include "misc.h"
+#include "os.h"
+
+#include "misc.h"
+#include <stdio.h>
+
+typedef struct {
+  int ln;
+  int  m;
+  int **linearmap;
+  int  n[2];
+
+  vorbis_info_floor0 *vi;
+
+  long bits;
+  long frames;
+} vorbis_look_floor0;
+
+
+/***********************************************/
+
+static void floor0_free_info(vorbis_info_floor *i){
+  vorbis_info_floor0 *info=(vorbis_info_floor0 *)i;
+  if(info){
+    memset(info,0,sizeof(*info));
+    _ogg_free(info);
+  }
+}
+
+static void floor0_free_look(vorbis_look_floor *i){
+  vorbis_look_floor0 *look=(vorbis_look_floor0 *)i;
+  if(look){
+
+    if(look->linearmap){
+
+      if(look->linearmap[0])_ogg_free(look->linearmap[0]);
+      if(look->linearmap[1])_ogg_free(look->linearmap[1]);
+
+      _ogg_free(look->linearmap);
+    }
+    memset(look,0,sizeof(*look));
+    _ogg_free(look);
+  }
+}
+
+static vorbis_info_floor *floor0_unpack (vorbis_info *vi,oggpack_buffer *opb){
+  codec_setup_info     *ci=(codec_setup_info*)vi->codec_setup;
+  int j;
+
+  vorbis_info_floor0 *info=(vorbis_info_floor0*)_ogg_malloc(sizeof(*info));
+  info->order=oggpack_read(opb,8);
+  info->rate=oggpack_read(opb,16);
+  info->barkmap=oggpack_read(opb,16);
+  info->ampbits=oggpack_read(opb,6);
+  info->ampdB=oggpack_read(opb,8);
+  info->numbooks=oggpack_read(opb,4)+1;
+
+  if(info->order<1)goto err_out;
+  if(info->rate<1)goto err_out;
+  if(info->barkmap<1)goto err_out;
+  if(info->numbooks<1)goto err_out;
+
+  for(j=0;j<info->numbooks;j++){
+    info->books[j]=oggpack_read(opb,8);
+    if(info->books[j]<0 || info->books[j]>=ci->books)goto err_out;
+    if(ci->book_param[info->books[j]]->maptype==0)goto err_out;
+    if(ci->book_param[info->books[j]]->dim<1)goto err_out;
+  }
+  return(info);
+
+ err_out:
+  floor0_free_info(info);
+  return(NULL);
+}
+
+/* initialize Bark scale and normalization lookups.  We could do this
+   with static tables, but Vorbis allows a number of possible
+   combinations, so it's best to do it computationally.
+
+   The below is authoritative in terms of defining scale mapping.
+   Note that the scale depends on the sampling rate as well as the
+   linear block and mapping sizes */
+
+static void floor0_map_lazy_init(vorbis_block      *vb,
+                                 vorbis_info_floor *infoX,
+                                 vorbis_look_floor0 *look){
+  if(!look->linearmap[vb->W]){
+    vorbis_dsp_state   *vd=vb->vd;
+    vorbis_info        *vi=vd->vi;
+    codec_setup_info   *ci=(codec_setup_info*)vi->codec_setup;
+    vorbis_info_floor0 *info=(vorbis_info_floor0 *)infoX;
+    int W=vb->W;
+    int n=ci->blocksizes[W]/2,j;
+
+    /* we choose a scaling constant so that:
+       floor(bark(rate/2-1)*C)=mapped-1
+     floor(bark(rate/2)*C)=mapped */
+    float scale=look->ln/toBARK(info->rate/2.f);
+
+    /* the mapping from a linear scale to a smaller bark scale is
+       straightforward.  We do *not* make sure that the linear mapping
+       does not skip bark-scale bins; the decoder simply skips them and
+       the encoder may do what it wishes in filling them.  They're
+       necessary in some mapping combinations to keep the scale spacing
+       accurate */
+    look->linearmap[W]=(int*)_ogg_malloc((n+1)*sizeof(**look->linearmap));
+    for(j=0;j<n;j++){
+      int val=floor( toBARK((info->rate/2.f)/n*j)
+                     *scale); /* bark numbers represent band edges */
+      if(val>=look->ln)val=look->ln-1; /* guard against the approximation */
+      look->linearmap[W][j]=val;
+    }
+    look->linearmap[W][j]=-1;
+    look->n[W]=n;
+  }
+}
+
+static vorbis_look_floor *floor0_look(vorbis_dsp_state* /* vd */,
+                                      vorbis_info_floor *i){
+  vorbis_info_floor0 *info=(vorbis_info_floor0 *)i;
+  vorbis_look_floor0 *look=(vorbis_look_floor0*)_ogg_calloc(1,sizeof(*look));
+  look->m=info->order;
+  look->ln=info->barkmap;
+  look->vi=info;
+
+  look->linearmap=(int**)_ogg_calloc(2,sizeof(*look->linearmap));
+
+  return look;
+}
+
+static void *floor0_inverse1(vorbis_block *vb,vorbis_look_floor *i){
+  vorbis_look_floor0 *look=(vorbis_look_floor0 *)i;
+  vorbis_info_floor0 *info=look->vi;
+  int j,k;
+
+  int ampraw=oggpack_read(&vb->opb,info->ampbits);
+  if(ampraw>0){ /* also handles the -1 out of data case */
+    long maxval=(1<<info->ampbits)-1;
+    float amp=(float)ampraw/maxval*info->ampdB;
+    int booknum=oggpack_read(&vb->opb,_ilog(info->numbooks));
+
+    if(booknum!=-1 && booknum<info->numbooks){ /* be paranoid */
+      codec_setup_info  *ci=(codec_setup_info *)vb->vd->vi->codec_setup;
+      codebook *b=ci->fullbooks+info->books[booknum];
+      float last=0.f;
+
+      /* the additional b->dim is a guard against any possible stack
+         smash; b->dim is provably more than we can overflow the
+         vector */
+      float *lsp=(float*)_vorbis_block_alloc(vb,sizeof(*lsp)*(look->m+b->dim+1));
+
+      for(j=0;j<look->m;j+=b->dim)
+        if(vorbis_book_decodev_set(b,lsp+j,&vb->opb,b->dim)==-1)goto eop;
+      for(j=0;j<look->m;){
+        for(k=0;k<b->dim;k++,j++)lsp[j]+=last;
+        last=lsp[j-1];
+      }
+
+      lsp[look->m]=amp;
+      return(lsp);
+    }
+  }
+ eop:
+  return(NULL);
+}
+
+static int floor0_inverse2(vorbis_block *vb,vorbis_look_floor *i,
+                           void *memo,float *out){
+  vorbis_look_floor0 *look=(vorbis_look_floor0 *)i;
+  vorbis_info_floor0 *info=look->vi;
+
+  floor0_map_lazy_init(vb,info,look);
+
+  if(memo){
+    float *lsp=(float *)memo;
+    float amp=lsp[look->m];
+
+    /* take the coefficients back to a spectral envelope curve */
+    vorbis_lsp_to_curve(out,
+                        look->linearmap[vb->W],
+                        look->n[vb->W],
+                        look->ln,
+                        lsp,look->m,amp,(float)info->ampdB);
+    return(1);
+  }
+  memset(out,0,sizeof(*out)*look->n[vb->W]);
+  return(0);
+}
+
+/* export hooks */
+const vorbis_func_floor floor0_exportbundle={
+  NULL,&floor0_unpack,&floor0_look,&floor0_free_info,
+  &floor0_free_look,&floor0_inverse1,&floor0_inverse2
+};

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/highlevel.h

@@ -0,0 +1,58 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: highlevel encoder setup struct separated out for vorbisenc clarity
+ last mod: $Id: highlevel.h 17195 2010-05-05 21:49:51Z giles $
+
+ ********************************************************************/
+
+typedef struct highlevel_byblocktype {
+  double tone_mask_setting;
+  double tone_peaklimit_setting;
+  double noise_bias_setting;
+  double noise_compand_setting;
+} highlevel_byblocktype;
+
+typedef struct highlevel_encode_setup {
+  int   set_in_stone;
+  const void *setup;
+  double base_setting;
+
+  double impulse_noisetune;
+
+  /* bitrate management below all settable */
+  float  req;
+  int    managed;
+  long   bitrate_min;
+  long   bitrate_av;
+  double bitrate_av_damp;
+  long   bitrate_max;
+  long   bitrate_reservoir;
+  double bitrate_reservoir_bias;
+
+  int impulse_block_p;
+  int noise_normalize_p;
+  int coupling_p;
+
+  double stereo_point_setting;
+  double lowpass_kHz;
+  int    lowpass_altered;
+
+  double ath_floating_dB;
+  double ath_absolute_dB;
+
+  double amplitude_track_dBpersec;
+  double trigger_setting;
+
+  highlevel_byblocktype block[4]; /* padding, impulse, transition, long */
+
+} highlevel_encode_setup;

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/info.c

@@ -0,0 +1,660 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2010             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: maintain the info structure, info <-> header packets
+ last mod: $Id: info.c 17584 2010-11-01 19:26:16Z xiphmont $
+
+ ********************************************************************/
+
+/* general handling of the header and the vorbis_info structure (and
+   substructures) */
+
+#include <stdlib.h>
+#include <string.h>
+#include <ctype.h>
+#include "../../ogg.h"
+#include "../../codec.h"
+#include "codec_internal.h"
+#include "codebook.h"
+#include "registry.h"
+#include "window.h"
+#include "psy.h"
+#include "misc.h"
+#include "os.h"
+
+#define GENERAL_VENDOR_STRING "Xiph.Org libVorbis 1.3.2"
+#define ENCODE_VENDOR_STRING "Xiph.Org libVorbis I 20101101 (Schaufenugget)"
+
+/* helpers */
+
+static void _v_writestring(oggpack_buffer *o,const char *s, int bytes){
+
+  while(bytes--){
+    oggpack_write(o,*s++,8);
+  }
+}
+
+static void _v_readstring(oggpack_buffer *o,char *buf,int bytes){
+  while(bytes--){
+    *buf++=oggpack_read(o,8);
+  }
+}
+
+void vorbis_comment_init(vorbis_comment *vc){
+  memset(vc,0,sizeof(*vc));
+}
+
+void vorbis_comment_add(vorbis_comment *vc,const char *comment){
+  vc->user_comments=(char**)_ogg_realloc(vc->user_comments,
+                            (vc->comments+2)*sizeof(*vc->user_comments));
+  vc->comment_lengths=(int*)_ogg_realloc(vc->comment_lengths,
+                                  (vc->comments+2)*sizeof(*vc->comment_lengths));
+  vc->comment_lengths[vc->comments]=strlen(comment);
+  vc->user_comments[vc->comments]=(char*)_ogg_malloc(vc->comment_lengths[vc->comments]+1);
+  strcpy(vc->user_comments[vc->comments], comment);
+  vc->comments++;
+  vc->user_comments[vc->comments]=NULL;
+}
+
+void vorbis_comment_add_tag(vorbis_comment *vc, const char *tag, const char *contents){
+  char *comment=(char*)alloca(strlen(tag)+strlen(contents)+2); /* +2 for = and \0 */
+  strcpy(comment, tag);
+  strcat(comment, "=");
+  strcat(comment, contents);
+  vorbis_comment_add(vc, comment);
+}
+
+/* This is more or less the same as strncasecmp - but that doesn't exist
+ * everywhere, and this is a fairly trivial function, so we include it */
+static int tagcompare(const char *s1, const char *s2, int n){
+  int c=0;
+  while(c < n){
+    if(toupper(s1[c]) != toupper(s2[c]))
+      return !0;
+    c++;
+  }
+  return 0;
+}
+
+char *vorbis_comment_query(vorbis_comment *vc, const char *tag, int count){
+  long i;
+  int found = 0;
+  int taglen = strlen(tag)+1; /* +1 for the = we append */
+  char *fulltag = (char*)alloca(taglen+ 1);
+
+  strcpy(fulltag, tag);
+  strcat(fulltag, "=");
+
+  for(i=0;i<vc->comments;i++){
+    if(!tagcompare(vc->user_comments[i], fulltag, taglen)){
+      if(count == found)
+        /* We return a pointer to the data, not a copy */
+              return vc->user_comments[i] + taglen;
+      else
+        found++;
+    }
+  }
+  return NULL; /* didn't find anything */
+}
+
+int vorbis_comment_query_count(vorbis_comment *vc, const char *tag){
+  int i,count=0;
+  int taglen = strlen(tag)+1; /* +1 for the = we append */
+  char *fulltag = (char*)alloca(taglen+1);
+  strcpy(fulltag,tag);
+  strcat(fulltag, "=");
+
+  for(i=0;i<vc->comments;i++){
+    if(!tagcompare(vc->user_comments[i], fulltag, taglen))
+      count++;
+  }
+
+  return count;
+}
+
+void vorbis_comment_clear(vorbis_comment *vc){
+  if(vc){
+    long i;
+    if(vc->user_comments){
+      for(i=0;i<vc->comments;i++)
+        if(vc->user_comments[i])_ogg_free(vc->user_comments[i]);
+      _ogg_free(vc->user_comments);
+    }
+    if(vc->comment_lengths)_ogg_free(vc->comment_lengths);
+    if(vc->vendor)_ogg_free(vc->vendor);
+    memset(vc,0,sizeof(*vc));
+  }
+}
+
+/* blocksize 0 is guaranteed to be short, 1 is guaranteed to be long.
+   They may be equal, but short will never ge greater than long */
+int vorbis_info_blocksize(vorbis_info *vi,int zo){
+  codec_setup_info *ci = (codec_setup_info*)vi->codec_setup;
+  return ci ? ci->blocksizes[zo] : -1;
+}
+
+/* used by synthesis, which has a full, alloced vi */
+void vorbis_info_init(vorbis_info *vi){
+  memset(vi,0,sizeof(*vi));
+  vi->codec_setup=_ogg_calloc(1,sizeof(codec_setup_info));
+}
+
+void vorbis_info_clear(vorbis_info *vi){
+  codec_setup_info     *ci=(codec_setup_info*)vi->codec_setup;
+  int i;
+
+  if(ci){
+
+    for(i=0;i<ci->modes;i++)
+      if(ci->mode_param[i])_ogg_free(ci->mode_param[i]);
+
+    for(i=0;i<ci->maps;i++) /* unpack does the range checking */
+      if(ci->map_param[i]) /* this may be cleaning up an aborted
+                              unpack, in which case the below type
+                              cannot be trusted */
+        _mapping_P[ci->map_type[i]]->free_info(ci->map_param[i]);
+
+    for(i=0;i<ci->floors;i++) /* unpack does the range checking */
+      if(ci->floor_param[i]) /* this may be cleaning up an aborted
+                                unpack, in which case the below type
+                                cannot be trusted */
+        _floor_P[ci->floor_type[i]]->free_info(ci->floor_param[i]);
+
+    for(i=0;i<ci->residues;i++) /* unpack does the range checking */
+      if(ci->residue_param[i]) /* this may be cleaning up an aborted
+                                  unpack, in which case the below type
+                                  cannot be trusted */
+        _residue_P[ci->residue_type[i]]->free_info(ci->residue_param[i]);
+
+    for(i=0;i<ci->books;i++){
+      if(ci->book_param[i]){
+        /* knows if the book was not alloced */
+        vorbis_staticbook_destroy(ci->book_param[i]);
+      }
+      if(ci->fullbooks)
+        vorbis_book_clear(ci->fullbooks+i);
+    }
+    if(ci->fullbooks)
+        _ogg_free(ci->fullbooks);
+
+    for(i=0;i<ci->psys;i++)
+      _vi_psy_free(ci->psy_param[i]);
+
+    _ogg_free(ci);
+  }
+
+  memset(vi,0,sizeof(*vi));
+}
+
+/* Header packing/unpacking ********************************************/
+
+static int _vorbis_unpack_info(vorbis_info *vi,oggpack_buffer *opb){
+  codec_setup_info     *ci=(codec_setup_info*)vi->codec_setup;
+  if(!ci)return(OV_EFAULT);
+
+  vi->version=oggpack_read(opb,32);
+  if(vi->version!=0)return(OV_EVERSION);
+
+  vi->channels=oggpack_read(opb,8);
+  vi->rate=oggpack_read(opb,32);
+
+  vi->bitrate_upper=oggpack_read(opb,32);
+  vi->bitrate_nominal=oggpack_read(opb,32);
+  vi->bitrate_lower=oggpack_read(opb,32);
+
+  ci->blocksizes[0]=1<<oggpack_read(opb,4);
+  ci->blocksizes[1]=1<<oggpack_read(opb,4);
+
+  if(vi->rate<1)goto err_out;
+  if(vi->channels<1)goto err_out;
+  if(ci->blocksizes[0]<64)goto err_out;
+  if(ci->blocksizes[1]<ci->blocksizes[0])goto err_out;
+  if(ci->blocksizes[1]>8192)goto err_out;
+
+  if(oggpack_read(opb,1)!=1)goto err_out; /* EOP check */
+
+  return(0);
+ err_out:
+  vorbis_info_clear(vi);
+  return(OV_EBADHEADER);
+}
+
+static int _vorbis_unpack_comment(vorbis_comment *vc,oggpack_buffer *opb){
+  int i;
+  int vendorlen=oggpack_read(opb,32);
+  if(vendorlen<0)goto err_out;
+  if(vendorlen>opb->storage-8)goto err_out;
+  vc->vendor=(char*)_ogg_calloc(vendorlen+1,1);
+  _v_readstring(opb,vc->vendor,vendorlen);
+  i=oggpack_read(opb,32);
+  if(i<0)goto err_out;
+  if(i>((opb->storage-oggpack_bytes(opb))>>2))goto err_out;
+  vc->comments=i;
+  vc->user_comments=(char**)_ogg_calloc(vc->comments+1,sizeof(*vc->user_comments));
+  vc->comment_lengths=(int*)_ogg_calloc(vc->comments+1, sizeof(*vc->comment_lengths));
+
+  for(i=0;i<vc->comments;i++){
+    int len=oggpack_read(opb,32);
+    if(len<0)goto err_out;
+    if(len>opb->storage-oggpack_bytes(opb))goto err_out;
+    vc->comment_lengths[i]=len;
+    vc->user_comments[i]=(char*)_ogg_calloc(len+1,1);
+    _v_readstring(opb,vc->user_comments[i],len);
+  }
+  if(oggpack_read(opb,1)!=1)goto err_out; /* EOP check */
+
+  return(0);
+ err_out:
+  vorbis_comment_clear(vc);
+  return(OV_EBADHEADER);
+}
+
+/* all of the real encoding details are here.  The modes, books,
+   everything */
+static int _vorbis_unpack_books(vorbis_info *vi,oggpack_buffer *opb){
+  codec_setup_info     *ci=(codec_setup_info*)vi->codec_setup;
+  int i;
+  if(!ci)return(OV_EFAULT);
+
+  /* codebooks */
+  ci->books=oggpack_read(opb,8)+1;
+  if(ci->books<=0)goto err_out;
+  for(i=0;i<ci->books;i++){
+    ci->book_param[i]=vorbis_staticbook_unpack(opb);
+    if(!ci->book_param[i])goto err_out;
+  }
+
+  /* time backend settings; hooks are unused */
+  {
+    int times=oggpack_read(opb,6)+1;
+    if(times<=0)goto err_out;
+    for(i=0;i<times;i++){
+      int test=oggpack_read(opb,16);
+      if(test<0 || test>=VI_TIMEB)goto err_out;
+    }
+  }
+
+  /* floor backend settings */
+  ci->floors=oggpack_read(opb,6)+1;
+  if(ci->floors<=0)goto err_out;
+  for(i=0;i<ci->floors;i++){
+    ci->floor_type[i]=oggpack_read(opb,16);
+    if(ci->floor_type[i]<0 || ci->floor_type[i]>=VI_FLOORB)goto err_out;
+    ci->floor_param[i]=_floor_P[ci->floor_type[i]]->unpack(vi,opb);
+    if(!ci->floor_param[i])goto err_out;
+  }
+
+  /* residue backend settings */
+  ci->residues=oggpack_read(opb,6)+1;
+  if(ci->residues<=0)goto err_out;
+  for(i=0;i<ci->residues;i++){
+    ci->residue_type[i]=oggpack_read(opb,16);
+    if(ci->residue_type[i]<0 || ci->residue_type[i]>=VI_RESB)goto err_out;
+    ci->residue_param[i]=_residue_P[ci->residue_type[i]]->unpack(vi,opb);
+    if(!ci->residue_param[i])goto err_out;
+  }
+
+  /* map backend settings */
+  ci->maps=oggpack_read(opb,6)+1;
+  if(ci->maps<=0)goto err_out;
+  for(i=0;i<ci->maps;i++){
+    ci->map_type[i]=oggpack_read(opb,16);
+    if(ci->map_type[i]<0 || ci->map_type[i]>=VI_MAPB)goto err_out;
+    ci->map_param[i]=_mapping_P[ci->map_type[i]]->unpack(vi,opb);
+    if(!ci->map_param[i])goto err_out;
+  }
+
+  /* mode settings */
+  ci->modes=oggpack_read(opb,6)+1;
+  if(ci->modes<=0)goto err_out;
+  for(i=0;i<ci->modes;i++){
+    ci->mode_param[i]=(vorbis_info_mode*)_ogg_calloc(1,sizeof(*ci->mode_param[i]));
+    ci->mode_param[i]->blockflag=oggpack_read(opb,1);
+    ci->mode_param[i]->windowtype=oggpack_read(opb,16);
+    ci->mode_param[i]->transformtype=oggpack_read(opb,16);
+    ci->mode_param[i]->mapping=oggpack_read(opb,8);
+
+    if(ci->mode_param[i]->windowtype>=VI_WINDOWB)goto err_out;
+    if(ci->mode_param[i]->transformtype>=VI_WINDOWB)goto err_out;
+    if(ci->mode_param[i]->mapping>=ci->maps)goto err_out;
+    if(ci->mode_param[i]->mapping<0)goto err_out;
+  }
+
+  if(oggpack_read(opb,1)!=1)goto err_out; /* top level EOP check */
+
+  return(0);
+ err_out:
+  vorbis_info_clear(vi);
+  return(OV_EBADHEADER);
+}
+
+/* Is this packet a vorbis ID header? */
+int vorbis_synthesis_idheader(ogg_packet *op){
+  oggpack_buffer opb;
+  char buffer[6];
+
+  if(op){
+    oggpack_readinit(&opb,op->packet,op->bytes);
+
+    if(!op->b_o_s)
+      return(0); /* Not the initial packet */
+
+    if(oggpack_read(&opb,8) != 1)
+      return 0; /* not an ID header */
+
+    memset(buffer,0,6);
+    _v_readstring(&opb,buffer,6);
+    if(memcmp(buffer,"vorbis",6))
+      return 0; /* not vorbis */
+
+    return 1;
+  }
+
+  return 0;
+}
+
+/* The Vorbis header is in three packets; the initial small packet in
+   the first page that identifies basic parameters, a second packet
+   with bitstream comments and a third packet that holds the
+   codebook. */
+
+int vorbis_synthesis_headerin(vorbis_info *vi,vorbis_comment *vc,ogg_packet *op){
+  oggpack_buffer opb;
+
+  if(op){
+    oggpack_readinit(&opb,op->packet,op->bytes);
+
+    /* Which of the three types of header is this? */
+    /* Also verify header-ness, vorbis */
+    {
+      char buffer[6];
+      int packtype=oggpack_read(&opb,8);
+      memset(buffer,0,6);
+      _v_readstring(&opb,buffer,6);
+      if(memcmp(buffer,"vorbis",6)){
+        /* not a vorbis header */
+        return(OV_ENOTVORBIS);
+      }
+      switch(packtype){
+      case 0x01: /* least significant *bit* is read first */
+        if(!op->b_o_s){
+          /* Not the initial packet */
+          return(OV_EBADHEADER);
+        }
+        if(vi->rate!=0){
+          /* previously initialized info header */
+          return(OV_EBADHEADER);
+        }
+
+        return(_vorbis_unpack_info(vi,&opb));
+
+      case 0x03: /* least significant *bit* is read first */
+        if(vi->rate==0){
+          /* um... we didn't get the initial header */
+          return(OV_EBADHEADER);
+        }
+
+        return(_vorbis_unpack_comment(vc,&opb));
+
+      case 0x05: /* least significant *bit* is read first */
+        if(vi->rate==0 || vc->vendor==NULL){
+          /* um... we didn;t get the initial header or comments yet */
+          return(OV_EBADHEADER);
+        }
+
+        return(_vorbis_unpack_books(vi,&opb));
+
+      default:
+        /* Not a valid vorbis header type */
+        return(OV_EBADHEADER);
+        break;
+      }
+    }
+  }
+  return(OV_EBADHEADER);
+}
+
+/* pack side **********************************************************/
+
+static int _vorbis_pack_info(oggpack_buffer *opb,vorbis_info *vi){
+  codec_setup_info     *ci=(codec_setup_info*)vi->codec_setup;
+  if(!ci)return(OV_EFAULT);
+
+  /* preamble */
+  oggpack_write(opb,0x01,8);
+  _v_writestring(opb,"vorbis", 6);
+
+  /* basic information about the stream */
+  oggpack_write(opb,0x00,32);
+  oggpack_write(opb,vi->channels,8);
+  oggpack_write(opb,vi->rate,32);
+
+  oggpack_write(opb,vi->bitrate_upper,32);
+  oggpack_write(opb,vi->bitrate_nominal,32);
+  oggpack_write(opb,vi->bitrate_lower,32);
+
+  oggpack_write(opb,ilog2(ci->blocksizes[0]),4);
+  oggpack_write(opb,ilog2(ci->blocksizes[1]),4);
+  oggpack_write(opb,1,1);
+
+  return(0);
+}
+
+static int _vorbis_pack_comment(oggpack_buffer *opb,vorbis_comment *vc){
+  int bytes = strlen(ENCODE_VENDOR_STRING);
+
+  /* preamble */
+  oggpack_write(opb,0x03,8);
+  _v_writestring(opb,"vorbis", 6);
+
+  /* vendor */
+  oggpack_write(opb,bytes,32);
+  _v_writestring(opb,ENCODE_VENDOR_STRING, bytes);
+
+  /* comments */
+
+  oggpack_write(opb,vc->comments,32);
+  if(vc->comments){
+    int i;
+    for(i=0;i<vc->comments;i++){
+      if(vc->user_comments[i]){
+        oggpack_write(opb,vc->comment_lengths[i],32);
+        _v_writestring(opb,vc->user_comments[i], vc->comment_lengths[i]);
+      }else{
+        oggpack_write(opb,0,32);
+      }
+    }
+  }
+  oggpack_write(opb,1,1);
+
+  return(0);
+}
+
+static int _vorbis_pack_books(oggpack_buffer *opb,vorbis_info *vi){
+  codec_setup_info     *ci=(codec_setup_info*)vi->codec_setup;
+  int i;
+  if(!ci)return(OV_EFAULT);
+
+  oggpack_write(opb,0x05,8);
+  _v_writestring(opb,"vorbis", 6);
+
+  /* books */
+  oggpack_write(opb,ci->books-1,8);
+  for(i=0;i<ci->books;i++)
+    if(vorbis_staticbook_pack(ci->book_param[i],opb))goto err_out;
+
+  /* times; hook placeholders */
+  oggpack_write(opb,0,6);
+  oggpack_write(opb,0,16);
+
+  /* floors */
+  oggpack_write(opb,ci->floors-1,6);
+  for(i=0;i<ci->floors;i++){
+    oggpack_write(opb,ci->floor_type[i],16);
+    if(_floor_P[ci->floor_type[i]]->pack)
+      _floor_P[ci->floor_type[i]]->pack(ci->floor_param[i],opb);
+    else
+      goto err_out;
+  }
+
+  /* residues */
+  oggpack_write(opb,ci->residues-1,6);
+  for(i=0;i<ci->residues;i++){
+    oggpack_write(opb,ci->residue_type[i],16);
+    _residue_P[ci->residue_type[i]]->pack(ci->residue_param[i],opb);
+  }
+
+  /* maps */
+  oggpack_write(opb,ci->maps-1,6);
+  for(i=0;i<ci->maps;i++){
+    oggpack_write(opb,ci->map_type[i],16);
+    _mapping_P[ci->map_type[i]]->pack(vi,ci->map_param[i],opb);
+  }
+
+  /* modes */
+  oggpack_write(opb,ci->modes-1,6);
+  for(i=0;i<ci->modes;i++){
+    oggpack_write(opb,ci->mode_param[i]->blockflag,1);
+    oggpack_write(opb,ci->mode_param[i]->windowtype,16);
+    oggpack_write(opb,ci->mode_param[i]->transformtype,16);
+    oggpack_write(opb,ci->mode_param[i]->mapping,8);
+  }
+  oggpack_write(opb,1,1);
+
+  return(0);
+err_out:
+  return(-1);
+}
+
+int vorbis_commentheader_out(vorbis_comment *vc,
+                                          ogg_packet *op){
+
+  oggpack_buffer opb;
+
+  oggpack_writeinit(&opb);
+  if(_vorbis_pack_comment(&opb,vc)) return OV_EIMPL;
+
+  op->packet = (unsigned char*) _ogg_malloc(oggpack_bytes(&opb));
+  memcpy(op->packet, opb.buffer, oggpack_bytes(&opb));
+
+  op->bytes=oggpack_bytes(&opb);
+  op->b_o_s=0;
+  op->e_o_s=0;
+  op->granulepos=0;
+  op->packetno=1;
+
+  return 0;
+}
+
+int vorbis_analysis_headerout(vorbis_dsp_state *v,
+                              vorbis_comment *vc,
+                              ogg_packet *op,
+                              ogg_packet *op_comm,
+                              ogg_packet *op_code){
+  int ret=OV_EIMPL;
+  vorbis_info *vi=v->vi;
+  oggpack_buffer opb;
+  private_state *b=(private_state*)v->backend_state;
+
+  if(!b){
+    ret=OV_EFAULT;
+    goto err_out;
+  }
+
+  /* first header packet **********************************************/
+
+  oggpack_writeinit(&opb);
+  if(_vorbis_pack_info(&opb,vi))goto err_out;
+
+  /* build the packet */
+  if(b->header)_ogg_free(b->header);
+  b->header=(unsigned char*) _ogg_malloc(oggpack_bytes(&opb));
+  memcpy(b->header,opb.buffer,oggpack_bytes(&opb));
+  op->packet=b->header;
+  op->bytes=oggpack_bytes(&opb);
+  op->b_o_s=1;
+  op->e_o_s=0;
+  op->granulepos=0;
+  op->packetno=0;
+
+  /* second header packet (comments) **********************************/
+
+  oggpack_reset(&opb);
+  if(_vorbis_pack_comment(&opb,vc))goto err_out;
+
+  if(b->header1)_ogg_free(b->header1);
+  b->header1=(unsigned char*) _ogg_malloc(oggpack_bytes(&opb));
+  memcpy(b->header1,opb.buffer,oggpack_bytes(&opb));
+  op_comm->packet=b->header1;
+  op_comm->bytes=oggpack_bytes(&opb);
+  op_comm->b_o_s=0;
+  op_comm->e_o_s=0;
+  op_comm->granulepos=0;
+  op_comm->packetno=1;
+
+  /* third header packet (modes/codebooks) ****************************/
+
+  oggpack_reset(&opb);
+  if(_vorbis_pack_books(&opb,vi))goto err_out;
+
+  if(b->header2)_ogg_free(b->header2);
+  b->header2=(unsigned char*) _ogg_malloc(oggpack_bytes(&opb));
+  memcpy(b->header2,opb.buffer,oggpack_bytes(&opb));
+  op_code->packet=b->header2;
+  op_code->bytes=oggpack_bytes(&opb);
+  op_code->b_o_s=0;
+  op_code->e_o_s=0;
+  op_code->granulepos=0;
+  op_code->packetno=2;
+
+  oggpack_writeclear(&opb);
+  return(0);
+ err_out:
+  memset(op,0,sizeof(*op));
+  memset(op_comm,0,sizeof(*op_comm));
+  memset(op_code,0,sizeof(*op_code));
+
+  if(b){
+    oggpack_writeclear(&opb);
+    if(b->header)_ogg_free(b->header);
+    if(b->header1)_ogg_free(b->header1);
+    if(b->header2)_ogg_free(b->header2);
+    b->header=NULL;
+    b->header1=NULL;
+    b->header2=NULL;
+  }
+  return(ret);
+}
+
+double vorbis_granule_time(vorbis_dsp_state *v,ogg_int64_t granulepos){
+  if(granulepos == -1) return -1;
+
+  /* We're not guaranteed a 64 bit unsigned type everywhere, so we
+     have to put the unsigned granpo in a signed type. */
+  if(granulepos>=0){
+    return((double)granulepos/v->vi->rate);
+  }else{
+    ogg_int64_t granuleoff=0xffffffff;
+    granuleoff<<=31;
+
+#ifdef __GNUC__
+    granuleoff |= 0x7ffffffffLL;
+#else
+    granuleoff |= 0x7ffffffff;
+#endif
+    return(((double)granulepos+2+granuleoff+granuleoff)/v->vi->rate);
+  }
+}
+
+const char *vorbis_version_string(void){
+  return GENERAL_VENDOR_STRING;
+}

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/lookup.c

@@ -0,0 +1,94 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+  function: lookup based functions
+  last mod: $Id: lookup.c 16227 2009-07-08 06:58:46Z xiphmont $
+
+ ********************************************************************/
+
+#include <math.h>
+#include "lookup.h"
+#include "lookup_data.h"
+#include "os.h"
+#include "misc.h"
+
+#ifdef FLOAT_LOOKUP
+
+/* interpolated lookup based cos function, domain 0 to PI only */
+float vorbis_coslook(float a){
+  double d=a*(.31830989*(float)COS_LOOKUP_SZ);
+  int i=vorbis_ftoi(d-.5);
+
+  return COS_LOOKUP[i]+ (d-i)*(COS_LOOKUP[i+1]-COS_LOOKUP[i]);
+}
+
+/* interpolated 1./sqrt(p) where .5 <= p < 1. */
+float vorbis_invsqlook(float a){
+  double d=a*(2.f*(float)INVSQ_LOOKUP_SZ)-(float)INVSQ_LOOKUP_SZ;
+  int i=vorbis_ftoi(d-.5f);
+  return INVSQ_LOOKUP[i]+ (d-i)*(INVSQ_LOOKUP[i+1]-INVSQ_LOOKUP[i]);
+}
+
+/* interpolated 1./sqrt(p) where .5 <= p < 1. */
+float vorbis_invsq2explook(int a){
+  return INVSQ2EXP_LOOKUP[a-INVSQ2EXP_LOOKUP_MIN];
+}
+
+#include <stdio.h>
+/* interpolated lookup based fromdB function, domain -140dB to 0dB only */
+float vorbis_fromdBlook(float a){
+  int i=vorbis_ftoi(a*((float)(-(1<<FROMdB2_SHIFT)))-.5f);
+  return (i<0)?1.f:
+    ((i>=(FROMdB_LOOKUP_SZ<<FROMdB_SHIFT))?0.f:
+     FROMdB_LOOKUP[i>>FROMdB_SHIFT]*FROMdB2_LOOKUP[i&FROMdB2_MASK]);
+}
+
+#endif
+
+#ifdef INT_LOOKUP
+/* interpolated 1./sqrt(p) where .5 <= a < 1. (.100000... to .111111...) in
+   16.16 format
+
+   returns in m.8 format */
+long vorbis_invsqlook_i(long a,long e){
+  long i=(a&0x7fff)>>(INVSQ_LOOKUP_I_SHIFT-1);
+  long d=(a&INVSQ_LOOKUP_I_MASK)<<(16-INVSQ_LOOKUP_I_SHIFT); /*  0.16 */
+  long val=INVSQ_LOOKUP_I[i]-                                /*  1.16 */
+    (((INVSQ_LOOKUP_I[i]-INVSQ_LOOKUP_I[i+1])*               /*  0.16 */
+      d)>>16);                                               /* result 1.16 */
+
+  e+=32;
+  if(e&1)val=(val*5792)>>13; /* multiply val by 1/sqrt(2) */
+  e=(e>>1)-8;
+
+  return(val>>e);
+}
+
+/* interpolated lookup based fromdB function, domain -140dB to 0dB only */
+/* a is in n.12 format */
+float vorbis_fromdBlook_i(long a){
+  int i=(-a)>>(12-FROMdB2_SHIFT);
+  return (i<0)?1.f:
+    ((i>=(FROMdB_LOOKUP_SZ<<FROMdB_SHIFT))?0.f:
+     FROMdB_LOOKUP[i>>FROMdB_SHIFT]*FROMdB2_LOOKUP[i&FROMdB2_MASK]);
+}
+
+/* interpolated lookup based cos function, domain 0 to PI only */
+/* a is in 0.16 format, where 0==0, 2^^16-1==PI, return 0.14 */
+long vorbis_coslook_i(long a){
+  int i=a>>COS_LOOKUP_I_SHIFT;
+  int d=a&COS_LOOKUP_I_MASK;
+  return COS_LOOKUP_I[i]- ((d*(COS_LOOKUP_I[i]-COS_LOOKUP_I[i+1]))>>
+                           COS_LOOKUP_I_SHIFT);
+}
+
+#endif

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/lookup.h

@@ -0,0 +1,32 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+  function: lookup based functions
+  last mod: $Id: lookup.h 16227 2009-07-08 06:58:46Z xiphmont $
+
+ ********************************************************************/
+
+#ifndef _V_LOOKUP_H_
+
+#ifdef FLOAT_LOOKUP
+extern float vorbis_coslook(float a);
+extern float vorbis_invsqlook(float a);
+extern float vorbis_invsq2explook(int a);
+extern float vorbis_fromdBlook(float a);
+#endif
+#ifdef INT_LOOKUP
+extern long vorbis_invsqlook_i(long a,long e);
+extern long vorbis_coslook_i(long a);
+extern float vorbis_fromdBlook_i(long a);
+#endif
+
+#endif

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/lookup_data.h

@@ -0,0 +1,192 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+  function: lookup data; generated by lookups.pl; edit there
+  last mod: $Id: lookup_data.h 16037 2009-05-26 21:10:58Z xiphmont $
+
+ ********************************************************************/
+
+#ifndef _V_LOOKUP_DATA_H_
+
+#ifdef FLOAT_LOOKUP
+#define COS_LOOKUP_SZ 128
+static const float COS_LOOKUP[COS_LOOKUP_SZ+1]={
+        +1.0000000000000f,+0.9996988186962f,+0.9987954562052f,+0.9972904566787f,
+        +0.9951847266722f,+0.9924795345987f,+0.9891765099648f,+0.9852776423889f,
+        +0.9807852804032f,+0.9757021300385f,+0.9700312531945f,+0.9637760657954f,
+        +0.9569403357322f,+0.9495281805930f,+0.9415440651830f,+0.9329927988347f,
+        +0.9238795325113f,+0.9142097557035f,+0.9039892931234f,+0.8932243011955f,
+        +0.8819212643484f,+0.8700869911087f,+0.8577286100003f,+0.8448535652497f,
+        +0.8314696123025f,+0.8175848131516f,+0.8032075314806f,+0.7883464276266f,
+        +0.7730104533627f,+0.7572088465065f,+0.7409511253550f,+0.7242470829515f,
+        +0.7071067811865f,+0.6895405447371f,+0.6715589548470f,+0.6531728429538f,
+        +0.6343932841636f,+0.6152315905806f,+0.5956993044924f,+0.5758081914178f,
+        +0.5555702330196f,+0.5349976198871f,+0.5141027441932f,+0.4928981922298f,
+        +0.4713967368260f,+0.4496113296546f,+0.4275550934303f,+0.4052413140050f,
+        +0.3826834323651f,+0.3598950365350f,+0.3368898533922f,+0.3136817403989f,
+        +0.2902846772545f,+0.2667127574749f,+0.2429801799033f,+0.2191012401569f,
+        +0.1950903220161f,+0.1709618887603f,+0.1467304744554f,+0.1224106751992f,
+        +0.0980171403296f,+0.0735645635997f,+0.0490676743274f,+0.0245412285229f,
+        +0.0000000000000f,-0.0245412285229f,-0.0490676743274f,-0.0735645635997f,
+        -0.0980171403296f,-0.1224106751992f,-0.1467304744554f,-0.1709618887603f,
+        -0.1950903220161f,-0.2191012401569f,-0.2429801799033f,-0.2667127574749f,
+        -0.2902846772545f,-0.3136817403989f,-0.3368898533922f,-0.3598950365350f,
+        -0.3826834323651f,-0.4052413140050f,-0.4275550934303f,-0.4496113296546f,
+        -0.4713967368260f,-0.4928981922298f,-0.5141027441932f,-0.5349976198871f,
+        -0.5555702330196f,-0.5758081914178f,-0.5956993044924f,-0.6152315905806f,
+        -0.6343932841636f,-0.6531728429538f,-0.6715589548470f,-0.6895405447371f,
+        -0.7071067811865f,-0.7242470829515f,-0.7409511253550f,-0.7572088465065f,
+        -0.7730104533627f,-0.7883464276266f,-0.8032075314806f,-0.8175848131516f,
+        -0.8314696123025f,-0.8448535652497f,-0.8577286100003f,-0.8700869911087f,
+        -0.8819212643484f,-0.8932243011955f,-0.9039892931234f,-0.9142097557035f,
+        -0.9238795325113f,-0.9329927988347f,-0.9415440651830f,-0.9495281805930f,
+        -0.9569403357322f,-0.9637760657954f,-0.9700312531945f,-0.9757021300385f,
+        -0.9807852804032f,-0.9852776423889f,-0.9891765099648f,-0.9924795345987f,
+        -0.9951847266722f,-0.9972904566787f,-0.9987954562052f,-0.9996988186962f,
+        -1.0000000000000f,
+};
+
+#define INVSQ_LOOKUP_SZ 32
+static const float INVSQ_LOOKUP[INVSQ_LOOKUP_SZ+1]={
+        1.414213562373f,1.392621247646f,1.371988681140f,1.352246807566f,
+        1.333333333333f,1.315191898443f,1.297771369046f,1.281025230441f,
+        1.264911064067f,1.249390095109f,1.234426799697f,1.219988562661f,
+        1.206045378311f,1.192569588000f,1.179535649239f,1.166919931983f,
+        1.154700538379f,1.142857142857f,1.131370849898f,1.120224067222f,
+        1.109400392450f,1.098884511590f,1.088662107904f,1.078719779941f,
+        1.069044967650f,1.059625885652f,1.050451462878f,1.041511287847f,
+        1.032795558989f,1.024295039463f,1.016001016002f,1.007905261358f,
+        1.000000000000f,
+};
+
+#define INVSQ2EXP_LOOKUP_MIN (-32)
+#define INVSQ2EXP_LOOKUP_MAX 32
+static const float INVSQ2EXP_LOOKUP[INVSQ2EXP_LOOKUP_MAX-\
+                              INVSQ2EXP_LOOKUP_MIN+1]={
+                 65536.f,    46340.95001f,         32768.f,    23170.47501f,
+                 16384.f,     11585.2375f,          8192.f,    5792.618751f,
+                  4096.f,    2896.309376f,          2048.f,    1448.154688f,
+                  1024.f,    724.0773439f,           512.f,     362.038672f,
+                   256.f,     181.019336f,           128.f,    90.50966799f,
+                    64.f,      45.254834f,            32.f,      22.627417f,
+                    16.f,     11.3137085f,             8.f,    5.656854249f,
+                     4.f,    2.828427125f,             2.f,    1.414213562f,
+                     1.f,   0.7071067812f,            0.5f,   0.3535533906f,
+                   0.25f,   0.1767766953f,          0.125f,  0.08838834765f,
+                 0.0625f,  0.04419417382f,        0.03125f,  0.02209708691f,
+               0.015625f,  0.01104854346f,      0.0078125f, 0.005524271728f,
+             0.00390625f, 0.002762135864f,    0.001953125f, 0.001381067932f,
+           0.0009765625f, 0.000690533966f,  0.00048828125f, 0.000345266983f,
+         0.000244140625f,0.0001726334915f,0.0001220703125f,8.631674575e-05f,
+        6.103515625e-05f,4.315837288e-05f,3.051757812e-05f,2.157918644e-05f,
+        1.525878906e-05f,
+};
+
+#endif
+
+#define FROMdB_LOOKUP_SZ 35
+#define FROMdB2_LOOKUP_SZ 32
+#define FROMdB_SHIFT 5
+#define FROMdB2_SHIFT 3
+#define FROMdB2_MASK 31
+
+#ifdef FLOAT_LOOKUP
+static const float FROMdB_LOOKUP[FROMdB_LOOKUP_SZ]={
+                     1.f,   0.6309573445f,   0.3981071706f,   0.2511886432f,
+           0.1584893192f,            0.1f,  0.06309573445f,  0.03981071706f,
+          0.02511886432f,  0.01584893192f,           0.01f, 0.006309573445f,
+         0.003981071706f, 0.002511886432f, 0.001584893192f,          0.001f,
+        0.0006309573445f,0.0003981071706f,0.0002511886432f,0.0001584893192f,
+                 0.0001f,6.309573445e-05f,3.981071706e-05f,2.511886432e-05f,
+        1.584893192e-05f,          1e-05f,6.309573445e-06f,3.981071706e-06f,
+        2.511886432e-06f,1.584893192e-06f,          1e-06f,6.309573445e-07f,
+        3.981071706e-07f,2.511886432e-07f,1.584893192e-07f,
+};
+
+static const float FROMdB2_LOOKUP[FROMdB2_LOOKUP_SZ]={
+           0.9928302478f,   0.9786445908f,   0.9646616199f,   0.9508784391f,
+           0.9372921937f,     0.92390007f,   0.9106992942f,   0.8976871324f,
+           0.8848608897f,   0.8722179097f,   0.8597555737f,   0.8474713009f,
+            0.835362547f,   0.8234268041f,   0.8116616003f,   0.8000644989f,
+           0.7886330981f,   0.7773650302f,   0.7662579617f,    0.755309592f,
+           0.7445176537f,   0.7338799116f,   0.7233941627f,   0.7130582353f,
+           0.7028699885f,   0.6928273125f,   0.6829281272f,   0.6731703824f,
+           0.6635520573f,   0.6540711597f,   0.6447257262f,   0.6355138211f,
+};
+#endif
+
+#ifdef INT_LOOKUP
+
+#define INVSQ_LOOKUP_I_SHIFT 10
+#define INVSQ_LOOKUP_I_MASK 1023
+static const long INVSQ_LOOKUP_I[64+1]={
+           92682l,   91966l,   91267l,   90583l,
+           89915l,   89261l,   88621l,   87995l,
+           87381l,   86781l,   86192l,   85616l,
+           85051l,   84497l,   83953l,   83420l,
+           82897l,   82384l,   81880l,   81385l,
+           80899l,   80422l,   79953l,   79492l,
+           79039l,   78594l,   78156l,   77726l,
+           77302l,   76885l,   76475l,   76072l,
+           75674l,   75283l,   74898l,   74519l,
+           74146l,   73778l,   73415l,   73058l,
+           72706l,   72359l,   72016l,   71679l,
+           71347l,   71019l,   70695l,   70376l,
+           70061l,   69750l,   69444l,   69141l,
+           68842l,   68548l,   68256l,   67969l,
+           67685l,   67405l,   67128l,   66855l,
+           66585l,   66318l,   66054l,   65794l,
+           65536l,
+};
+
+#define COS_LOOKUP_I_SHIFT 9
+#define COS_LOOKUP_I_MASK 511
+#define COS_LOOKUP_I_SZ 128
+static const long COS_LOOKUP_I[COS_LOOKUP_I_SZ+1]={
+           16384l,   16379l,   16364l,   16340l,
+           16305l,   16261l,   16207l,   16143l,
+           16069l,   15986l,   15893l,   15791l,
+           15679l,   15557l,   15426l,   15286l,
+           15137l,   14978l,   14811l,   14635l,
+           14449l,   14256l,   14053l,   13842l,
+           13623l,   13395l,   13160l,   12916l,
+           12665l,   12406l,   12140l,   11866l,
+           11585l,   11297l,   11003l,   10702l,
+           10394l,   10080l,    9760l,    9434l,
+            9102l,    8765l,    8423l,    8076l,
+            7723l,    7366l,    7005l,    6639l,
+            6270l,    5897l,    5520l,    5139l,
+            4756l,    4370l,    3981l,    3590l,
+            3196l,    2801l,    2404l,    2006l,
+            1606l,    1205l,     804l,     402l,
+               0l,    -401l,    -803l,   -1204l,
+           -1605l,   -2005l,   -2403l,   -2800l,
+           -3195l,   -3589l,   -3980l,   -4369l,
+           -4755l,   -5138l,   -5519l,   -5896l,
+           -6269l,   -6638l,   -7004l,   -7365l,
+           -7722l,   -8075l,   -8422l,   -8764l,
+           -9101l,   -9433l,   -9759l,  -10079l,
+          -10393l,  -10701l,  -11002l,  -11296l,
+          -11584l,  -11865l,  -12139l,  -12405l,
+          -12664l,  -12915l,  -13159l,  -13394l,
+          -13622l,  -13841l,  -14052l,  -14255l,
+          -14448l,  -14634l,  -14810l,  -14977l,
+          -15136l,  -15285l,  -15425l,  -15556l,
+          -15678l,  -15790l,  -15892l,  -15985l,
+          -16068l,  -16142l,  -16206l,  -16260l,
+          -16304l,  -16339l,  -16363l,  -16378l,
+          -16383l,
+};
+
+#endif
+
+#endif

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/lpc.c

@@ -0,0 +1,160 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+  function: LPC low level routines
+  last mod: $Id: lpc.c 16227 2009-07-08 06:58:46Z xiphmont $
+
+ ********************************************************************/
+
+/* Some of these routines (autocorrelator, LPC coefficient estimator)
+   are derived from code written by Jutta Degener and Carsten Bormann;
+   thus we include their copyright below.  The entirety of this file
+   is freely redistributable on the condition that both of these
+   copyright notices are preserved without modification.  */
+
+/* Preserved Copyright: *********************************************/
+
+/* Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
+Technische Universita"t Berlin
+
+Any use of this software is permitted provided that this notice is not
+removed and that neither the authors nor the Technische Universita"t
+Berlin are deemed to have made any representations as to the
+suitability of this software for any purpose nor are held responsible
+for any defects of this software. THERE IS ABSOLUTELY NO WARRANTY FOR
+THIS SOFTWARE.
+
+As a matter of courtesy, the authors request to be informed about uses
+this software has found, about bugs in this software, and about any
+improvements that may be of general interest.
+
+Berlin, 28.11.1994
+Jutta Degener
+Carsten Bormann
+
+*********************************************************************/
+
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+#include "os.h"
+#include "smallft.h"
+#include "lpc.h"
+#include "scales.h"
+#include "misc.h"
+
+/* Autocorrelation LPC coeff generation algorithm invented by
+   N. Levinson in 1947, modified by J. Durbin in 1959. */
+
+/* Input : n elements of time doamin data
+   Output: m lpc coefficients, excitation energy */
+
+float vorbis_lpc_from_data(float *data,float *lpci,int n,int m){
+  double *aut=(double*)alloca(sizeof(*aut)*(m+1));
+  double *lpc=(double*)alloca(sizeof(*lpc)*(m));
+  double error;
+  double epsilon;
+  int i,j;
+
+  /* autocorrelation, p+1 lag coefficients */
+  j=m+1;
+  while(j--){
+    double d=0; /* double needed for accumulator depth */
+    for(i=j;i<n;i++)d+=(double)data[i]*data[i-j];
+    aut[j]=d;
+  }
+
+  /* Generate lpc coefficients from autocorr values */
+
+  /* set our noise floor to about -100dB */
+  error=aut[0] * (1. + 1e-10);
+  epsilon=1e-9*aut[0]+1e-10;
+
+  for(i=0;i<m;i++){
+    double r= -aut[i+1];
+
+    if(error<epsilon){
+      memset(lpc+i,0,(m-i)*sizeof(*lpc));
+      goto done;
+    }
+
+    /* Sum up this iteration's reflection coefficient; note that in
+       Vorbis we don't save it.  If anyone wants to recycle this code
+       and needs reflection coefficients, save the results of 'r' from
+       each iteration. */
+
+    for(j=0;j<i;j++)r-=lpc[j]*aut[i-j];
+    r/=error;
+
+    /* Update LPC coefficients and total error */
+
+    lpc[i]=r;
+    for(j=0;j<i/2;j++){
+      double tmp=lpc[j];
+
+      lpc[j]+=r*lpc[i-1-j];
+      lpc[i-1-j]+=r*tmp;
+    }
+    if(i&1)lpc[j]+=lpc[j]*r;
+
+    error*=1.-r*r;
+
+  }
+
+ done:
+
+  /* slightly damp the filter */
+  {
+    double g = .99;
+    double damp = g;
+    for(j=0;j<m;j++){
+      lpc[j]*=damp;
+      damp*=g;
+    }
+  }
+
+  for(j=0;j<m;j++)lpci[j]=(float)lpc[j];
+
+  /* we need the error value to know how big an impulse to hit the
+     filter with later */
+
+  return error;
+}
+
+void vorbis_lpc_predict(float *coeff,float *prime,int m,
+                     float *data,long n){
+
+  /* in: coeff[0...m-1] LPC coefficients
+         prime[0...m-1] initial values (allocated size of n+m-1)
+    out: data[0...n-1] data samples */
+
+  long i,j,o,p;
+  float y;
+  float *work=(float*)alloca(sizeof(*work)*(m+n));
+
+  if(!prime)
+    for(i=0;i<m;i++)
+      work[i]=0.f;
+  else
+    for(i=0;i<m;i++)
+      work[i]=prime[i];
+
+  for(i=0;i<n;i++){
+    y=0;
+    o=i;
+    p=m;
+    for(j=0;j<m;j++)
+      y-=work[o++]*coeff[--p];
+
+    data[i]=work[o]=y;
+  }
+}

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/lpc.h

@@ -0,0 +1,29 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+  function: LPC low level routines
+  last mod: $Id: lpc.h 16037 2009-05-26 21:10:58Z xiphmont $
+
+ ********************************************************************/
+
+#ifndef _V_LPC_H_
+#define _V_LPC_H_
+
+#include "../../codec.h"
+
+/* simple linear scale LPC code */
+extern float vorbis_lpc_from_data(float *data,float *lpc,int n,int m);
+
+extern void vorbis_lpc_predict(float *coeff,float *prime,int m,
+                               float *data,long n);
+
+#endif

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/lsp.c

@@ -0,0 +1,454 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+  function: LSP (also called LSF) conversion routines
+  last mod: $Id: lsp.c 17538 2010-10-15 02:52:29Z tterribe $
+
+  The LSP generation code is taken (with minimal modification and a
+  few bugfixes) from "On the Computation of the LSP Frequencies" by
+  Joseph Rothweiler (see http://www.rothweiler.us for contact info).
+  The paper is available at:
+
+  http://www.myown1.com/joe/lsf
+
+ ********************************************************************/
+
+/* Note that the lpc-lsp conversion finds the roots of polynomial with
+   an iterative root polisher (CACM algorithm 283).  It *is* possible
+   to confuse this algorithm into not converging; that should only
+   happen with absurdly closely spaced roots (very sharp peaks in the
+   LPC f response) which in turn should be impossible in our use of
+   the code.  If this *does* happen anyway, it's a bug in the floor
+   finder; find the cause of the confusion (probably a single bin
+   spike or accidental near-float-limit resolution problems) and
+   correct it. */
+
+#include <math.h>
+#include <string.h>
+#include <stdlib.h>
+#include "lsp.h"
+#include "os.h"
+#include "misc.h"
+#include "lookup.h"
+#include "scales.h"
+
+/* three possible LSP to f curve functions; the exact computation
+   (float), a lookup based float implementation, and an integer
+   implementation.  The float lookup is likely the optimal choice on
+   any machine with an FPU.  The integer implementation is *not* fixed
+   point (due to the need for a large dynamic range and thus a
+   separately tracked exponent) and thus much more complex than the
+   relatively simple float implementations. It's mostly for future
+   work on a fully fixed point implementation for processors like the
+   ARM family. */
+
+/* define either of these (preferably FLOAT_LOOKUP) to have faster
+   but less precise implementation. */
+#undef FLOAT_LOOKUP
+#undef INT_LOOKUP
+
+#ifdef FLOAT_LOOKUP
+#include "lookup.c" /* catch this in the build system; we #include for
+                       compilers (like gcc) that can't inline across
+                       modules */
+
+/* side effect: changes *lsp to cosines of lsp */
+void vorbis_lsp_to_curve(float *curve,int *map,int n,int ln,float *lsp,int m,
+                            float amp,float ampoffset){
+  int i;
+  float wdel=M_PI/ln;
+  vorbis_fpu_control fpu;
+
+  vorbis_fpu_setround(&fpu);
+  for(i=0;i<m;i++)lsp[i]=vorbis_coslook(lsp[i]);
+
+  i=0;
+  while(i<n){
+    int k=map[i];
+    int qexp;
+    float p=.7071067812f;
+    float q=.7071067812f;
+    float w=vorbis_coslook(wdel*k);
+    float *ftmp=lsp;
+    int c=m>>1;
+
+    while(c--){
+      q*=ftmp[0]-w;
+      p*=ftmp[1]-w;
+      ftmp+=2;
+    }
+
+    if(m&1){
+      /* odd order filter; slightly assymetric */
+      /* the last coefficient */
+      q*=ftmp[0]-w;
+      q*=q;
+      p*=p*(1.f-w*w);
+    }else{
+      /* even order filter; still symmetric */
+      q*=q*(1.f+w);
+      p*=p*(1.f-w);
+    }
+
+    q=frexp(p+q,&qexp);
+    q=vorbis_fromdBlook(amp*
+                        vorbis_invsqlook(q)*
+                        vorbis_invsq2explook(qexp+m)-
+                        ampoffset);
+
+    do{
+      curve[i++]*=q;
+    }while(map[i]==k);
+  }
+  vorbis_fpu_restore(fpu);
+}
+
+#else
+
+#ifdef INT_LOOKUP
+#include "lookup.c" /* catch this in the build system; we #include for
+                       compilers (like gcc) that can't inline across
+                       modules */
+
+static const int MLOOP_1[64]={
+   0,10,11,11, 12,12,12,12, 13,13,13,13, 13,13,13,13,
+  14,14,14,14, 14,14,14,14, 14,14,14,14, 14,14,14,14,
+  15,15,15,15, 15,15,15,15, 15,15,15,15, 15,15,15,15,
+  15,15,15,15, 15,15,15,15, 15,15,15,15, 15,15,15,15,
+};
+
+static const int MLOOP_2[64]={
+  0,4,5,5, 6,6,6,6, 7,7,7,7, 7,7,7,7,
+  8,8,8,8, 8,8,8,8, 8,8,8,8, 8,8,8,8,
+  9,9,9,9, 9,9,9,9, 9,9,9,9, 9,9,9,9,
+  9,9,9,9, 9,9,9,9, 9,9,9,9, 9,9,9,9,
+};
+
+static const int MLOOP_3[8]={0,1,2,2,3,3,3,3};
+
+
+/* side effect: changes *lsp to cosines of lsp */
+void vorbis_lsp_to_curve(float *curve,int *map,int n,int ln,float *lsp,int m,
+                            float amp,float ampoffset){
+
+  /* 0 <= m < 256 */
+
+  /* set up for using all int later */
+  int i;
+  int ampoffseti=rint(ampoffset*4096.f);
+  int ampi=rint(amp*16.f);
+  long *ilsp=(long*)alloca(m*sizeof(*ilsp));
+  for(i=0;i<m;i++)ilsp[i]=vorbis_coslook_i(lsp[i]/M_PI*65536.f+.5f);
+
+  i=0;
+  while(i<n){
+    int j,k=map[i];
+    unsigned long pi=46341; /* 2**-.5 in 0.16 */
+    unsigned long qi=46341;
+    int qexp=0,shift;
+    long wi=vorbis_coslook_i(k*65536/ln);
+
+    qi*=labs(ilsp[0]-wi);
+    pi*=labs(ilsp[1]-wi);
+
+    for(j=3;j<m;j+=2){
+      if(!(shift=MLOOP_1[(pi|qi)>>25]))
+        if(!(shift=MLOOP_2[(pi|qi)>>19]))
+          shift=MLOOP_3[(pi|qi)>>16];
+      qi=(qi>>shift)*labs(ilsp[j-1]-wi);
+      pi=(pi>>shift)*labs(ilsp[j]-wi);
+      qexp+=shift;
+    }
+    if(!(shift=MLOOP_1[(pi|qi)>>25]))
+      if(!(shift=MLOOP_2[(pi|qi)>>19]))
+        shift=MLOOP_3[(pi|qi)>>16];
+
+    /* pi,qi normalized collectively, both tracked using qexp */
+
+    if(m&1){
+      /* odd order filter; slightly assymetric */
+      /* the last coefficient */
+      qi=(qi>>shift)*labs(ilsp[j-1]-wi);
+      pi=(pi>>shift)<<14;
+      qexp+=shift;
+
+      if(!(shift=MLOOP_1[(pi|qi)>>25]))
+        if(!(shift=MLOOP_2[(pi|qi)>>19]))
+          shift=MLOOP_3[(pi|qi)>>16];
+
+      pi>>=shift;
+      qi>>=shift;
+      qexp+=shift-14*((m+1)>>1);
+
+      pi=((pi*pi)>>16);
+      qi=((qi*qi)>>16);
+      qexp=qexp*2+m;
+
+      pi*=(1<<14)-((wi*wi)>>14);
+      qi+=pi>>14;
+
+    }else{
+      /* even order filter; still symmetric */
+
+      /* p*=p(1-w), q*=q(1+w), let normalization drift because it isn't
+         worth tracking step by step */
+
+      pi>>=shift;
+      qi>>=shift;
+      qexp+=shift-7*m;
+
+      pi=((pi*pi)>>16);
+      qi=((qi*qi)>>16);
+      qexp=qexp*2+m;
+
+      pi*=(1<<14)-wi;
+      qi*=(1<<14)+wi;
+      qi=(qi+pi)>>14;
+
+    }
+
+
+    /* we've let the normalization drift because it wasn't important;
+       however, for the lookup, things must be normalized again.  We
+       need at most one right shift or a number of left shifts */
+
+    if(qi&0xffff0000){ /* checks for 1.xxxxxxxxxxxxxxxx */
+      qi>>=1; qexp++;
+    }else
+      while(qi && !(qi&0x8000)){ /* checks for 0.0xxxxxxxxxxxxxxx or less*/
+        qi<<=1; qexp--;
+      }
+
+    amp=vorbis_fromdBlook_i(ampi*                     /*  n.4         */
+                            vorbis_invsqlook_i(qi,qexp)-
+                                                      /*  m.8, m+n<=8 */
+                            ampoffseti);              /*  8.12[0]     */
+
+    curve[i]*=amp;
+    while(map[++i]==k)curve[i]*=amp;
+  }
+}
+
+#else
+
+/* old, nonoptimized but simple version for any poor sap who needs to
+   figure out what the hell this code does, or wants the other
+   fraction of a dB precision */
+
+/* side effect: changes *lsp to cosines of lsp */
+void vorbis_lsp_to_curve(float *curve,int *map,int n,int ln,float *lsp,int m,
+                            float amp,float ampoffset){
+  int i;
+  float wdel=M_PI/ln;
+  for(i=0;i<m;i++)lsp[i]=2.f*cos(lsp[i]);
+
+  i=0;
+  while(i<n){
+    int j,k=map[i];
+    float p=.5f;
+    float q=.5f;
+    float w=2.f*cos(wdel*k);
+    for(j=1;j<m;j+=2){
+      q *= w-lsp[j-1];
+      p *= w-lsp[j];
+    }
+    if(j==m){
+      /* odd order filter; slightly assymetric */
+      /* the last coefficient */
+      q*=w-lsp[j-1];
+      p*=p*(4.f-w*w);
+      q*=q;
+    }else{
+      /* even order filter; still symmetric */
+      p*=p*(2.f-w);
+      q*=q*(2.f+w);
+    }
+
+    q=fromdB(amp/sqrt(p+q)-ampoffset);
+
+    curve[i]*=q;
+    while(map[++i]==k)curve[i]*=q;
+  }
+}
+
+#endif
+#endif
+
+static void cheby(float *g, int ord) {
+  int i, j;
+
+  g[0] *= .5f;
+  for(i=2; i<= ord; i++) {
+    for(j=ord; j >= i; j--) {
+      g[j-2] -= g[j];
+      g[j] += g[j];
+    }
+  }
+}
+
+static int JUCE_CDECL comp(const void *a,const void *b){
+  return (*(float *)a<*(float *)b)-(*(float *)a>*(float *)b);
+}
+
+/* Newton-Raphson-Maehly actually functioned as a decent root finder,
+   but there are root sets for which it gets into limit cycles
+   (exacerbated by zero suppression) and fails.  We can't afford to
+   fail, even if the failure is 1 in 100,000,000, so we now use
+   Laguerre and later polish with Newton-Raphson (which can then
+   afford to fail) */
+
+#define EPSILON 10e-7
+static int Laguerre_With_Deflation(float *a,int ord,float *r){
+  int i,m;
+  double *defl=(double*)alloca(sizeof(*defl)*(ord+1));
+  for(i=0;i<=ord;i++)defl[i]=a[i];
+
+  for(m=ord;m>0;m--){
+    double newx=0.f,delta;
+
+    /* iterate a root */
+    while(1){
+      double p=defl[m],pp=0.f,ppp=0.f,denom;
+
+      /* eval the polynomial and its first two derivatives */
+      for(i=m;i>0;i--){
+	ppp = newx*ppp + pp;
+	pp  = newx*pp  + p;
+	p   = newx*p   + defl[i-1];
+      }
+
+      /* Laguerre's method */
+      denom=(m-1) * ((m-1)*pp*pp - m*p*ppp);
+      if(denom<0)
+        return(-1);  /* complex root!  The LPC generator handed us a bad filter */
+
+      if(pp>0){
+        denom = pp + sqrt(denom);
+        if(denom<EPSILON)denom=EPSILON;
+      }else{
+        denom = pp - sqrt(denom);
+        if(denom>-(EPSILON))denom=-(EPSILON);
+      }
+
+      delta  = m*p/denom;
+      newx   -= delta;
+
+      if(delta<0.f)delta*=-1;
+
+      if(fabs(delta/newx)<10e-12)break;
+    }
+
+    r[m-1]=newx;
+
+    /* forward deflation */
+
+    for(i=m;i>0;i--)
+      defl[i-1]+=newx*defl[i];
+    defl++;
+
+  }
+  return(0);
+}
+
+
+/* for spit-and-polish only */
+static int Newton_Raphson(float *a,int ord,float *r){
+  int i, k, count=0;
+  double error=1.f;
+  double *root=(double*)alloca(ord*sizeof(*root));
+
+  for(i=0; i<ord;i++) root[i] = r[i];
+
+  while(error>1e-20){
+    error=0;
+
+    for(i=0; i<ord; i++) { /* Update each point. */
+      double pp=0.,delta;
+      double rooti=root[i];
+      double p=a[ord];
+      for(k=ord-1; k>= 0; k--) {
+
+        pp= pp* rooti + p;
+        p = p * rooti + a[k];
+      }
+
+      delta = p/pp;
+      root[i] -= delta;
+      error+= delta*delta;
+    }
+
+    if(count>40)return(-1);
+
+    count++;
+  }
+
+  /* Replaced the original bubble sort with a real sort.  With your
+     help, we can eliminate the bubble sort in our lifetime. --Monty */
+
+  for(i=0; i<ord;i++) r[i] = root[i];
+  return(0);
+}
+
+
+/* Convert lpc coefficients to lsp coefficients */
+int vorbis_lpc_to_lsp(float *lpc,float *lsp,int m){
+  int order2=(m+1)>>1;
+  int g1_order,g2_order;
+  float *g1=(float*)alloca(sizeof(*g1)*(order2+1));
+  float *g2=(float*)alloca(sizeof(*g2)*(order2+1));
+  float *g1r=(float*)alloca(sizeof(*g1r)*(order2+1));
+  float *g2r=(float*)alloca(sizeof(*g2r)*(order2+1));
+  int i;
+
+  /* even and odd are slightly different base cases */
+  g1_order=(m+1)>>1;
+  g2_order=(m)  >>1;
+
+  /* Compute the lengths of the x polynomials. */
+  /* Compute the first half of K & R F1 & F2 polynomials. */
+  /* Compute half of the symmetric and antisymmetric polynomials. */
+  /* Remove the roots at +1 and -1. */
+
+  g1[g1_order] = 1.f;
+  for(i=1;i<=g1_order;i++) g1[g1_order-i] = lpc[i-1]+lpc[m-i];
+  g2[g2_order] = 1.f;
+  for(i=1;i<=g2_order;i++) g2[g2_order-i] = lpc[i-1]-lpc[m-i];
+
+  if(g1_order>g2_order){
+    for(i=2; i<=g2_order;i++) g2[g2_order-i] += g2[g2_order-i+2];
+  }else{
+    for(i=1; i<=g1_order;i++) g1[g1_order-i] -= g1[g1_order-i+1];
+    for(i=1; i<=g2_order;i++) g2[g2_order-i] += g2[g2_order-i+1];
+  }
+
+  /* Convert into polynomials in cos(alpha) */
+  cheby(g1,g1_order);
+  cheby(g2,g2_order);
+
+  /* Find the roots of the 2 even polynomials.*/
+  if(Laguerre_With_Deflation(g1,g1_order,g1r) ||
+     Laguerre_With_Deflation(g2,g2_order,g2r))
+    return(-1);
+
+  Newton_Raphson(g1,g1_order,g1r); /* if it fails, it leaves g1r alone */
+  Newton_Raphson(g2,g2_order,g2r); /* if it fails, it leaves g2r alone */
+
+  qsort(g1r,g1_order,sizeof(*g1r),comp);
+  qsort(g2r,g2_order,sizeof(*g2r),comp);
+
+  for(i=0;i<g1_order;i++)
+    lsp[i*2] = acos(g1r[i]);
+
+  for(i=0;i<g2_order;i++)
+    lsp[i*2+1] = acos(g2r[i]);
+  return(0);
+}

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/lsp.h

@@ -0,0 +1,28 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+  function: LSP (also called LSF) conversion routines
+  last mod: $Id: lsp.h 16227 2009-07-08 06:58:46Z xiphmont $
+
+ ********************************************************************/
+
+
+#ifndef _V_LSP_H_
+#define _V_LSP_H_
+
+extern int vorbis_lpc_to_lsp(float *lpc,float *lsp,int m);
+
+extern void vorbis_lsp_to_curve(float *curve,int *map,int n,int ln,
+                                float *lsp,int m,
+                                float amp,float ampoffset);
+
+#endif

source/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/mapping0.c

@@ -0,0 +1,816 @@
+/********************************************************************
+ *                                                                  *
+ * THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE.   *
+ * USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS     *
+ * GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
+ * IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING.       *
+ *                                                                  *
+ * THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2010             *
+ * by the Xiph.Org Foundation http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: channel mapping 0 implementation
+ last mod: $Id: mapping0.c 17022 2010-03-25 03:45:42Z xiphmont $
+
+ ********************************************************************/
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+#include "../../ogg.h"
+#include "../../codec.h"
+#include "codec_internal.h"
+#include "codebook.h"
+#include "window.h"
+#include "registry.h"
+#include "psy.h"
+#include "misc.h"
+
+/* simplistic, wasteful way of doing this (unique lookup for each
+   mode/submapping); there should be a central repository for
+   identical lookups.  That will require minor work, so I'm putting it
+   off as low priority.
+
+   Why a lookup for each backend in a given mode?  Because the
+   blocksize is set by the mode, and low backend lookups may require
+   parameters from other areas of the mode/mapping */
+
+static void mapping0_free_info(vorbis_info_mapping *i){
+  vorbis_info_mapping0 *info=(vorbis_info_mapping0 *)i;
+  if(info){
+    memset(info,0,sizeof(*info));
+    _ogg_free(info);
+  }
+}
+
+static int ilog3(unsigned int v){
+  int ret=0;
+  if(v)--v;
+  while(v){
+    ret++;
+    v>>=1;
+  }
+  return(ret);
+}
+
+static void mapping0_pack(vorbis_info *vi,vorbis_info_mapping *vm,
+                          oggpack_buffer *opb){
+  int i;
+  vorbis_info_mapping0 *info=(vorbis_info_mapping0 *)vm;
+
+  /* another 'we meant to do it this way' hack...  up to beta 4, we
+     packed 4 binary zeros here to signify one submapping in use.  We
+     now redefine that to mean four bitflags that indicate use of
+     deeper features; bit0:submappings, bit1:coupling,
+     bit2,3:reserved. This is backward compatable with all actual uses
+     of the beta code. */
+
+  if(info->submaps>1){
+    oggpack_write(opb,1,1);
+    oggpack_write(opb,info->submaps-1,4);
+  }else
+    oggpack_write(opb,0,1);
+
+  if(info->coupling_steps>0){
+    oggpack_write(opb,1,1);
+    oggpack_write(opb,info->coupling_steps-1,8);
+
+    for(i=0;i<info->coupling_steps;i++){
+      oggpack_write(opb,info->coupling_mag[i],ilog3(vi->channels));
+      oggpack_write(opb,info->coupling_ang[i],ilog3(vi->channels));
+    }
+  }else
+    oggpack_write(opb,0,1);
+
+  oggpack_write(opb,0,2); /* 2,3:reserved */
+
+  /* we don't write the channel submappings if we only have one... */
+  if(info->submaps>1){
+    for(i=0;i<vi->channels;i++)
+      oggpack_write(opb,info->chmuxlist[i],4);
+  }
+  for(i=0;i<info->submaps;i++){
+    oggpack_write(opb,0,8); /* time submap unused */
+    oggpack_write(opb,info->floorsubmap[i],8);
+    oggpack_write(opb,info->residuesubmap[i],8);
+  }
+}
+
+/* also responsible for range checking */
+static vorbis_info_mapping *mapping0_unpack(vorbis_info *vi,oggpack_buffer *opb){
+  int i,b;
+  vorbis_info_mapping0 *info=(vorbis_info_mapping0*)_ogg_calloc(1,sizeof(*info));
+  codec_setup_info     *ci=(codec_setup_info*)vi->codec_setup;
+  memset(info,0,sizeof(*info));
+
+  b=oggpack_read(opb,1);
+  if(b<0)goto err_out;
+  if(b){
+    info->submaps=oggpack_read(opb,4)+1;
+    if(info->submaps<=0)goto err_out;
+  }else
+    info->submaps=1;
+
+  b=oggpack_read(opb,1);
+  if(b<0)goto err_out;
+  if(b){
+    info->coupling_steps=oggpack_read(opb,8)+1;
+    if(info->coupling_steps<=0)goto err_out;
+    for(i=0;i<info->coupling_steps;i++){
+      int testM=info->coupling_mag[i]=oggpack_read(opb,ilog3(vi->channels));
+      int testA=info->coupling_ang[i]=oggpack_read(opb,ilog3(vi->channels));
+
+      if(testM<0 ||
+         testA<0 ||
+         testM==testA ||
+         testM>=vi->channels ||
+         testA>=vi->channels) goto err_out;
+    }
+
+  }
+
+  if(oggpack_read(opb,2)!=0)goto err_out; /* 2,3:reserved */
+
+  if(info->submaps>1){
+    for(i=0;i<vi->channels;i++){
+      info->chmuxlist[i]=oggpack_read(opb,4);
+      if(info->chmuxlist[i]>=info->submaps || info->chmuxlist[i]<0)goto err_out;
+    }
+  }
+  for(i=0;i<info->submaps;i++){
+    oggpack_read(opb,8); /* time submap unused */
+    info->floorsubmap[i]=oggpack_read(opb,8);
+    if(info->floorsubmap[i]>=ci->floors || info->floorsubmap[i]<0)goto err_out;
+    info->residuesubmap[i]=oggpack_read(opb,8);
+    if(info->residuesubmap[i]>=ci->residues || info->residuesubmap[i]<0)goto err_out;
+  }
+
+  return info;
+
+ err_out:
+  mapping0_free_info(info);
+  return(NULL);
+}
+
+#include "os.h"
+#include "lpc.h"
+#include "lsp.h"
+#include "envelope.h"
+#include "mdct.h"
+#include "psy.h"
+#include "scales.h"
+
+#if 0
+static long seq=0;
+static ogg_int64_t total=0;
+static float FLOOR1_fromdB_LOOKUP[256]={
+  1.0649863e-07F, 1.1341951e-07F, 1.2079015e-07F, 1.2863978e-07F,
+  1.3699951e-07F, 1.4590251e-07F, 1.5538408e-07F, 1.6548181e-07F,
+  1.7623575e-07F, 1.8768855e-07F, 1.9988561e-07F, 2.128753e-07F,
+  2.2670913e-07F, 2.4144197e-07F, 2.5713223e-07F, 2.7384213e-07F,
+  2.9163793e-07F, 3.1059021e-07F, 3.3077411e-07F, 3.5226968e-07F,
+  3.7516214e-07F, 3.9954229e-07F, 4.2550680e-07F, 4.5315863e-07F,
+  4.8260743e-07F, 5.1396998e-07F, 5.4737065e-07F, 5.8294187e-07F,
+  6.2082472e-07F, 6.6116941e-07F, 7.0413592e-07F, 7.4989464e-07F,
+  7.9862701e-07F, 8.5052630e-07F, 9.0579828e-07F, 9.6466216e-07F,
+  1.0273513e-06F, 1.0941144e-06F, 1.1652161e-06F, 1.2409384e-06F,
+  1.3215816e-06F, 1.4074654e-06F, 1.4989305e-06F, 1.5963394e-06F,
+  1.7000785e-06F, 1.8105592e-06F, 1.9282195e-06F, 2.0535261e-06F,
+  2.1869758e-06F, 2.3290978e-06F, 2.4804557e-06F, 2.6416497e-06F,
+  2.8133190e-06F, 2.9961443e-06F, 3.1908506e-06F, 3.3982101e-06F,
+  3.6190449e-06F, 3.8542308e-06F, 4.1047004e-06F, 4.3714470e-06F,
+  4.6555282e-06F, 4.9580707e-06F, 5.2802740e-06F, 5.6234160e-06F,
+  5.9888572e-06F, 6.3780469e-06F, 6.7925283e-06F, 7.2339451e-06F,
+  7.7040476e-06F, 8.2047000e-06F, 8.7378876e-06F, 9.3057248e-06F,
+  9.9104632e-06F, 1.0554501e-05F, 1.1240392e-05F, 1.1970856e-05F,
+  1.2748789e-05F, 1.3577278e-05F, 1.4459606e-05F, 1.5399272e-05F,
+  1.6400004e-05F, 1.7465768e-05F, 1.8600792e-05F, 1.9809576e-05F,
+  2.1096914e-05F, 2.2467911e-05F, 2.3928002e-05F, 2.5482978e-05F,
+  2.7139006e-05F, 2.8902651e-05F, 3.0780908e-05F, 3.2781225e-05F,
+  3.4911534e-05F, 3.7180282e-05F, 3.9596466e-05F, 4.2169667e-05F,
+  4.4910090e-05F, 4.7828601e-05F, 5.0936773e-05F, 5.4246931e-05F,
+  5.7772202e-05F, 6.1526565e-05F, 6.5524908e-05F, 6.9783085e-05F,
+  7.4317983e-05F, 7.9147585e-05F, 8.4291040e-05F, 8.9768747e-05F,
+  9.5602426e-05F, 0.00010181521F, 0.00010843174F, 0.00011547824F,
+  0.00012298267F, 0.00013097477F, 0.00013948625F, 0.00014855085F,
+  0.00015820453F, 0.00016848555F, 0.00017943469F, 0.00019109536F,
+  0.00020351382F, 0.00021673929F, 0.00023082423F, 0.00024582449F,
+  0.00026179955F, 0.00027881276F, 0.00029693158F, 0.00031622787F,
+  0.00033677814F, 0.00035866388F, 0.00038197188F, 0.00040679456F,
+  0.00043323036F, 0.00046138411F, 0.00049136745F, 0.00052329927F,
+  0.00055730621F, 0.00059352311F, 0.00063209358F, 0.00067317058F,
+  0.00071691700F, 0.00076350630F, 0.00081312324F, 0.00086596457F,
+  0.00092223983F, 0.00098217216F, 0.0010459992F, 0.0011139742F,
+  0.0011863665F, 0.0012634633F, 0.0013455702F, 0.0014330129F,
+  0.0015261382F, 0.0016253153F, 0.0017309374F, 0.0018434235F,
+  0.0019632195F, 0.0020908006F, 0.0022266726F, 0.0023713743F,
+  0.0025254795F, 0.0026895994F, 0.0028643847F, 0.0030505286F,
+  0.0032487691F, 0.0034598925F, 0.0036847358F, 0.0039241906F,
+  0.0041792066F, 0.0044507950F, 0.0047400328F, 0.0050480668F,
+  0.0053761186F, 0.0057254891F, 0.0060975636F, 0.0064938176F,
+  0.0069158225F, 0.0073652516F, 0.0078438871F, 0.0083536271F,
+  0.0088964928F, 0.009474637F, 0.010090352F, 0.010746080F,
+  0.011444421F, 0.012188144F, 0.012980198F, 0.013823725F,
+  0.014722068F, 0.015678791F, 0.016697687F, 0.017782797F,
+  0.018938423F, 0.020169149F, 0.021479854F, 0.022875735F,
+  0.024362330F, 0.025945531F, 0.027631618F, 0.029427276F,
+  0.031339626F, 0.033376252F, 0.035545228F, 0.037855157F,
+  0.040315199F, 0.042935108F, 0.045725273F, 0.048696758F,
+  0.051861348F, 0.055231591F, 0.058820850F, 0.062643361F,
+  0.066714279F, 0.071049749F, 0.075666962F, 0.080584227F,
+  0.085821044F, 0.091398179F, 0.097337747F, 0.10366330F,
+  0.11039993F, 0.11757434F, 0.12521498F, 0.13335215F,
+  0.14201813F, 0.15124727F, 0.16107617F, 0.17154380F,
+  0.18269168F, 0.19456402F, 0.20720788F, 0.22067342F,
+  0.23501402F, 0.25028656F, 0.26655159F, 0.28387361F,
+  0.30232132F, 0.32196786F, 0.34289114F, 0.36517414F,
+  0.38890521F, 0.41417847F, 0.44109412F, 0.46975890F,
+  0.50028648F, 0.53279791F, 0.56742212F, 0.60429640F,
+  0.64356699F, 0.68538959F, 0.72993007F, 0.77736504F,
+  0.82788260F, 0.88168307F, 0.9389798F, 1.F,
+};
+
+#endif
+
+
+static int mapping0_forward(vorbis_block *vb){
+  vorbis_dsp_state      *vd=vb->vd;
+  vorbis_info           *vi=vd->vi;
+  codec_setup_info      *ci=(codec_setup_info*)vi->codec_setup;
+  private_state         *b=(private_state*)vb->vd->backend_state;
+  vorbis_block_internal *vbi=(vorbis_block_internal *)vb->internal;
+  int                    n=vb->pcmend;
+  int i,j,k;
+
+  int    *nonzero    = (int*)alloca(sizeof(*nonzero)*vi->channels);
+  float  **gmdct     = (float**)_vorbis_block_alloc(vb,vi->channels*sizeof(*gmdct));
+  int    **iwork      = (int**)_vorbis_block_alloc(vb,vi->channels*sizeof(*iwork));
+  int ***floor_posts = (int***)_vorbis_block_alloc(vb,vi->channels*sizeof(*floor_posts));
+
+  float global_ampmax=vbi->ampmax;
+  float *local_ampmax=(float*)alloca(sizeof(*local_ampmax)*vi->channels);
+  int blocktype=vbi->blocktype;
+
+  int modenumber=vb->W;
+  vorbis_info_mapping0 *info=(vorbis_info_mapping0*)ci->map_param[modenumber];
+  vorbis_look_psy *psy_look=b->psy+blocktype+(vb->W?2:0);
+
+  vb->mode=modenumber;
+
+  for(i=0;i<vi->channels;i++){
+    float scale=4.f/n;
+    float scale_dB;
+
+    float *pcm     =vb->pcm[i];
+    float *logfft  =pcm;
+
+    iwork[i]=(int*)_vorbis_block_alloc(vb,n/2*sizeof(**iwork));
+    gmdct[i]=(float*)_vorbis_block_alloc(vb,n/2*sizeof(**gmdct));
+
+    scale_dB=todB(&scale) + .345; /* + .345 is a hack; the original
+                                     todB estimation used on IEEE 754
+                                     compliant machines had a bug that
+                                     returned dB values about a third
+                                     of a decibel too high.  The bug
+                                     was harmless because tunings
+                                     implicitly took that into
+                                     account.  However, fixing the bug
+                                     in the estimator requires
+                                     changing all the tunings as well.
+                                     For now, it's easier to sync
+                                     things back up here, and
+                                     recalibrate the tunings in the
+                                     next major model upgrade. */
+
+#if 0
+    if(vi->channels==2){
+      if(i==0)
+        _analysis_output("pcmL",seq,pcm,n,0,0,total-n/2);
+      else
+        _analysis_output("pcmR",seq,pcm,n,0,0,total-n/2);
+    }else{
+      _analysis_output("pcm",seq,pcm,n,0,0,total-n/2);
+    }
+#endif
+
+    /* window the PCM data */
+    _vorbis_apply_window(pcm,b->window,ci->blocksizes,vb->lW,vb->W,vb->nW);
+
+#if 0
+    if(vi->channels==2){
+      if(i==0)
+        _analysis_output("windowedL",seq,pcm,n,0,0,total-n/2);
+      else
+        _analysis_output("windowedR",seq,pcm,n,0,0,total-n/2);
+    }else{
+      _analysis_output("windowed",seq,pcm,n,0,0,total-n/2);
+    }
+#endif
+
+    /* transform the PCM data */
+    /* only MDCT right now.... */
+    mdct_forward((mdct_lookup*) b->transform[vb->W][0],pcm,gmdct[i]);
+
+    /* FFT yields more accurate tonal estimation (not phase sensitive) */
+    drft_forward(&b->fft_look[vb->W],pcm);
+    logfft[0]=scale_dB+todB(pcm)  + .345; /* + .345 is a hack; the
+                                     original todB estimation used on
+                                     IEEE 754 compliant machines had a
+                                     bug that returned dB values about
+                                     a third of a decibel too high.
+                                     The bug was harmless because
+                                     tunings implicitly took that into
+                                     account.  However, fixing the bug
+                                     in the estimator requires
+                                     changing all the tunings as well.
+                                     For now, it's easier to sync
+                                     things back up here, and
+                                     recalibrate the tunings in the
+                                     next major model upgrade. */
+    local_ampmax[i]=logfft[0];
+    for(j=1;j<n-1;j+=2){
+      float temp=pcm[j]*pcm[j]+pcm[j+1]*pcm[j+1];
+      temp=logfft[(j+1)>>1]=scale_dB+.5f*todB(&temp)  + .345; /* +
+                                     .345 is a hack; the original todB
+                                     estimation used on IEEE 754
+                                     compliant machines had a bug that
+                                     returned dB values about a third
+                                     of a decibel too high.  The bug
+                                     was harmless because tunings
+                                     implicitly took that into
+                                     account.  However, fixing the bug
+                                     in the estimator requires
+                                     changing all the tunings as well.
+                                     For now, it's easier to sync
+                                     things back up here, and
+                                     recalibrate the tunings in the
+                                     next major model upgrade. */
+      if(temp>local_ampmax[i])local_ampmax[i]=temp;
+    }
+
+    if(local_ampmax[i]>0.f)local_ampmax[i]=0.f;
+    if(local_ampmax[i]>global_ampmax)global_ampmax=local_ampmax[i];
+
+#if 0
+    if(vi->channels==2){
+      if(i==0){
+        _analysis_output("fftL",seq,logfft,n/2,1,0,0);
+      }else{
+        _analysis_output("fftR",seq,logfft,n/2,1,0,0);
+      }
+    }else{
+      _analysis_output("fft",seq,logfft,n/2,1,0,0);
+    }
+#endif
+
+  }
+
+  {
+    float   *noise        = (float*)_vorbis_block_alloc(vb,n/2*sizeof(*noise));
+    float   *tone         = (float*)_vorbis_block_alloc(vb,n/2*sizeof(*tone));
+
+    for(i=0;i<vi->channels;i++){
+      /* the encoder setup assumes that all the modes used by any
+         specific bitrate tweaking use the same floor */
+
+      int submap=info->chmuxlist[i];
+
+      /* the following makes things clearer to *me* anyway */
+      float *mdct    =gmdct[i];
+      float *logfft  =vb->pcm[i];
+
+      float *logmdct =logfft+n/2;
+      float *logmask =logfft;
+
+      vb->mode=modenumber;
+
+      floor_posts[i]=(int**)_vorbis_block_alloc(vb,PACKETBLOBS*sizeof(**floor_posts));
+      memset(floor_posts[i],0,sizeof(**floor_posts)*PACKETBLOBS);
+
+      for(j=0;j<n/2;j++)
+        logmdct[j]=todB(mdct+j)  + .345; /* + .345 is a hack; the original
+                                     todB estimation used on IEEE 754
+                                     compliant machines had a bug that
+                                     returned dB values about a third
+                                     of a decibel too high.  The bug
+                                     was harmless because tunings
+                                     implicitly took that into
+                                     account.  However, fixing the bug
+                                     in the estimator requires
+                                     changing all the tunings as well.
+                                     For now, it's easier to sync
+                                     things back up here, and
+                                     recalibrate the tunings in the
+                                     next major model upgrade. */
+
+#if 0
+      if(vi->channels==2){
+        if(i==0)
+          _analysis_output("mdctL",seq,logmdct,n/2,1,0,0);
+        else
+          _analysis_output("mdctR",seq,logmdct,n/2,1,0,0);
+      }else{
+        _analysis_output("mdct",seq,logmdct,n/2,1,0,0);
+      }
+#endif
+
+      /* first step; noise masking.  Not only does 'noise masking'
+         give us curves from which we can decide how much resolution
+         to give noise parts of the spectrum, it also implicitly hands
+         us a tonality estimate (the larger the value in the
+         'noise_depth' vector, the more tonal that area is) */
+
+      _vp_noisemask(psy_look,
+                    logmdct,
+                    noise); /* noise does not have by-frequency offset
+                               bias applied yet */
+#if 0
+      if(vi->channels==2){
+        if(i==0)
+          _analysis_output("noiseL",seq,noise,n/2,1,0,0);
+        else
+          _analysis_output("noiseR",seq,noise,n/2,1,0,0);
+      }else{
+        _analysis_output("noise",seq,noise,n/2,1,0,0);
+      }
+#endif
+
+      /* second step: 'all the other crap'; all the stuff that isn't
+         computed/fit for bitrate management goes in the second psy
+         vector.  This includes tone masking, peak limiting and ATH */
+
+      _vp_tonemask(psy_look,
+                   logfft,
+                   tone,
+                   global_ampmax,
+                   local_ampmax[i]);
+
+#if 0
+      if(vi->channels==2){
+        if(i==0)
+          _analysis_output("toneL",seq,tone,n/2,1,0,0);
+        else
+          _analysis_output("toneR",seq,tone,n/2,1,0,0);
+      }else{
+        _analysis_output("tone",seq,tone,n/2,1,0,0);
+      }
+#endif
+
+      /* third step; we offset the noise vectors, overlay tone
+         masking.  We then do a floor1-specific line fit.  If we're
+         performing bitrate management, the line fit is performed
+         multiple times for up/down tweakage on demand. */
+
+#if 0
+      {
+      float aotuv[psy_look->n];
+#endif
+
+        _vp_offset_and_mix(psy_look,
+                           noise,
+                           tone,
+                           1,
+                           logmask,
+                           mdct,
+                           logmdct);
+
+#if 0
+        if(vi->channels==2){
+          if(i==0)
+            _analysis_output("aotuvM1_L",seq,aotuv,psy_look->n,1,1,0);
+          else
+            _analysis_output("aotuvM1_R",seq,aotuv,psy_look->n,1,1,0);
+        }else{
+          _analysis_output("aotuvM1",seq,aotuv,psy_look->n,1,1,0);
+        }
+      }
+#endif
+
+
+#if 0
+      if(vi->channels==2){
+        if(i==0)
+          _analysis_output("mask1L",seq,logmask,n/2,1,0,0);
+        else
+          _analysis_output("mask1R",seq,logmask,n/2,1,0,0);
+      }else{
+        _analysis_output("mask1",seq,logmask,n/2,1,0,0);
+      }
+#endif
+
+      /* this algorithm is hardwired to floor 1 for now; abort out if
+         we're *not* floor1.  This won't happen unless someone has
+         broken the encode setup lib.  Guard it anyway. */
+      if(ci->floor_type[info->floorsubmap[submap]]!=1)return(-1);
+
+      floor_posts[i][PACKETBLOBS/2]=
+        floor1_fit(vb,(vorbis_look_floor1*)(b->flr[info->floorsubmap[submap]]),
+                   logmdct,
+                   logmask);
+
+      /* are we managing bitrate?  If so, perform two more fits for
+         later rate tweaking (fits represent hi/lo) */
+      if(vorbis_bitrate_managed(vb) && floor_posts[i][PACKETBLOBS/2]){
+        /* higher rate by way of lower noise curve */
+
+        _vp_offset_and_mix(psy_look,
+                           noise,
+                           tone,
+                           2,
+                           logmask,
+                           mdct,
+                           logmdct);
+
+#if 0
+        if(vi->channels==2){
+          if(i==0)
+            _analysis_output("mask2L",seq,logmask,n/2,1,0,0);
+          else
+            _analysis_output("mask2R",seq,logmask,n/2,1,0,0);
+        }else{
+          _analysis_output("mask2",seq,logmask,n/2,1,0,0);
+        }
+#endif
+
+        floor_posts[i][PACKETBLOBS-1]=
+          floor1_fit(vb,(vorbis_look_floor1*)(b->flr[info->floorsubmap[submap]]),
+                     logmdct,
+                     logmask);
+
+        /* lower rate by way of higher noise curve */
+        _vp_offset_and_mix(psy_look,
+                           noise,
+                           tone,
+                           0,
+                           logmask,
+                           mdct,
+                           logmdct);
+
+#if 0
+        if(vi->channels==2){
+          if(i==0)
+            _analysis_output("mask0L",seq,logmask,n/2,1,0,0);
+          else
+            _analysis_output("mask0R",seq,logmask,n/2,1,0,0);
+        }else{
+          _analysis_output("mask0",seq,logmask,n/2,1,0,0);
+        }
+#endif
+
+        floor_posts[i][0]=
+          floor1_fit(vb,(vorbis_look_floor1*)(b->flr[info->floorsubmap[submap]]),
+                     logmdct,
+                     logmask);
+
+        /* we also interpolate a range of intermediate curves for
+           intermediate rates */
+        for(k=1;k<PACKETBLOBS/2;k++)
+          floor_posts[i][k]=
+            floor1_interpolate_fit(vb,(vorbis_look_floor1*)(b->flr[info->floorsubmap[submap]]),
+                                   floor_posts[i][0],
+                                   floor_posts[i][PACKETBLOBS/2],
+                                   k*65536/(PACKETBLOBS/2));
+        for(k=PACKETBLOBS/2+1;k<PACKETBLOBS-1;k++)
+          floor_posts[i][k]=
+            floor1_interpolate_fit(vb,(vorbis_look_floor1*)(b->flr[info->floorsubmap[submap]]),
+                                   floor_posts[i][PACKETBLOBS/2],
+                                   floor_posts[i][PACKETBLOBS-1],
+                                   (k-PACKETBLOBS/2)*65536/(PACKETBLOBS/2));
+      }
+    }
+  }
+  vbi->ampmax=global_ampmax;
+
+  /*
+    the next phases are performed once for vbr-only and PACKETBLOB
+    times for bitrate managed modes.
+
+    1) encode actual mode being used
+    2) encode the floor for each channel, compute coded mask curve/res
+    3) normalize and couple.
+    4) encode residue
+    5) save packet bytes to the packetblob vector
+
+  */
+
+  /* iterate over the many masking curve fits we've created */
+
+  {
+    int **couple_bundle=(int**)alloca(sizeof(*couple_bundle)*vi->channels);
+    int *zerobundle=(int*)alloca(sizeof(*zerobundle)*vi->channels);
+
+    for(k=(vorbis_bitrate_managed(vb)?0:PACKETBLOBS/2);
+        k<=(vorbis_bitrate_managed(vb)?PACKETBLOBS-1:PACKETBLOBS/2);
+        k++){
+      oggpack_buffer *opb=vbi->packetblob[k];
+
+      /* start out our new packet blob with packet type and mode */
+      /* Encode the packet type */
+      oggpack_write(opb,0,1);
+      /* Encode the modenumber */
+      /* Encode frame mode, pre,post windowsize, then dispatch */
+      oggpack_write(opb,modenumber,b->modebits);
+      if(vb->W){
+        oggpack_write(opb,vb->lW,1);
+        oggpack_write(opb,vb->nW,1);
+      }
+
+      /* encode floor, compute masking curve, sep out residue */
+      for(i=0;i<vi->channels;i++){
+        int submap=info->chmuxlist[i];
+        int *ilogmask=iwork[i];
+
+        nonzero[i]=floor1_encode(opb,vb,(vorbis_look_floor1*)(b->flr[info->floorsubmap[submap]]),
+                                 floor_posts[i][k],
+                                 ilogmask);
+#if 0
+        {
+          char buf[80];
+          sprintf(buf,"maskI%c%d",i?'R':'L',k);
+          float work[n/2];
+          for(j=0;j<n/2;j++)
+            work[j]=FLOOR1_fromdB_LOOKUP[iwork[i][j]];
+          _analysis_output(buf,seq,work,n/2,1,1,0);
+        }
+#endif
+      }
+
+      /* our iteration is now based on masking curve, not prequant and
+         coupling.  Only one prequant/coupling step */
+
+      /* quantize/couple */
+      /* incomplete implementation that assumes the tree is all depth
+         one, or no tree at all */
+      _vp_couple_quantize_normalize(k,
+                                    &ci->psy_g_param,
+                                    psy_look,
+                                    info,
+                                    gmdct,
+                                    iwork,
+                                    nonzero,
+                                    ci->psy_g_param.sliding_lowpass[vb->W][k],
+                                    vi->channels);
+
+#if 0
+      for(i=0;i<vi->channels;i++){
+        char buf[80];
+        sprintf(buf,"res%c%d",i?'R':'L',k);
+        float work[n/2];
+        for(j=0;j<n/2;j++)
+          work[j]=iwork[i][j];
+        _analysis_output(buf,seq,work,n/2,1,0,0);
+      }
+#endif
+
+      /* classify and encode by submap */
+      for(i=0;i<info->submaps;i++){
+        int ch_in_bundle=0;
+        long **classifications;
+        int resnum=info->residuesubmap[i];
+
+        for(j=0;j<vi->channels;j++){
+          if(info->chmuxlist[j]==i){
+            zerobundle[ch_in_bundle]=0;
+            if(nonzero[j])zerobundle[ch_in_bundle]=1;
+            couple_bundle[ch_in_bundle++]=iwork[j];
+          }
+        }
+
+        classifications=_residue_P[ci->residue_type[resnum]]->
+          classx(vb,b->residue[resnum],couple_bundle,zerobundle,ch_in_bundle);
+
+        ch_in_bundle=0;
+        for(j=0;j<vi->channels;j++)
+          if(info->chmuxlist[j]==i)
+            couple_bundle[ch_in_bundle++]=iwork[j];
+
+        _residue_P[ci->residue_type[resnum]]->
+          forward(opb,vb,b->residue[resnum],
+                  couple_bundle,zerobundle,ch_in_bundle,classifications,i);
+      }
+
+      /* ok, done encoding.  Next protopacket. */
+    }
+
+  }
+
+#if 0
+  seq++;
+  total+=ci->blocksizes[vb->W]/4+ci->blocksizes[vb->nW]/4;
+#endif
+  return(0);
+}
+
+static int mapping0_inverse(vorbis_block *vb,vorbis_info_mapping *l){
+  vorbis_dsp_state     *vd=vb->vd;
+  vorbis_info          *vi=vd->vi;
+  codec_setup_info     *ci=(codec_setup_info*)vi->codec_setup;
+  private_state        *b=(private_state*)vd->backend_state;
+  vorbis_info_mapping0 *info=(vorbis_info_mapping0 *)l;
+
+  int                   i,j;
+  long                  n=vb->pcmend=ci->blocksizes[vb->W];
+
+  float **pcmbundle=(float**) alloca(sizeof(*pcmbundle)*vi->channels);
+  int    *zerobundle=(int*) alloca(sizeof(*zerobundle)*vi->channels);
+
+  int   *nonzero  =(int*) alloca(sizeof(*nonzero)*vi->channels);
+  void **floormemo=(void**) alloca(sizeof(*floormemo)*vi->channels);
+
+  /* recover the spectral envelope; store it in the PCM vector for now */
+  for(i=0;i<vi->channels;i++){
+    int submap=info->chmuxlist[i];
+    floormemo[i]=_floor_P[ci->floor_type[info->floorsubmap[submap]]]->
+      inverse1(vb,b->flr[info->floorsubmap[submap]]);
+    if(floormemo[i])
+      nonzero[i]=1;
+    else
+      nonzero[i]=0;
+    memset(vb->pcm[i],0,sizeof(*vb->pcm[i])*n/2);
+  }
+
+  /* channel coupling can 'dirty' the nonzero listing */
+  for(i=0;i<info->coupling_steps;i++){
+    if(nonzero[info->coupling_mag[i]] ||
+       nonzero[info->coupling_ang[i]]){
+      nonzero[info->coupling_mag[i]]=1;
+      nonzero[info->coupling_ang[i]]=1;
+    }
+  }
+
+  /* recover the residue into our working vectors */
+  for(i=0;i<info->submaps;i++){
+    int ch_in_bundle=0;
+    for(j=0;j<vi->channels;j++){
+      if(info->chmuxlist[j]==i){
+        if(nonzero[j])
+          zerobundle[ch_in_bundle]=1;
+        else
+          zerobundle[ch_in_bundle]=0;
+        pcmbundle[ch_in_bundle++]=vb->pcm[j];
+      }
+    }
+
+    _residue_P[ci->residue_type[info->residuesubmap[i]]]->
+      inverse(vb,b->residue[info->residuesubmap[i]],
+              pcmbundle,zerobundle,ch_in_bundle);
+  }
+
+  /* channel coupling */
+  for(i=info->coupling_steps-1;i>=0;i--){
+    float *pcmM=vb->pcm[info->coupling_mag[i]];
+    float *pcmA=vb->pcm[info->coupling_ang[i]];
+
+    for(j=0;j<n/2;j++){
+      float mag=pcmM[j];
+      float ang=pcmA[j];
+
+      if(mag>0)
+        if(ang>0){
+          pcmM[j]=mag;
+          pcmA[j]=mag-ang;
+        }else{
+          pcmA[j]=mag;
+          pcmM[j]=mag+ang;
+        }
+      else
+        if(ang>0){
+          pcmM[j]=mag;
+          pcmA[j]=mag+ang;
+        }else{
+          pcmA[j]=mag;
+          pcmM[j]=mag-ang;
+        }
+    }
+  }
+
+  /* compute and apply spectral envelope */
+  for(i=0;i<vi->channels;i++){
+    float *pcm=vb->pcm[i];
+    int submap=info->chmuxlist[i];
+    _floor_P[ci->floor_type[info->floorsubmap[submap]]]->
+      inverse2(vb,b->flr[info->floorsubmap[submap]],
+               floormemo[i],pcm);
+  }
+
+  /* transform the PCM data; takes PCM vector, vb; modifies PCM vector */
+  /* only MDCT right now.... */
+  for(i=0;i<vi->channels;i++){
+    float *pcm=vb->pcm[i];
+    mdct_backward((mdct_lookup*) b->transform[vb->W][0],pcm,pcm);
+  }
+
+  /* all done! */
+  return(0);
+}
+
+/* export hooks */
+const vorbis_func_mapping mapping0_exportbundle={
+  &mapping0_pack,
+  &mapping0_unpack,
+  &mapping0_free_info,
+  &mapping0_forward,
+  &mapping0_inverse
+};