(Added missing files from last check-in)

src/audio/audio_file_formats/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.

src/audio/audio_file_formats/flac/all.h

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

src/audio/audio_file_formats/flac/alloc.h

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

src/audio/audio_file_formats/flac/assert.h

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

src/audio/audio_file_formats/flac/callback.h

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

src/audio/audio_file_formats/flac/export.h

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

src/audio/audio_file_formats/flac/libFLAC/bitmath.c

@@ -0,0 +1,155 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "juce_FlacHeader.h"
+#if JUCE_USE_FLAC
+
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include "include/private/bitmath.h"
+#include "../assert.h"
+
+/* An example of what FLAC__bitmath_ilog2() computes:
+ *
+ * ilog2( 0) = assertion failure
+ * ilog2( 1) = 0
+ * ilog2( 2) = 1
+ * ilog2( 3) = 1
+ * ilog2( 4) = 2
+ * ilog2( 5) = 2
+ * ilog2( 6) = 2
+ * ilog2( 7) = 2
+ * ilog2( 8) = 3
+ * ilog2( 9) = 3
+ * ilog2(10) = 3
+ * ilog2(11) = 3
+ * ilog2(12) = 3
+ * ilog2(13) = 3
+ * ilog2(14) = 3
+ * ilog2(15) = 3
+ * ilog2(16) = 4
+ * ilog2(17) = 4
+ * ilog2(18) = 4
+ */
+unsigned FLAC__bitmath_ilog2(FLAC__uint32 v)
+{
+	unsigned l = 0;
+	FLAC__ASSERT(v > 0);
+	while(v >>= 1)
+		l++;
+	return l;
+}
+
+unsigned FLAC__bitmath_ilog2_wide(FLAC__uint64 v)
+{
+	unsigned l = 0;
+	FLAC__ASSERT(v > 0);
+	while(v >>= 1)
+		l++;
+	return l;
+}
+
+/* An example of what FLAC__bitmath_silog2() computes:
+ *
+ * silog2(-10) = 5
+ * silog2(- 9) = 5
+ * silog2(- 8) = 4
+ * silog2(- 7) = 4
+ * silog2(- 6) = 4
+ * silog2(- 5) = 4
+ * silog2(- 4) = 3
+ * silog2(- 3) = 3
+ * silog2(- 2) = 2
+ * silog2(- 1) = 2
+ * silog2(  0) = 0
+ * silog2(  1) = 2
+ * silog2(  2) = 3
+ * silog2(  3) = 3
+ * silog2(  4) = 4
+ * silog2(  5) = 4
+ * silog2(  6) = 4
+ * silog2(  7) = 4
+ * silog2(  8) = 5
+ * silog2(  9) = 5
+ * silog2( 10) = 5
+ */
+unsigned FLAC__bitmath_silog2(int v)
+{
+	while(1) {
+		if(v == 0) {
+			return 0;
+		}
+		else if(v > 0) {
+			unsigned l = 0;
+			while(v) {
+				l++;
+				v >>= 1;
+			}
+			return l+1;
+		}
+		else if(v == -1) {
+			return 2;
+		}
+		else {
+			v++;
+			v = -v;
+		}
+	}
+}
+
+unsigned FLAC__bitmath_silog2_wide(FLAC__int64 v)
+{
+	while(1) {
+		if(v == 0) {
+			return 0;
+		}
+		else if(v > 0) {
+			unsigned l = 0;
+			while(v) {
+				l++;
+				v >>= 1;
+			}
+			return l+1;
+		}
+		else if(v == -1) {
+			return 2;
+		}
+		else {
+			v++;
+			v = -v;
+		}
+	}
+}
+
+#endif

src/audio/audio_file_formats/flac/libFLAC/bitwriter.c

@@ -0,0 +1,885 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "juce_FlacHeader.h"
+#if JUCE_USE_FLAC
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include <stdlib.h> /* for malloc() */
+#include <string.h> /* for memcpy(), memset() */
+#ifdef _MSC_VER
+#include <winsock.h> /* for ntohl() */
+#elif defined FLAC__SYS_DARWIN
+#include <machine/endian.h> /* for ntohl() */
+#elif defined __MINGW32__
+#include <winsock.h> /* for ntohl() */
+#else
+#include <netinet/in.h> /* for ntohl() */
+#endif
+#if 0 /* UNUSED */
+#include "include/private/bitmath.h"
+#endif
+#include "include/private/bitwriter.h"
+#include "include/private/crc.h"
+#include "../assert.h"
+#include "../alloc.h"
+
+/* Things should be fastest when this matches the machine word size */
+/* WATCHOUT: if you change this you must also change the following #defines down to SWAP_BE_WORD_TO_HOST below to match */
+/* WATCHOUT: there are a few places where the code will not work unless bwword is >= 32 bits wide */
+typedef FLAC__uint32 bwword;
+#define FLAC__BYTES_PER_WORD 4
+#define FLAC__BITS_PER_WORD 32
+#define FLAC__WORD_ALL_ONES ((FLAC__uint32)0xffffffff)
+/* SWAP_BE_WORD_TO_HOST swaps bytes in a bwword (which is always big-endian) if necessary to match host byte order */
+#if WORDS_BIGENDIAN
+#define SWAP_BE_WORD_TO_HOST(x) (x)
+#else
+#ifdef _MSC_VER
+#define SWAP_BE_WORD_TO_HOST(x) local_swap32_(x)
+#else
+#define SWAP_BE_WORD_TO_HOST(x) ntohl(x)
+#endif
+#endif
+
+/*
+ * The default capacity here doesn't matter too much.  The buffer always grows
+ * to hold whatever is written to it.  Usually the encoder will stop adding at
+ * a frame or metadata block, then write that out and clear the buffer for the
+ * next one.
+ */
+static const unsigned FLAC__BITWRITER_DEFAULT_CAPACITY = 32768u / sizeof(bwword); /* size in words */
+/* When growing, increment 4K at a time */
+static const unsigned FLAC__BITWRITER_DEFAULT_INCREMENT = 4096u / sizeof(bwword); /* size in words */
+
+#define FLAC__WORDS_TO_BITS(words) ((words) * FLAC__BITS_PER_WORD)
+#define FLAC__TOTAL_BITS(bw) (FLAC__WORDS_TO_BITS((bw)->words) + (bw)->bits)
+
+#ifdef min
+#undef min
+#endif
+#define min(x,y) ((x)<(y)?(x):(y))
+
+/* adjust for compilers that can't understand using LLU suffix for uint64_t literals */
+#ifdef _MSC_VER
+#define FLAC__U64L(x) x
+#else
+#define FLAC__U64L(x) x##LLU
+#endif
+
+#ifndef FLaC__INLINE
+#define FLaC__INLINE
+#endif
+
+struct FLAC__BitWriter {
+	bwword *buffer;
+	bwword accum; /* accumulator; bits are right-justified; when full, accum is appended to buffer */
+	unsigned capacity; /* capacity of buffer in words */
+	unsigned words; /* # of complete words in buffer */
+	unsigned bits; /* # of used bits in accum */
+};
+
+/* * WATCHOUT: The current implementation only grows the buffer. */
+static FLAC__bool bitwriter_grow_(FLAC__BitWriter *bw, unsigned bits_to_add)
+{
+	unsigned new_capacity;
+	bwword *new_buffer;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+
+	/* calculate total words needed to store 'bits_to_add' additional bits */
+	new_capacity = bw->words + ((bw->bits + bits_to_add + FLAC__BITS_PER_WORD - 1) / FLAC__BITS_PER_WORD);
+
+	/* it's possible (due to pessimism in the growth estimation that
+	 * leads to this call) that we don't actually need to grow
+	 */
+	if(bw->capacity >= new_capacity)
+		return true;
+
+	/* round up capacity increase to the nearest FLAC__BITWRITER_DEFAULT_INCREMENT */
+	if((new_capacity - bw->capacity) % FLAC__BITWRITER_DEFAULT_INCREMENT)
+		new_capacity += FLAC__BITWRITER_DEFAULT_INCREMENT - ((new_capacity - bw->capacity) % FLAC__BITWRITER_DEFAULT_INCREMENT);
+	/* make sure we got everything right */
+	FLAC__ASSERT(0 == (new_capacity - bw->capacity) % FLAC__BITWRITER_DEFAULT_INCREMENT);
+	FLAC__ASSERT(new_capacity > bw->capacity);
+	FLAC__ASSERT(new_capacity >= bw->words + ((bw->bits + bits_to_add + FLAC__BITS_PER_WORD - 1) / FLAC__BITS_PER_WORD));
+
+	new_buffer = (bwword*)safe_realloc_mul_2op_(bw->buffer, sizeof(bwword), /*times*/new_capacity);
+	if(new_buffer == 0)
+		return false;
+	bw->buffer = new_buffer;
+	bw->capacity = new_capacity;
+	return true;
+}
+
+
+/***********************************************************************
+ *
+ * Class constructor/destructor
+ *
+ ***********************************************************************/
+
+FLAC__BitWriter *FLAC__bitwriter_new(void)
+{
+	FLAC__BitWriter *bw = (FLAC__BitWriter*)calloc(1, sizeof(FLAC__BitWriter));
+	/* note that calloc() sets all members to 0 for us */
+	return bw;
+}
+
+void FLAC__bitwriter_delete(FLAC__BitWriter *bw)
+{
+	FLAC__ASSERT(0 != bw);
+
+	FLAC__bitwriter_free(bw);
+	free(bw);
+}
+
+/***********************************************************************
+ *
+ * Public class methods
+ *
+ ***********************************************************************/
+
+FLAC__bool FLAC__bitwriter_init(FLAC__BitWriter *bw)
+{
+	FLAC__ASSERT(0 != bw);
+
+	bw->words = bw->bits = 0;
+	bw->capacity = FLAC__BITWRITER_DEFAULT_CAPACITY;
+	bw->buffer = (bwword*)malloc(sizeof(bwword) * bw->capacity);
+	if(bw->buffer == 0)
+		return false;
+
+	return true;
+}
+
+void FLAC__bitwriter_free(FLAC__BitWriter *bw)
+{
+	FLAC__ASSERT(0 != bw);
+
+	if(0 != bw->buffer)
+		free(bw->buffer);
+	bw->buffer = 0;
+	bw->capacity = 0;
+	bw->words = bw->bits = 0;
+}
+
+void FLAC__bitwriter_clear(FLAC__BitWriter *bw)
+{
+	bw->words = bw->bits = 0;
+}
+
+void FLAC__bitwriter_dump(const FLAC__BitWriter *bw, FILE *out)
+{
+	unsigned i, j;
+	if(bw == 0) {
+		fprintf(out, "bitwriter is NULL\n");
+	}
+	else {
+		fprintf(out, "bitwriter: capacity=%u words=%u bits=%u total_bits=%u\n", bw->capacity, bw->words, bw->bits, FLAC__TOTAL_BITS(bw));
+
+		for(i = 0; i < bw->words; i++) {
+			fprintf(out, "%08X: ", i);
+			for(j = 0; j < FLAC__BITS_PER_WORD; j++)
+				fprintf(out, "%01u", bw->buffer[i] & (1 << (FLAC__BITS_PER_WORD-j-1)) ? 1:0);
+			fprintf(out, "\n");
+		}
+		if(bw->bits > 0) {
+			fprintf(out, "%08X: ", i);
+			for(j = 0; j < bw->bits; j++)
+				fprintf(out, "%01u", bw->accum & (1 << (bw->bits-j-1)) ? 1:0);
+			fprintf(out, "\n");
+		}
+	}
+}
+
+FLAC__bool FLAC__bitwriter_get_write_crc16(FLAC__BitWriter *bw, FLAC__uint16 *crc)
+{
+	const FLAC__byte *buffer;
+	size_t bytes;
+
+	FLAC__ASSERT((bw->bits & 7) == 0); /* assert that we're byte-aligned */
+
+	if(!FLAC__bitwriter_get_buffer(bw, &buffer, &bytes))
+		return false;
+
+	*crc = (FLAC__uint16)FLAC__crc16(buffer, bytes);
+	FLAC__bitwriter_release_buffer(bw);
+	return true;
+}
+
+FLAC__bool FLAC__bitwriter_get_write_crc8(FLAC__BitWriter *bw, FLAC__byte *crc)
+{
+	const FLAC__byte *buffer;
+	size_t bytes;
+
+	FLAC__ASSERT((bw->bits & 7) == 0); /* assert that we're byte-aligned */
+
+	if(!FLAC__bitwriter_get_buffer(bw, &buffer, &bytes))
+		return false;
+
+	*crc = FLAC__crc8(buffer, bytes);
+	FLAC__bitwriter_release_buffer(bw);
+	return true;
+}
+
+FLAC__bool FLAC__bitwriter_is_byte_aligned(const FLAC__BitWriter *bw)
+{
+	return ((bw->bits & 7) == 0);
+}
+
+unsigned FLAC__bitwriter_get_input_bits_unconsumed(const FLAC__BitWriter *bw)
+{
+	return FLAC__TOTAL_BITS(bw);
+}
+
+FLAC__bool FLAC__bitwriter_get_buffer(FLAC__BitWriter *bw, const FLAC__byte **buffer, size_t *bytes)
+{
+	FLAC__ASSERT((bw->bits & 7) == 0);
+	/* double protection */
+	if(bw->bits & 7)
+		return false;
+	/* if we have bits in the accumulator we have to flush those to the buffer first */
+	if(bw->bits) {
+		FLAC__ASSERT(bw->words <= bw->capacity);
+		if(bw->words == bw->capacity && !bitwriter_grow_(bw, FLAC__BITS_PER_WORD))
+			return false;
+		/* append bits as complete word to buffer, but don't change bw->accum or bw->bits */
+		bw->buffer[bw->words] = SWAP_BE_WORD_TO_HOST(bw->accum << (FLAC__BITS_PER_WORD-bw->bits));
+	}
+	/* now we can just return what we have */
+	*buffer = (FLAC__byte*)bw->buffer;
+	*bytes = (FLAC__BYTES_PER_WORD * bw->words) + (bw->bits >> 3);
+	return true;
+}
+
+void FLAC__bitwriter_release_buffer(FLAC__BitWriter *bw)
+{
+	/* nothing to do.  in the future, strict checking of a 'writer-is-in-
+	 * get-mode' flag could be added everywhere and then cleared here
+	 */
+	(void)bw;
+}
+
+FLaC__INLINE FLAC__bool FLAC__bitwriter_write_zeroes(FLAC__BitWriter *bw, unsigned bits)
+{
+	unsigned n;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+
+	if(bits == 0)
+		return true;
+	/* slightly pessimistic size check but faster than "<= bw->words + (bw->bits+bits+FLAC__BITS_PER_WORD-1)/FLAC__BITS_PER_WORD" */
+	if(bw->capacity <= bw->words + bits && !bitwriter_grow_(bw, bits))
+		return false;
+	/* first part gets to word alignment */
+	if(bw->bits) {
+		n = min(FLAC__BITS_PER_WORD - bw->bits, bits);
+		bw->accum <<= n;
+		bits -= n;
+		bw->bits += n;
+		if(bw->bits == FLAC__BITS_PER_WORD) {
+			bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
+			bw->bits = 0;
+		}
+		else
+			return true;
+	}
+	/* do whole words */
+	while(bits >= FLAC__BITS_PER_WORD) {
+		bw->buffer[bw->words++] = 0;
+		bits -= FLAC__BITS_PER_WORD;
+	}
+	/* do any leftovers */
+	if(bits > 0) {
+		bw->accum = 0;
+		bw->bits = bits;
+	}
+	return true;
+}
+
+FLaC__INLINE FLAC__bool FLAC__bitwriter_write_raw_uint32(FLAC__BitWriter *bw, FLAC__uint32 val, unsigned bits)
+{
+	register unsigned left;
+
+	/* WATCHOUT: code does not work with <32bit words; we can make things much faster with this assertion */
+	FLAC__ASSERT(FLAC__BITS_PER_WORD >= 32);
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+
+	FLAC__ASSERT(bits <= 32);
+	if(bits == 0)
+		return true;
+
+	/* slightly pessimistic size check but faster than "<= bw->words + (bw->bits+bits+FLAC__BITS_PER_WORD-1)/FLAC__BITS_PER_WORD" */
+	if(bw->capacity <= bw->words + bits && !bitwriter_grow_(bw, bits))
+		return false;
+
+	left = FLAC__BITS_PER_WORD - bw->bits;
+	if(bits < left) {
+		bw->accum <<= bits;
+		bw->accum |= val;
+		bw->bits += bits;
+	}
+	else if(bw->bits) { /* WATCHOUT: if bw->bits == 0, left==FLAC__BITS_PER_WORD and bw->accum<<=left is a NOP instead of setting to 0 */
+		bw->accum <<= left;
+		bw->accum |= val >> (bw->bits = bits - left);
+		bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
+		bw->accum = val;
+	}
+	else {
+		bw->accum = val;
+		bw->bits = 0;
+		bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(val);
+	}
+
+	return true;
+}
+
+FLaC__INLINE FLAC__bool FLAC__bitwriter_write_raw_int32(FLAC__BitWriter *bw, FLAC__int32 val, unsigned bits)
+{
+	/* zero-out unused bits */
+	if(bits < 32)
+		val &= (~(0xffffffff << bits));
+
+	return FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)val, bits);
+}
+
+FLaC__INLINE FLAC__bool FLAC__bitwriter_write_raw_uint64(FLAC__BitWriter *bw, FLAC__uint64 val, unsigned bits)
+{
+	/* this could be a little faster but it's not used for much */
+	if(bits > 32) {
+		return
+			FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)(val>>32), bits-32) &&
+			FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)val, 32);
+	}
+	else
+		return FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)val, bits);
+}
+
+FLaC__INLINE FLAC__bool FLAC__bitwriter_write_raw_uint32_little_endian(FLAC__BitWriter *bw, FLAC__uint32 val)
+{
+	/* this doesn't need to be that fast as currently it is only used for vorbis comments */
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, val & 0xff, 8))
+		return false;
+	if(!FLAC__bitwriter_write_raw_uint32(bw, (val>>8) & 0xff, 8))
+		return false;
+	if(!FLAC__bitwriter_write_raw_uint32(bw, (val>>16) & 0xff, 8))
+		return false;
+	if(!FLAC__bitwriter_write_raw_uint32(bw, val>>24, 8))
+		return false;
+
+	return true;
+}
+
+FLaC__INLINE FLAC__bool FLAC__bitwriter_write_byte_block(FLAC__BitWriter *bw, const FLAC__byte vals[], unsigned nvals)
+{
+	unsigned i;
+
+	/* this could be faster but currently we don't need it to be since it's only used for writing metadata */
+	for(i = 0; i < nvals; i++) {
+		if(!FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)(vals[i]), 8))
+			return false;
+	}
+
+	return true;
+}
+
+FLAC__bool FLAC__bitwriter_write_unary_unsigned(FLAC__BitWriter *bw, unsigned val)
+{
+	if(val < 32)
+		return FLAC__bitwriter_write_raw_uint32(bw, 1, ++val);
+	else
+		return
+			FLAC__bitwriter_write_zeroes(bw, val) &&
+			FLAC__bitwriter_write_raw_uint32(bw, 1, 1);
+}
+
+unsigned FLAC__bitwriter_rice_bits(FLAC__int32 val, unsigned parameter)
+{
+	FLAC__uint32 uval;
+
+	FLAC__ASSERT(parameter < sizeof(unsigned)*8);
+
+	/* fold signed to unsigned; actual formula is: negative(v)? -2v-1 : 2v */
+	uval = (val<<1) ^ (val>>31);
+
+	return 1 + parameter + (uval >> parameter);
+}
+
+#if 0 /* UNUSED */
+unsigned FLAC__bitwriter_golomb_bits_signed(int val, unsigned parameter)
+{
+	unsigned bits, msbs, uval;
+	unsigned k;
+
+	FLAC__ASSERT(parameter > 0);
+
+	/* fold signed to unsigned */
+	if(val < 0)
+		uval = (unsigned)(((-(++val)) << 1) + 1);
+	else
+		uval = (unsigned)(val << 1);
+
+	k = FLAC__bitmath_ilog2(parameter);
+	if(parameter == 1u<<k) {
+		FLAC__ASSERT(k <= 30);
+
+		msbs = uval >> k;
+		bits = 1 + k + msbs;
+	}
+	else {
+		unsigned q, r, d;
+
+		d = (1 << (k+1)) - parameter;
+		q = uval / parameter;
+		r = uval - (q * parameter);
+
+		bits = 1 + q + k;
+		if(r >= d)
+			bits++;
+	}
+	return bits;
+}
+
+unsigned FLAC__bitwriter_golomb_bits_unsigned(unsigned uval, unsigned parameter)
+{
+	unsigned bits, msbs;
+	unsigned k;
+
+	FLAC__ASSERT(parameter > 0);
+
+	k = FLAC__bitmath_ilog2(parameter);
+	if(parameter == 1u<<k) {
+		FLAC__ASSERT(k <= 30);
+
+		msbs = uval >> k;
+		bits = 1 + k + msbs;
+	}
+	else {
+		unsigned q, r, d;
+
+		d = (1 << (k+1)) - parameter;
+		q = uval / parameter;
+		r = uval - (q * parameter);
+
+		bits = 1 + q + k;
+		if(r >= d)
+			bits++;
+	}
+	return bits;
+}
+#endif /* UNUSED */
+
+FLAC__bool FLAC__bitwriter_write_rice_signed(FLAC__BitWriter *bw, FLAC__int32 val, unsigned parameter)
+{
+	unsigned total_bits, interesting_bits, msbs;
+	FLAC__uint32 uval, pattern;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+	FLAC__ASSERT(parameter < 8*sizeof(uval));
+
+	/* fold signed to unsigned; actual formula is: negative(v)? -2v-1 : 2v */
+	uval = (val<<1) ^ (val>>31);
+
+	msbs = uval >> parameter;
+	interesting_bits = 1 + parameter;
+	total_bits = interesting_bits + msbs;
+	pattern = 1 << parameter; /* the unary end bit */
+	pattern |= (uval & ((1<<parameter)-1)); /* the binary LSBs */
+
+	if(total_bits <= 32)
+		return FLAC__bitwriter_write_raw_uint32(bw, pattern, total_bits);
+	else
+		return
+			FLAC__bitwriter_write_zeroes(bw, msbs) && /* write the unary MSBs */
+			FLAC__bitwriter_write_raw_uint32(bw, pattern, interesting_bits); /* write the unary end bit and binary LSBs */
+}
+
+FLAC__bool FLAC__bitwriter_write_rice_signed_block(FLAC__BitWriter *bw, const FLAC__int32 *vals, unsigned nvals, unsigned parameter)
+{
+	const FLAC__uint32 mask1 = FLAC__WORD_ALL_ONES << parameter; /* we val|=mask1 to set the stop bit above it... */
+	const FLAC__uint32 mask2 = FLAC__WORD_ALL_ONES >> (31-parameter); /* ...then mask off the bits above the stop bit with val&=mask2*/
+	FLAC__uint32 uval;
+	unsigned left;
+	const unsigned lsbits = 1 + parameter;
+	unsigned msbits;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+	FLAC__ASSERT(parameter < 8*sizeof(bwword)-1);
+	/* WATCHOUT: code does not work with <32bit words; we can make things much faster with this assertion */
+	FLAC__ASSERT(FLAC__BITS_PER_WORD >= 32);
+
+	while(nvals) {
+		/* fold signed to unsigned; actual formula is: negative(v)? -2v-1 : 2v */
+		uval = (*vals<<1) ^ (*vals>>31);
+
+		msbits = uval >> parameter;
+
+#if 0 /* OPT: can remove this special case if it doesn't make up for the extra compare (doesn't make a statistically significant difference with msvc or gcc/x86) */
+		if(bw->bits && bw->bits + msbits + lsbits <= FLAC__BITS_PER_WORD) { /* i.e. if the whole thing fits in the current bwword */
+			/* ^^^ if bw->bits is 0 then we may have filled the buffer and have no free bwword to work in */
+			bw->bits = bw->bits + msbits + lsbits;
+			uval |= mask1; /* set stop bit */
+			uval &= mask2; /* mask off unused top bits */
+			/* NOT: bw->accum <<= msbits + lsbits because msbits+lsbits could be 32, then the shift would be a NOP */
+			bw->accum <<= msbits;
+			bw->accum <<= lsbits;
+			bw->accum |= uval;
+			if(bw->bits == FLAC__BITS_PER_WORD) {
+				bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
+				bw->bits = 0;
+				/* burying the capacity check down here means we have to grow the buffer a little if there are more vals to do */
+				if(bw->capacity <= bw->words && nvals > 1 && !bitwriter_grow_(bw, 1)) {
+					FLAC__ASSERT(bw->capacity == bw->words);
+					return false;
+				}
+			}
+		}
+		else {
+#elif 1 /*@@@@@@ OPT: try this version with MSVC6 to see if better, not much difference for gcc-4 */
+		if(bw->bits && bw->bits + msbits + lsbits < FLAC__BITS_PER_WORD) { /* i.e. if the whole thing fits in the current bwword */
+			/* ^^^ if bw->bits is 0 then we may have filled the buffer and have no free bwword to work in */
+			bw->bits = bw->bits + msbits + lsbits;
+			uval |= mask1; /* set stop bit */
+			uval &= mask2; /* mask off unused top bits */
+			bw->accum <<= msbits + lsbits;
+			bw->accum |= uval;
+		}
+		else {
+#endif
+			/* slightly pessimistic size check but faster than "<= bw->words + (bw->bits+msbits+lsbits+FLAC__BITS_PER_WORD-1)/FLAC__BITS_PER_WORD" */
+			/* OPT: pessimism may cause flurry of false calls to grow_ which eat up all savings before it */
+			if(bw->capacity <= bw->words + bw->bits + msbits + 1/*lsbits always fit in 1 bwword*/ && !bitwriter_grow_(bw, msbits+lsbits))
+				return false;
+
+			if(msbits) {
+				/* first part gets to word alignment */
+				if(bw->bits) {
+					left = FLAC__BITS_PER_WORD - bw->bits;
+					if(msbits < left) {
+						bw->accum <<= msbits;
+						bw->bits += msbits;
+						goto break1;
+					}
+					else {
+						bw->accum <<= left;
+						msbits -= left;
+						bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
+						bw->bits = 0;
+					}
+				}
+				/* do whole words */
+				while(msbits >= FLAC__BITS_PER_WORD) {
+					bw->buffer[bw->words++] = 0;
+					msbits -= FLAC__BITS_PER_WORD;
+				}
+				/* do any leftovers */
+				if(msbits > 0) {
+					bw->accum = 0;
+					bw->bits = msbits;
+				}
+			}
+break1:
+			uval |= mask1; /* set stop bit */
+			uval &= mask2; /* mask off unused top bits */
+
+			left = FLAC__BITS_PER_WORD - bw->bits;
+			if(lsbits < left) {
+				bw->accum <<= lsbits;
+				bw->accum |= uval;
+				bw->bits += lsbits;
+			}
+			else {
+				/* if bw->bits == 0, left==FLAC__BITS_PER_WORD which will always
+				 * be > lsbits (because of previous assertions) so it would have
+				 * triggered the (lsbits<left) case above.
+				 */
+				FLAC__ASSERT(bw->bits);
+				FLAC__ASSERT(left < FLAC__BITS_PER_WORD);
+				bw->accum <<= left;
+				bw->accum |= uval >> (bw->bits = lsbits - left);
+				bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
+				bw->accum = uval;
+			}
+#if 1
+		}
+#endif
+		vals++;
+		nvals--;
+	}
+	return true;
+}
+
+#if 0 /* UNUSED */
+FLAC__bool FLAC__bitwriter_write_golomb_signed(FLAC__BitWriter *bw, int val, unsigned parameter)
+{
+	unsigned total_bits, msbs, uval;
+	unsigned k;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+	FLAC__ASSERT(parameter > 0);
+
+	/* fold signed to unsigned */
+	if(val < 0)
+		uval = (unsigned)(((-(++val)) << 1) + 1);
+	else
+		uval = (unsigned)(val << 1);
+
+	k = FLAC__bitmath_ilog2(parameter);
+	if(parameter == 1u<<k) {
+		unsigned pattern;
+
+		FLAC__ASSERT(k <= 30);
+
+		msbs = uval >> k;
+		total_bits = 1 + k + msbs;
+		pattern = 1 << k; /* the unary end bit */
+		pattern |= (uval & ((1u<<k)-1)); /* the binary LSBs */
+
+		if(total_bits <= 32) {
+			if(!FLAC__bitwriter_write_raw_uint32(bw, pattern, total_bits))
+				return false;
+		}
+		else {
+			/* write the unary MSBs */
+			if(!FLAC__bitwriter_write_zeroes(bw, msbs))
+				return false;
+			/* write the unary end bit and binary LSBs */
+			if(!FLAC__bitwriter_write_raw_uint32(bw, pattern, k+1))
+				return false;
+		}
+	}
+	else {
+		unsigned q, r, d;
+
+		d = (1 << (k+1)) - parameter;
+		q = uval / parameter;
+		r = uval - (q * parameter);
+		/* write the unary MSBs */
+		if(!FLAC__bitwriter_write_zeroes(bw, q))
+			return false;
+		/* write the unary end bit */
+		if(!FLAC__bitwriter_write_raw_uint32(bw, 1, 1))
+			return false;
+		/* write the binary LSBs */
+		if(r >= d) {
+			if(!FLAC__bitwriter_write_raw_uint32(bw, r+d, k+1))
+				return false;
+		}
+		else {
+			if(!FLAC__bitwriter_write_raw_uint32(bw, r, k))
+				return false;
+		}
+	}
+	return true;
+}
+
+FLAC__bool FLAC__bitwriter_write_golomb_unsigned(FLAC__BitWriter *bw, unsigned uval, unsigned parameter)
+{
+	unsigned total_bits, msbs;
+	unsigned k;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+	FLAC__ASSERT(parameter > 0);
+
+	k = FLAC__bitmath_ilog2(parameter);
+	if(parameter == 1u<<k) {
+		unsigned pattern;
+
+		FLAC__ASSERT(k <= 30);
+
+		msbs = uval >> k;
+		total_bits = 1 + k + msbs;
+		pattern = 1 << k; /* the unary end bit */
+		pattern |= (uval & ((1u<<k)-1)); /* the binary LSBs */
+
+		if(total_bits <= 32) {
+			if(!FLAC__bitwriter_write_raw_uint32(bw, pattern, total_bits))
+				return false;
+		}
+		else {
+			/* write the unary MSBs */
+			if(!FLAC__bitwriter_write_zeroes(bw, msbs))
+				return false;
+			/* write the unary end bit and binary LSBs */
+			if(!FLAC__bitwriter_write_raw_uint32(bw, pattern, k+1))
+				return false;
+		}
+	}
+	else {
+		unsigned q, r, d;
+
+		d = (1 << (k+1)) - parameter;
+		q = uval / parameter;
+		r = uval - (q * parameter);
+		/* write the unary MSBs */
+		if(!FLAC__bitwriter_write_zeroes(bw, q))
+			return false;
+		/* write the unary end bit */
+		if(!FLAC__bitwriter_write_raw_uint32(bw, 1, 1))
+			return false;
+		/* write the binary LSBs */
+		if(r >= d) {
+			if(!FLAC__bitwriter_write_raw_uint32(bw, r+d, k+1))
+				return false;
+		}
+		else {
+			if(!FLAC__bitwriter_write_raw_uint32(bw, r, k))
+				return false;
+		}
+	}
+	return true;
+}
+#endif /* UNUSED */
+
+FLAC__bool FLAC__bitwriter_write_utf8_uint32(FLAC__BitWriter *bw, FLAC__uint32 val)
+{
+	FLAC__bool ok = 1;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+
+	FLAC__ASSERT(!(val & 0x80000000)); /* this version only handles 31 bits */
+
+	if(val < 0x80) {
+		return FLAC__bitwriter_write_raw_uint32(bw, val, 8);
+	}
+	else if(val < 0x800) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xC0 | (val>>6), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
+	}
+	else if(val < 0x10000) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xE0 | (val>>12), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
+	}
+	else if(val < 0x200000) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xF0 | (val>>18), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>12)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
+	}
+	else if(val < 0x4000000) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xF8 | (val>>24), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>18)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>12)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
+	}
+	else {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xFC | (val>>30), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>24)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>18)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>12)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
+	}
+
+	return ok;
+}
+
+FLAC__bool FLAC__bitwriter_write_utf8_uint64(FLAC__BitWriter *bw, FLAC__uint64 val)
+{
+	FLAC__bool ok = 1;
+
+	FLAC__ASSERT(0 != bw);
+	FLAC__ASSERT(0 != bw->buffer);
+
+	FLAC__ASSERT(!(val & FLAC__U64L(0xFFFFFFF000000000))); /* this version only handles 36 bits */
+
+	if(val < 0x80) {
+		return FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)val, 8);
+	}
+	else if(val < 0x800) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xC0 | (FLAC__uint32)(val>>6), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
+	}
+	else if(val < 0x10000) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xE0 | (FLAC__uint32)(val>>12), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
+	}
+	else if(val < 0x200000) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xF0 | (FLAC__uint32)(val>>18), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>12)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
+	}
+	else if(val < 0x4000000) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xF8 | (FLAC__uint32)(val>>24), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>18)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>12)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
+	}
+	else if(val < 0x80000000) {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xFC | (FLAC__uint32)(val>>30), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>24)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>18)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>12)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
+	}
+	else {
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xFE, 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>30)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>24)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>18)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>12)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
+		ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
+	}
+
+	return ok;
+}
+
+FLAC__bool FLAC__bitwriter_zero_pad_to_byte_boundary(FLAC__BitWriter *bw)
+{
+	/* 0-pad to byte boundary */
+	if(bw->bits & 7u)
+		return FLAC__bitwriter_write_zeroes(bw, 8 - (bw->bits & 7u));
+	else
+		return true;
+}
+
+#endif

src/audio/audio_file_formats/flac/libFLAC/cpu.c

@@ -0,0 +1,423 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "juce_FlacHeader.h"
+#if JUCE_USE_FLAC
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include "include/private/cpu.h"
+#include <stdlib.h>
+#include <stdio.h>
+
+#if defined FLAC__CPU_IA32
+# include <signal.h>
+#elif defined FLAC__CPU_PPC
+# if !defined FLAC__NO_ASM
+#  if defined FLAC__SYS_DARWIN
+#   include <sys/sysctl.h>
+#   include <mach/mach.h>
+#   include <mach/mach_host.h>
+#   include <mach/host_info.h>
+#   include <mach/machine.h>
+#   ifndef CPU_SUBTYPE_POWERPC_970
+#    define CPU_SUBTYPE_POWERPC_970 ((cpu_subtype_t) 100)
+#   endif
+#  else /* FLAC__SYS_DARWIN */
+
+#   include <signal.h>
+#   include <setjmp.h>
+
+static sigjmp_buf jmpbuf;
+static volatile sig_atomic_t canjump = 0;
+
+static void sigill_handler (int sig)
+{
+	if (!canjump) {
+		signal (sig, SIG_DFL);
+		raise (sig);
+	}
+	canjump = 0;
+	siglongjmp (jmpbuf, 1);
+}
+#  endif /* FLAC__SYS_DARWIN */
+# endif /* FLAC__NO_ASM */
+#endif /* FLAC__CPU_PPC */
+
+#if defined (__NetBSD__) || defined(__OpenBSD__)
+#include <sys/param.h>
+#include <sys/sysctl.h>
+#include <machine/cpu.h>
+#endif
+
+#if defined(__FreeBSD__) || defined(__FreeBSD_kernel__) || defined(__DragonFly__)
+#include <sys/types.h>
+#include <sys/sysctl.h>
+#endif
+
+#if defined(__APPLE__)
+/* how to get sysctlbyname()? */
+#endif
+
+/* these are flags in EDX of CPUID AX=00000001 */
+static const unsigned FLAC__CPUINFO_IA32_CPUID_CMOV = 0x00008000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_MMX = 0x00800000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_FXSR = 0x01000000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE = 0x02000000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE2 = 0x04000000;
+/* these are flags in ECX of CPUID AX=00000001 */
+static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE3 = 0x00000001;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_SSSE3 = 0x00000200;
+/* these are flags in EDX of CPUID AX=80000001 */
+static const unsigned FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_3DNOW = 0x80000000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_EXT3DNOW = 0x40000000;
+static const unsigned FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_EXTMMX = 0x00400000;
+
+
+/*
+ * Extra stuff needed for detection of OS support for SSE on IA-32
+ */
+#if defined(FLAC__CPU_IA32) && !defined FLAC__NO_ASM && defined FLAC__HAS_NASM && !defined FLAC__NO_SSE_OS && !defined FLAC__SSE_OS
+# if defined(__linux__)
+/*
+ * If the OS doesn't support SSE, we will get here with a SIGILL.  We
+ * modify the return address to jump over the offending SSE instruction
+ * and also the operation following it that indicates the instruction
+ * executed successfully.  In this way we use no global variables and
+ * stay thread-safe.
+ *
+ * 3 + 3 + 6:
+ *   3 bytes for "xorps xmm0,xmm0"
+ *   3 bytes for estimate of how long the follwing "inc var" instruction is
+ *   6 bytes extra in case our estimate is wrong
+ * 12 bytes puts us in the NOP "landing zone"
+ */
+#  undef USE_OBSOLETE_SIGCONTEXT_FLAVOR /* #define this to use the older signal handler method */
+#  ifdef USE_OBSOLETE_SIGCONTEXT_FLAVOR
+	static void sigill_handler_sse_os(int signal, struct sigcontext sc)
+	{
+		(void)signal;
+		sc.eip += 3 + 3 + 6;
+	}
+#  else
+#   include <sys/ucontext.h>
+	static void sigill_handler_sse_os(int signal, siginfo_t *si, void *uc)
+	{
+		(void)signal, (void)si;
+		((ucontext_t*)uc)->uc_mcontext.gregs[14/*REG_EIP*/] += 3 + 3 + 6;
+	}
+#  endif
+# elif defined(_MSC_VER)
+#  include <windows.h>
+#  undef USE_TRY_CATCH_FLAVOR /* #define this to use the try/catch method for catching illegal opcode exception */
+#  ifdef USE_TRY_CATCH_FLAVOR
+#  else
+	LONG CALLBACK sigill_handler_sse_os(EXCEPTION_POINTERS *ep)
+	{
+		if(ep->ExceptionRecord->ExceptionCode == EXCEPTION_ILLEGAL_INSTRUCTION) {
+			ep->ContextRecord->Eip += 3 + 3 + 6;
+			return EXCEPTION_CONTINUE_EXECUTION;
+		}
+		return EXCEPTION_CONTINUE_SEARCH;
+	}
+#  endif
+# endif
+#endif
+
+
+void FLAC__cpu_info(FLAC__CPUInfo *info)
+{
+/*
+ * IA32-specific
+ */
+#ifdef FLAC__CPU_IA32
+	info->type = FLAC__CPUINFO_TYPE_IA32;
+#if !defined FLAC__NO_ASM && defined FLAC__HAS_NASM
+	info->use_asm = true; /* we assume a minimum of 80386 with FLAC__CPU_IA32 */
+	info->data.ia32.cpuid = FLAC__cpu_have_cpuid_asm_ia32()? true : false;
+	info->data.ia32.bswap = info->data.ia32.cpuid; /* CPUID => BSWAP since it came after */
+	info->data.ia32.cmov = false;
+	info->data.ia32.mmx = false;
+	info->data.ia32.fxsr = false;
+	info->data.ia32.sse = false;
+	info->data.ia32.sse2 = false;
+	info->data.ia32.sse3 = false;
+	info->data.ia32.ssse3 = false;
+	info->data.ia32._3dnow = false;
+	info->data.ia32.ext3dnow = false;
+	info->data.ia32.extmmx = false;
+	if(info->data.ia32.cpuid) {
+		/* http://www.sandpile.org/ia32/cpuid.htm */
+		FLAC__uint32 flags_edx, flags_ecx;
+		FLAC__cpu_info_asm_ia32(&flags_edx, &flags_ecx);
+		info->data.ia32.cmov  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_CMOV )? true : false;
+		info->data.ia32.mmx   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_MMX  )? true : false;
+		info->data.ia32.fxsr  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_FXSR )? true : false;
+		info->data.ia32.sse   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_SSE  )? true : false;
+		info->data.ia32.sse2  = (flags_edx & FLAC__CPUINFO_IA32_CPUID_SSE2 )? true : false;
+		info->data.ia32.sse3  = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE3 )? true : false;
+		info->data.ia32.ssse3 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSSE3)? true : false;
+
+#ifdef FLAC__USE_3DNOW
+		flags_edx = FLAC__cpu_info_extended_amd_asm_ia32();
+		info->data.ia32._3dnow   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_3DNOW   )? true : false;
+		info->data.ia32.ext3dnow = (flags_edx & FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_EXT3DNOW)? true : false;
+		info->data.ia32.extmmx   = (flags_edx & FLAC__CPUINFO_IA32_CPUID_EXTENDED_AMD_EXTMMX  )? true : false;
+#else
+		info->data.ia32._3dnow = info->data.ia32.ext3dnow = info->data.ia32.extmmx = false;
+#endif
+
+#ifdef DEBUG
+		fprintf(stderr, "CPU info (IA-32):\n");
+		fprintf(stderr, "  CPUID ...... %c\n", info->data.ia32.cpuid   ? 'Y' : 'n');
+		fprintf(stderr, "  BSWAP ...... %c\n", info->data.ia32.bswap   ? 'Y' : 'n');
+		fprintf(stderr, "  CMOV ....... %c\n", info->data.ia32.cmov    ? 'Y' : 'n');
+		fprintf(stderr, "  MMX ........ %c\n", info->data.ia32.mmx     ? 'Y' : 'n');
+		fprintf(stderr, "  FXSR ....... %c\n", info->data.ia32.fxsr    ? 'Y' : 'n');
+		fprintf(stderr, "  SSE ........ %c\n", info->data.ia32.sse     ? 'Y' : 'n');
+		fprintf(stderr, "  SSE2 ....... %c\n", info->data.ia32.sse2    ? 'Y' : 'n');
+		fprintf(stderr, "  SSE3 ....... %c\n", info->data.ia32.sse3    ? 'Y' : 'n');
+		fprintf(stderr, "  SSSE3 ...... %c\n", info->data.ia32.ssse3   ? 'Y' : 'n');
+		fprintf(stderr, "  3DNow! ..... %c\n", info->data.ia32._3dnow  ? 'Y' : 'n');
+		fprintf(stderr, "  3DNow!-ext . %c\n", info->data.ia32.ext3dnow? 'Y' : 'n');
+		fprintf(stderr, "  3DNow!-MMX . %c\n", info->data.ia32.extmmx  ? 'Y' : 'n');
+#endif
+
+		/*
+		 * now have to check for OS support of SSE/SSE2
+		 */
+		if(info->data.ia32.fxsr || info->data.ia32.sse || info->data.ia32.sse2) {
+#if defined FLAC__NO_SSE_OS
+			/* assume user knows better than us; turn it off */
+			info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+#elif defined FLAC__SSE_OS
+			/* assume user knows better than us; leave as detected above */
+#elif defined(__FreeBSD__) || defined(__FreeBSD_kernel__) || defined(__DragonFly__) || defined(__APPLE__)
+			int sse = 0;
+			size_t len;
+			/* at least one of these must work: */
+			len = sizeof(sse); sse = sse || (sysctlbyname("hw.instruction_sse", &sse, &len, NULL, 0) == 0 && sse);
+			len = sizeof(sse); sse = sse || (sysctlbyname("hw.optional.sse"   , &sse, &len, NULL, 0) == 0 && sse); /* __APPLE__ ? */
+			if(!sse)
+				info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+#elif defined(__NetBSD__) || defined (__OpenBSD__)
+# if __NetBSD_Version__ >= 105250000 || (defined __OpenBSD__)
+			int val = 0, mib[2] = { CTL_MACHDEP, CPU_SSE };
+			size_t len = sizeof(val);
+			if(sysctl(mib, 2, &val, &len, NULL, 0) < 0 || !val)
+				info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+			else { /* double-check SSE2 */
+				mib[1] = CPU_SSE2;
+				len = sizeof(val);
+				if(sysctl(mib, 2, &val, &len, NULL, 0) < 0 || !val)
+					info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+			}
+# else
+			info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+# endif
+#elif defined(__linux__)
+			int sse = 0;
+			struct sigaction sigill_save;
+#ifdef USE_OBSOLETE_SIGCONTEXT_FLAVOR
+			if(0 == sigaction(SIGILL, NULL, &sigill_save) && signal(SIGILL, (void (*)(int))sigill_handler_sse_os) != SIG_ERR)
+#else
+			struct sigaction sigill_sse;
+			sigill_sse.sa_sigaction = sigill_handler_sse_os;
+			__sigemptyset(&sigill_sse.sa_mask);
+			sigill_sse.sa_flags = SA_SIGINFO | SA_RESETHAND; /* SA_RESETHAND just in case our SIGILL return jump breaks, so we don't get stuck in a loop */
+			if(0 == sigaction(SIGILL, &sigill_sse, &sigill_save))
+#endif
+			{
+				/* http://www.ibiblio.org/gferg/ldp/GCC-Inline-Assembly-HOWTO.html */
+				/* see sigill_handler_sse_os() for an explanation of the following: */
+				asm volatile (
+					"xorl %0,%0\n\t"          /* for some reason, still need to do this to clear 'sse' var */
+					"xorps %%xmm0,%%xmm0\n\t" /* will cause SIGILL if unsupported by OS */
+					"incl %0\n\t"             /* SIGILL handler will jump over this */
+					/* landing zone */
+					"nop\n\t" /* SIGILL jump lands here if "inc" is 9 bytes */
+					"nop\n\t"
+					"nop\n\t"
+					"nop\n\t"
+					"nop\n\t"
+					"nop\n\t"
+					"nop\n\t" /* SIGILL jump lands here if "inc" is 3 bytes (expected) */
+					"nop\n\t"
+					"nop"     /* SIGILL jump lands here if "inc" is 1 byte */
+					: "=r"(sse)
+					: "r"(sse)
+				);
+
+				sigaction(SIGILL, &sigill_save, NULL);
+			}
+
+			if(!sse)
+				info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+#elif defined(_MSC_VER)
+# ifdef USE_TRY_CATCH_FLAVOR
+			_try {
+				__asm {
+#  if _MSC_VER <= 1200
+					/* VC6 assembler doesn't know SSE, have to emit bytecode instead */
+					_emit 0x0F
+					_emit 0x57
+					_emit 0xC0
+#  else
+					xorps xmm0,xmm0
+#  endif
+				}
+			}
+			_except(EXCEPTION_EXECUTE_HANDLER) {
+				if (_exception_code() == STATUS_ILLEGAL_INSTRUCTION)
+					info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+			}
+# else
+			int sse = 0;
+			LPTOP_LEVEL_EXCEPTION_FILTER save = SetUnhandledExceptionFilter(sigill_handler_sse_os);
+			/* see GCC version above for explanation */
+			/*  http://msdn2.microsoft.com/en-us/library/4ks26t93.aspx */
+			/*  http://www.codeproject.com/cpp/gccasm.asp */
+			/*  http://www.hick.org/~mmiller/msvc_inline_asm.html */
+			__asm {
+#  if _MSC_VER <= 1200
+				/* VC6 assembler doesn't know SSE, have to emit bytecode instead */
+				_emit 0x0F
+				_emit 0x57
+				_emit 0xC0
+#  else
+				xorps xmm0,xmm0
+#  endif
+				inc sse
+				nop
+				nop
+				nop
+				nop
+				nop
+				nop
+				nop
+				nop
+				nop
+			}
+			SetUnhandledExceptionFilter(save);
+			if(!sse)
+				info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+# endif
+#else
+			/* no way to test, disable to be safe */
+			info->data.ia32.fxsr = info->data.ia32.sse = info->data.ia32.sse2 = info->data.ia32.sse3 = info->data.ia32.ssse3 = false;
+#endif
+#ifdef DEBUG
+		fprintf(stderr, "  SSE OS sup . %c\n", info->data.ia32.sse     ? 'Y' : 'n');
+#endif
+
+		}
+	}
+#else
+	info->use_asm = false;
+#endif
+
+/*
+ * PPC-specific
+ */
+#elif defined FLAC__CPU_PPC
+	info->type = FLAC__CPUINFO_TYPE_PPC;
+# if !defined FLAC__NO_ASM
+	info->use_asm = true;
+#  ifdef FLAC__USE_ALTIVEC
+#   if defined FLAC__SYS_DARWIN
+	{
+		int val = 0, mib[2] = { CTL_HW, HW_VECTORUNIT };
+		size_t len = sizeof(val);
+		info->data.ppc.altivec = !(sysctl(mib, 2, &val, &len, NULL, 0) || !val);
+	}
+	{
+		host_basic_info_data_t hostInfo;
+		mach_msg_type_number_t infoCount;
+
+		infoCount = HOST_BASIC_INFO_COUNT;
+		host_info(mach_host_self(), HOST_BASIC_INFO, (host_info_t)&hostInfo, &infoCount);
+
+		info->data.ppc.ppc64 = (hostInfo.cpu_type == CPU_TYPE_POWERPC) && (hostInfo.cpu_subtype == CPU_SUBTYPE_POWERPC_970);
+	}
+#   else /* FLAC__USE_ALTIVEC && !FLAC__SYS_DARWIN */
+	{
+		/* no Darwin, do it the brute-force way */
+		/* @@@@@@ this is not thread-safe; replace with SSE OS method above or remove */
+		info->data.ppc.altivec = 0;
+		info->data.ppc.ppc64 = 0;
+
+		signal (SIGILL, sigill_handler);
+		canjump = 0;
+		if (!sigsetjmp (jmpbuf, 1)) {
+			canjump = 1;
+
+			asm volatile (
+				"mtspr 256, %0\n\t"
+				"vand %%v0, %%v0, %%v0"
+				:
+				: "r" (-1)
+			);
+
+			info->data.ppc.altivec = 1;
+		}
+		canjump = 0;
+		if (!sigsetjmp (jmpbuf, 1)) {
+			int x = 0;
+			canjump = 1;
+
+			/* PPC64 hardware implements the cntlzd instruction */
+			asm volatile ("cntlzd %0, %1" : "=r" (x) : "r" (x) );
+
+			info->data.ppc.ppc64 = 1;
+		}
+		signal (SIGILL, SIG_DFL); /*@@@@@@ should save and restore old signal */
+	}
+#   endif
+#  else /* !FLAC__USE_ALTIVEC */
+	info->data.ppc.altivec = 0;
+	info->data.ppc.ppc64 = 0;
+#  endif
+# else
+	info->use_asm = false;
+# endif
+
+/*
+ * unknown CPI
+ */
+#else
+	info->type = FLAC__CPUINFO_TYPE_UNKNOWN;
+	info->use_asm = false;
+#endif
+}
+
+#endif

src/audio/audio_file_formats/flac/libFLAC/crc.c

@@ -0,0 +1,148 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "juce_FlacHeader.h"
+#if JUCE_USE_FLAC
+
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include "include/private/crc.h"
+
+/* CRC-8, poly = x^8 + x^2 + x^1 + x^0, init = 0 */
+
+FLAC__byte const FLAC__crc8_table[256] = {
+	0x00, 0x07, 0x0E, 0x09, 0x1C, 0x1B, 0x12, 0x15,
+	0x38, 0x3F, 0x36, 0x31, 0x24, 0x23, 0x2A, 0x2D,
+	0x70, 0x77, 0x7E, 0x79, 0x6C, 0x6B, 0x62, 0x65,
+	0x48, 0x4F, 0x46, 0x41, 0x54, 0x53, 0x5A, 0x5D,
+	0xE0, 0xE7, 0xEE, 0xE9, 0xFC, 0xFB, 0xF2, 0xF5,
+	0xD8, 0xDF, 0xD6, 0xD1, 0xC4, 0xC3, 0xCA, 0xCD,
+	0x90, 0x97, 0x9E, 0x99, 0x8C, 0x8B, 0x82, 0x85,
+	0xA8, 0xAF, 0xA6, 0xA1, 0xB4, 0xB3, 0xBA, 0xBD,
+	0xC7, 0xC0, 0xC9, 0xCE, 0xDB, 0xDC, 0xD5, 0xD2,
+	0xFF, 0xF8, 0xF1, 0xF6, 0xE3, 0xE4, 0xED, 0xEA,
+	0xB7, 0xB0, 0xB9, 0xBE, 0xAB, 0xAC, 0xA5, 0xA2,
+	0x8F, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9D, 0x9A,
+	0x27, 0x20, 0x29, 0x2E, 0x3B, 0x3C, 0x35, 0x32,
+	0x1F, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0D, 0x0A,
+	0x57, 0x50, 0x59, 0x5E, 0x4B, 0x4C, 0x45, 0x42,
+	0x6F, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7D, 0x7A,
+	0x89, 0x8E, 0x87, 0x80, 0x95, 0x92, 0x9B, 0x9C,
+	0xB1, 0xB6, 0xBF, 0xB8, 0xAD, 0xAA, 0xA3, 0xA4,
+	0xF9, 0xFE, 0xF7, 0xF0, 0xE5, 0xE2, 0xEB, 0xEC,
+	0xC1, 0xC6, 0xCF, 0xC8, 0xDD, 0xDA, 0xD3, 0xD4,
+	0x69, 0x6E, 0x67, 0x60, 0x75, 0x72, 0x7B, 0x7C,
+	0x51, 0x56, 0x5F, 0x58, 0x4D, 0x4A, 0x43, 0x44,
+	0x19, 0x1E, 0x17, 0x10, 0x05, 0x02, 0x0B, 0x0C,
+	0x21, 0x26, 0x2F, 0x28, 0x3D, 0x3A, 0x33, 0x34,
+	0x4E, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5C, 0x5B,
+	0x76, 0x71, 0x78, 0x7F, 0x6A, 0x6D, 0x64, 0x63,
+	0x3E, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2C, 0x2B,
+	0x06, 0x01, 0x08, 0x0F, 0x1A, 0x1D, 0x14, 0x13,
+	0xAE, 0xA9, 0xA0, 0xA7, 0xB2, 0xB5, 0xBC, 0xBB,
+	0x96, 0x91, 0x98, 0x9F, 0x8A, 0x8D, 0x84, 0x83,
+	0xDE, 0xD9, 0xD0, 0xD7, 0xC2, 0xC5, 0xCC, 0xCB,
+	0xE6, 0xE1, 0xE8, 0xEF, 0xFA, 0xFD, 0xF4, 0xF3
+};
+
+/* CRC-16, poly = x^16 + x^15 + x^2 + x^0, init = 0 */
+
+unsigned FLAC__crc16_table[256] = {
+	0x0000,  0x8005,  0x800f,  0x000a,  0x801b,  0x001e,  0x0014,  0x8011,
+	0x8033,  0x0036,  0x003c,  0x8039,  0x0028,  0x802d,  0x8027,  0x0022,
+	0x8063,  0x0066,  0x006c,  0x8069,  0x0078,  0x807d,  0x8077,  0x0072,
+	0x0050,  0x8055,  0x805f,  0x005a,  0x804b,  0x004e,  0x0044,  0x8041,
+	0x80c3,  0x00c6,  0x00cc,  0x80c9,  0x00d8,  0x80dd,  0x80d7,  0x00d2,
+	0x00f0,  0x80f5,  0x80ff,  0x00fa,  0x80eb,  0x00ee,  0x00e4,  0x80e1,
+	0x00a0,  0x80a5,  0x80af,  0x00aa,  0x80bb,  0x00be,  0x00b4,  0x80b1,
+	0x8093,  0x0096,  0x009c,  0x8099,  0x0088,  0x808d,  0x8087,  0x0082,
+	0x8183,  0x0186,  0x018c,  0x8189,  0x0198,  0x819d,  0x8197,  0x0192,
+	0x01b0,  0x81b5,  0x81bf,  0x01ba,  0x81ab,  0x01ae,  0x01a4,  0x81a1,
+	0x01e0,  0x81e5,  0x81ef,  0x01ea,  0x81fb,  0x01fe,  0x01f4,  0x81f1,
+	0x81d3,  0x01d6,  0x01dc,  0x81d9,  0x01c8,  0x81cd,  0x81c7,  0x01c2,
+	0x0140,  0x8145,  0x814f,  0x014a,  0x815b,  0x015e,  0x0154,  0x8151,
+	0x8173,  0x0176,  0x017c,  0x8179,  0x0168,  0x816d,  0x8167,  0x0162,
+	0x8123,  0x0126,  0x012c,  0x8129,  0x0138,  0x813d,  0x8137,  0x0132,
+	0x0110,  0x8115,  0x811f,  0x011a,  0x810b,  0x010e,  0x0104,  0x8101,
+	0x8303,  0x0306,  0x030c,  0x8309,  0x0318,  0x831d,  0x8317,  0x0312,
+	0x0330,  0x8335,  0x833f,  0x033a,  0x832b,  0x032e,  0x0324,  0x8321,
+	0x0360,  0x8365,  0x836f,  0x036a,  0x837b,  0x037e,  0x0374,  0x8371,
+	0x8353,  0x0356,  0x035c,  0x8359,  0x0348,  0x834d,  0x8347,  0x0342,
+	0x03c0,  0x83c5,  0x83cf,  0x03ca,  0x83db,  0x03de,  0x03d4,  0x83d1,
+	0x83f3,  0x03f6,  0x03fc,  0x83f9,  0x03e8,  0x83ed,  0x83e7,  0x03e2,
+	0x83a3,  0x03a6,  0x03ac,  0x83a9,  0x03b8,  0x83bd,  0x83b7,  0x03b2,
+	0x0390,  0x8395,  0x839f,  0x039a,  0x838b,  0x038e,  0x0384,  0x8381,
+	0x0280,  0x8285,  0x828f,  0x028a,  0x829b,  0x029e,  0x0294,  0x8291,
+	0x82b3,  0x02b6,  0x02bc,  0x82b9,  0x02a8,  0x82ad,  0x82a7,  0x02a2,
+	0x82e3,  0x02e6,  0x02ec,  0x82e9,  0x02f8,  0x82fd,  0x82f7,  0x02f2,
+	0x02d0,  0x82d5,  0x82df,  0x02da,  0x82cb,  0x02ce,  0x02c4,  0x82c1,
+	0x8243,  0x0246,  0x024c,  0x8249,  0x0258,  0x825d,  0x8257,  0x0252,
+	0x0270,  0x8275,  0x827f,  0x027a,  0x826b,  0x026e,  0x0264,  0x8261,
+	0x0220,  0x8225,  0x822f,  0x022a,  0x823b,  0x023e,  0x0234,  0x8231,
+	0x8213,  0x0216,  0x021c,  0x8219,  0x0208,  0x820d,  0x8207,  0x0202
+};
+
+
+void FLAC__crc8_update(const FLAC__byte data, FLAC__uint8 *crc)
+{
+	*crc = FLAC__crc8_table[*crc ^ data];
+}
+
+void FLAC__crc8_update_block(const FLAC__byte *data, unsigned len, FLAC__uint8 *crc)
+{
+	while(len--)
+		*crc = FLAC__crc8_table[*crc ^ *data++];
+}
+
+FLAC__uint8 FLAC__crc8(const FLAC__byte *data, unsigned len)
+{
+	FLAC__uint8 crc = 0;
+
+	while(len--)
+		crc = FLAC__crc8_table[crc ^ *data++];
+
+	return crc;
+}
+
+unsigned FLAC__crc16(const FLAC__byte *data, unsigned len)
+{
+	unsigned crc = 0;
+
+	while(len--)
+		crc = ((crc<<8) ^ FLAC__crc16_table[(crc>>8) ^ *data++]) & 0xffff;
+
+	return crc;
+}
+
+#endif

src/audio/audio_file_formats/flac/libFLAC/fixed.c

@@ -0,0 +1,441 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "juce_FlacHeader.h"
+#if JUCE_USE_FLAC
+
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include <math.h>
+#include <string.h>
+#include "include/private/bitmath.h"
+#include "include/private/fixed.h"
+#include "../assert.h"
+
+#ifndef M_LN2
+/* math.h in VC++ doesn't seem to have this (how Microsoft is that?) */
+#define M_LN2 0.69314718055994530942
+#endif
+
+#ifdef min
+#undef min
+#endif
+#define min(x,y) ((x) < (y)? (x) : (y))
+
+#ifdef local_abs
+#undef local_abs
+#endif
+#define local_abs(x) ((unsigned)((x)<0? -(x) : (x)))
+
+#ifdef FLAC__INTEGER_ONLY_LIBRARY
+/* rbps stands for residual bits per sample
+ *
+ *             (ln(2) * err)
+ * rbps = log  (-----------)
+ *           2 (     n     )
+ */
+static FLAC__fixedpoint local__compute_rbps_integerized(FLAC__uint32 err, FLAC__uint32 n)
+{
+	FLAC__uint32 rbps;
+	unsigned bits; /* the number of bits required to represent a number */
+	int fracbits; /* the number of bits of rbps that comprise the fractional part */
+
+	FLAC__ASSERT(sizeof(rbps) == sizeof(FLAC__fixedpoint));
+	FLAC__ASSERT(err > 0);
+	FLAC__ASSERT(n > 0);
+
+	FLAC__ASSERT(n <= FLAC__MAX_BLOCK_SIZE);
+	if(err <= n)
+		return 0;
+	/*
+	 * The above two things tell us 1) n fits in 16 bits; 2) err/n > 1.
+	 * These allow us later to know we won't lose too much precision in the
+	 * fixed-point division (err<<fracbits)/n.
+	 */
+
+	fracbits = (8*sizeof(err)) - (FLAC__bitmath_ilog2(err)+1);
+
+	err <<= fracbits;
+	err /= n;
+	/* err now holds err/n with fracbits fractional bits */
+
+	/*
+	 * Whittle err down to 16 bits max.  16 significant bits is enough for
+	 * our purposes.
+	 */
+	FLAC__ASSERT(err > 0);
+	bits = FLAC__bitmath_ilog2(err)+1;
+	if(bits > 16) {
+		err >>= (bits-16);
+		fracbits -= (bits-16);
+	}
+	rbps = (FLAC__uint32)err;
+
+	/* Multiply by fixed-point version of ln(2), with 16 fractional bits */
+	rbps *= FLAC__FP_LN2;
+	fracbits += 16;
+	FLAC__ASSERT(fracbits >= 0);
+
+	/* FLAC__fixedpoint_log2 requires fracbits%4 to be 0 */
+	{
+		const int f = fracbits & 3;
+		if(f) {
+			rbps >>= f;
+			fracbits -= f;
+		}
+	}
+
+	rbps = FLAC__fixedpoint_log2(rbps, fracbits, (unsigned)(-1));
+
+	if(rbps == 0)
+		return 0;
+
+	/*
+	 * The return value must have 16 fractional bits.  Since the whole part
+	 * of the base-2 log of a 32 bit number must fit in 5 bits, and fracbits
+	 * must be >= -3, these assertion allows us to be able to shift rbps
+	 * left if necessary to get 16 fracbits without losing any bits of the
+	 * whole part of rbps.
+	 *
+	 * There is a slight chance due to accumulated error that the whole part
+	 * will require 6 bits, so we use 6 in the assertion.  Really though as
+	 * long as it fits in 13 bits (32 - (16 - (-3))) we are fine.
+	 */
+	FLAC__ASSERT((int)FLAC__bitmath_ilog2(rbps)+1 <= fracbits + 6);
+	FLAC__ASSERT(fracbits >= -3);
+
+	/* now shift the decimal point into place */
+	if(fracbits < 16)
+		return rbps << (16-fracbits);
+	else if(fracbits > 16)
+		return rbps >> (fracbits-16);
+	else
+		return rbps;
+}
+
+static FLAC__fixedpoint local__compute_rbps_wide_integerized(FLAC__uint64 err, FLAC__uint32 n)
+{
+	FLAC__uint32 rbps;
+	unsigned bits; /* the number of bits required to represent a number */
+	int fracbits; /* the number of bits of rbps that comprise the fractional part */
+
+	FLAC__ASSERT(sizeof(rbps) == sizeof(FLAC__fixedpoint));
+	FLAC__ASSERT(err > 0);
+	FLAC__ASSERT(n > 0);
+
+	FLAC__ASSERT(n <= FLAC__MAX_BLOCK_SIZE);
+	if(err <= n)
+		return 0;
+	/*
+	 * The above two things tell us 1) n fits in 16 bits; 2) err/n > 1.
+	 * These allow us later to know we won't lose too much precision in the
+	 * fixed-point division (err<<fracbits)/n.
+	 */
+
+	fracbits = (8*sizeof(err)) - (FLAC__bitmath_ilog2_wide(err)+1);
+
+	err <<= fracbits;
+	err /= n;
+	/* err now holds err/n with fracbits fractional bits */
+
+	/*
+	 * Whittle err down to 16 bits max.  16 significant bits is enough for
+	 * our purposes.
+	 */
+	FLAC__ASSERT(err > 0);
+	bits = FLAC__bitmath_ilog2_wide(err)+1;
+	if(bits > 16) {
+		err >>= (bits-16);
+		fracbits -= (bits-16);
+	}
+	rbps = (FLAC__uint32)err;
+
+	/* Multiply by fixed-point version of ln(2), with 16 fractional bits */
+	rbps *= FLAC__FP_LN2;
+	fracbits += 16;
+	FLAC__ASSERT(fracbits >= 0);
+
+	/* FLAC__fixedpoint_log2 requires fracbits%4 to be 0 */
+	{
+		const int f = fracbits & 3;
+		if(f) {
+			rbps >>= f;
+			fracbits -= f;
+		}
+	}
+
+	rbps = FLAC__fixedpoint_log2(rbps, fracbits, (unsigned)(-1));
+
+	if(rbps == 0)
+		return 0;
+
+	/*
+	 * The return value must have 16 fractional bits.  Since the whole part
+	 * of the base-2 log of a 32 bit number must fit in 5 bits, and fracbits
+	 * must be >= -3, these assertion allows us to be able to shift rbps
+	 * left if necessary to get 16 fracbits without losing any bits of the
+	 * whole part of rbps.
+	 *
+	 * There is a slight chance due to accumulated error that the whole part
+	 * will require 6 bits, so we use 6 in the assertion.  Really though as
+	 * long as it fits in 13 bits (32 - (16 - (-3))) we are fine.
+	 */
+	FLAC__ASSERT((int)FLAC__bitmath_ilog2(rbps)+1 <= fracbits + 6);
+	FLAC__ASSERT(fracbits >= -3);
+
+	/* now shift the decimal point into place */
+	if(fracbits < 16)
+		return rbps << (16-fracbits);
+	else if(fracbits > 16)
+		return rbps >> (fracbits-16);
+	else
+		return rbps;
+}
+#endif
+
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+unsigned FLAC__fixed_compute_best_predictor(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
+#else
+unsigned FLAC__fixed_compute_best_predictor(const FLAC__int32 data[], unsigned data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
+#endif
+{
+	FLAC__int32 last_error_0 = data[-1];
+	FLAC__int32 last_error_1 = data[-1] - data[-2];
+	FLAC__int32 last_error_2 = last_error_1 - (data[-2] - data[-3]);
+	FLAC__int32 last_error_3 = last_error_2 - (data[-2] - 2*data[-3] + data[-4]);
+	FLAC__int32 error, save;
+	FLAC__uint32 total_error_0 = 0, total_error_1 = 0, total_error_2 = 0, total_error_3 = 0, total_error_4 = 0;
+	unsigned i, order;
+
+	for(i = 0; i < data_len; i++) {
+		error  = data[i]     ; total_error_0 += local_abs(error);                      save = error;
+		error -= last_error_0; total_error_1 += local_abs(error); last_error_0 = save; save = error;
+		error -= last_error_1; total_error_2 += local_abs(error); last_error_1 = save; save = error;
+		error -= last_error_2; total_error_3 += local_abs(error); last_error_2 = save; save = error;
+		error -= last_error_3; total_error_4 += local_abs(error); last_error_3 = save;
+	}
+
+	if(total_error_0 < min(min(min(total_error_1, total_error_2), total_error_3), total_error_4))
+		order = 0;
+	else if(total_error_1 < min(min(total_error_2, total_error_3), total_error_4))
+		order = 1;
+	else if(total_error_2 < min(total_error_3, total_error_4))
+		order = 2;
+	else if(total_error_3 < total_error_4)
+		order = 3;
+	else
+		order = 4;
+
+	/* Estimate the expected number of bits per residual signal sample. */
+	/* 'total_error*' is linearly related to the variance of the residual */
+	/* signal, so we use it directly to compute E(|x|) */
+	FLAC__ASSERT(data_len > 0 || total_error_0 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_1 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_2 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_3 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_4 == 0);
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+	residual_bits_per_sample[0] = (FLAC__float)((total_error_0 > 0) ? log(M_LN2 * (FLAC__double)total_error_0 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[1] = (FLAC__float)((total_error_1 > 0) ? log(M_LN2 * (FLAC__double)total_error_1 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[2] = (FLAC__float)((total_error_2 > 0) ? log(M_LN2 * (FLAC__double)total_error_2 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[3] = (FLAC__float)((total_error_3 > 0) ? log(M_LN2 * (FLAC__double)total_error_3 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[4] = (FLAC__float)((total_error_4 > 0) ? log(M_LN2 * (FLAC__double)total_error_4 / (FLAC__double)data_len) / M_LN2 : 0.0);
+#else
+	residual_bits_per_sample[0] = (total_error_0 > 0) ? local__compute_rbps_integerized(total_error_0, data_len) : 0;
+	residual_bits_per_sample[1] = (total_error_1 > 0) ? local__compute_rbps_integerized(total_error_1, data_len) : 0;
+	residual_bits_per_sample[2] = (total_error_2 > 0) ? local__compute_rbps_integerized(total_error_2, data_len) : 0;
+	residual_bits_per_sample[3] = (total_error_3 > 0) ? local__compute_rbps_integerized(total_error_3, data_len) : 0;
+	residual_bits_per_sample[4] = (total_error_4 > 0) ? local__compute_rbps_integerized(total_error_4, data_len) : 0;
+#endif
+
+	return order;
+}
+
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+unsigned FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
+#else
+unsigned FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], unsigned data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
+#endif
+{
+	FLAC__int32 last_error_0 = data[-1];
+	FLAC__int32 last_error_1 = data[-1] - data[-2];
+	FLAC__int32 last_error_2 = last_error_1 - (data[-2] - data[-3]);
+	FLAC__int32 last_error_3 = last_error_2 - (data[-2] - 2*data[-3] + data[-4]);
+	FLAC__int32 error, save;
+	/* total_error_* are 64-bits to avoid overflow when encoding
+	 * erratic signals when the bits-per-sample and blocksize are
+	 * large.
+	 */
+	FLAC__uint64 total_error_0 = 0, total_error_1 = 0, total_error_2 = 0, total_error_3 = 0, total_error_4 = 0;
+	unsigned i, order;
+
+	for(i = 0; i < data_len; i++) {
+		error  = data[i]     ; total_error_0 += local_abs(error);                      save = error;
+		error -= last_error_0; total_error_1 += local_abs(error); last_error_0 = save; save = error;
+		error -= last_error_1; total_error_2 += local_abs(error); last_error_1 = save; save = error;
+		error -= last_error_2; total_error_3 += local_abs(error); last_error_2 = save; save = error;
+		error -= last_error_3; total_error_4 += local_abs(error); last_error_3 = save;
+	}
+
+	if(total_error_0 < min(min(min(total_error_1, total_error_2), total_error_3), total_error_4))
+		order = 0;
+	else if(total_error_1 < min(min(total_error_2, total_error_3), total_error_4))
+		order = 1;
+	else if(total_error_2 < min(total_error_3, total_error_4))
+		order = 2;
+	else if(total_error_3 < total_error_4)
+		order = 3;
+	else
+		order = 4;
+
+	/* Estimate the expected number of bits per residual signal sample. */
+	/* 'total_error*' is linearly related to the variance of the residual */
+	/* signal, so we use it directly to compute E(|x|) */
+	FLAC__ASSERT(data_len > 0 || total_error_0 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_1 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_2 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_3 == 0);
+	FLAC__ASSERT(data_len > 0 || total_error_4 == 0);
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+#if defined _MSC_VER || defined __MINGW32__
+	/* with MSVC you have to spoon feed it the casting */
+	residual_bits_per_sample[0] = (FLAC__float)((total_error_0 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_0 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[1] = (FLAC__float)((total_error_1 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_1 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[2] = (FLAC__float)((total_error_2 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_2 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[3] = (FLAC__float)((total_error_3 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_3 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[4] = (FLAC__float)((total_error_4 > 0) ? log(M_LN2 * (FLAC__double)(FLAC__int64)total_error_4 / (FLAC__double)data_len) / M_LN2 : 0.0);
+#else
+	residual_bits_per_sample[0] = (FLAC__float)((total_error_0 > 0) ? log(M_LN2 * (FLAC__double)total_error_0 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[1] = (FLAC__float)((total_error_1 > 0) ? log(M_LN2 * (FLAC__double)total_error_1 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[2] = (FLAC__float)((total_error_2 > 0) ? log(M_LN2 * (FLAC__double)total_error_2 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[3] = (FLAC__float)((total_error_3 > 0) ? log(M_LN2 * (FLAC__double)total_error_3 / (FLAC__double)data_len) / M_LN2 : 0.0);
+	residual_bits_per_sample[4] = (FLAC__float)((total_error_4 > 0) ? log(M_LN2 * (FLAC__double)total_error_4 / (FLAC__double)data_len) / M_LN2 : 0.0);
+#endif
+#else
+	residual_bits_per_sample[0] = (total_error_0 > 0) ? local__compute_rbps_wide_integerized(total_error_0, data_len) : 0;
+	residual_bits_per_sample[1] = (total_error_1 > 0) ? local__compute_rbps_wide_integerized(total_error_1, data_len) : 0;
+	residual_bits_per_sample[2] = (total_error_2 > 0) ? local__compute_rbps_wide_integerized(total_error_2, data_len) : 0;
+	residual_bits_per_sample[3] = (total_error_3 > 0) ? local__compute_rbps_wide_integerized(total_error_3, data_len) : 0;
+	residual_bits_per_sample[4] = (total_error_4 > 0) ? local__compute_rbps_wide_integerized(total_error_4, data_len) : 0;
+#endif
+
+	return order;
+}
+
+void FLAC__fixed_compute_residual(const FLAC__int32 data[], unsigned data_len, unsigned order, FLAC__int32 residual[])
+{
+	const int idata_len = (int)data_len;
+	int i;
+
+	switch(order) {
+		case 0:
+			FLAC__ASSERT(sizeof(residual[0]) == sizeof(data[0]));
+			memcpy(residual, data, sizeof(residual[0])*data_len);
+			break;
+		case 1:
+			for(i = 0; i < idata_len; i++)
+				residual[i] = data[i] - data[i-1];
+			break;
+		case 2:
+			for(i = 0; i < idata_len; i++)
+#if 1 /* OPT: may be faster with some compilers on some systems */
+				residual[i] = data[i] - (data[i-1] << 1) + data[i-2];
+#else
+				residual[i] = data[i] - 2*data[i-1] + data[i-2];
+#endif
+			break;
+		case 3:
+			for(i = 0; i < idata_len; i++)
+#if 1 /* OPT: may be faster with some compilers on some systems */
+				residual[i] = data[i] - (((data[i-1]-data[i-2])<<1) + (data[i-1]-data[i-2])) - data[i-3];
+#else
+				residual[i] = data[i] - 3*data[i-1] + 3*data[i-2] - data[i-3];
+#endif
+			break;
+		case 4:
+			for(i = 0; i < idata_len; i++)
+#if 1 /* OPT: may be faster with some compilers on some systems */
+				residual[i] = data[i] - ((data[i-1]+data[i-3])<<2) + ((data[i-2]<<2) + (data[i-2]<<1)) + data[i-4];
+#else
+				residual[i] = data[i] - 4*data[i-1] + 6*data[i-2] - 4*data[i-3] + data[i-4];
+#endif
+			break;
+		default:
+			FLAC__ASSERT(0);
+	}
+}
+
+void FLAC__fixed_restore_signal(const FLAC__int32 residual[], unsigned data_len, unsigned order, FLAC__int32 data[])
+{
+	int i, idata_len = (int)data_len;
+
+	switch(order) {
+		case 0:
+			FLAC__ASSERT(sizeof(residual[0]) == sizeof(data[0]));
+			memcpy(data, residual, sizeof(residual[0])*data_len);
+			break;
+		case 1:
+			for(i = 0; i < idata_len; i++)
+				data[i] = residual[i] + data[i-1];
+			break;
+		case 2:
+			for(i = 0; i < idata_len; i++)
+#if 1 /* OPT: may be faster with some compilers on some systems */
+				data[i] = residual[i] + (data[i-1]<<1) - data[i-2];
+#else
+				data[i] = residual[i] + 2*data[i-1] - data[i-2];
+#endif
+			break;
+		case 3:
+			for(i = 0; i < idata_len; i++)
+#if 1 /* OPT: may be faster with some compilers on some systems */
+				data[i] = residual[i] + (((data[i-1]-data[i-2])<<1) + (data[i-1]-data[i-2])) + data[i-3];
+#else
+				data[i] = residual[i] + 3*data[i-1] - 3*data[i-2] + data[i-3];
+#endif
+			break;
+		case 4:
+			for(i = 0; i < idata_len; i++)
+#if 1 /* OPT: may be faster with some compilers on some systems */
+				data[i] = residual[i] + ((data[i-1]+data[i-3])<<2) - ((data[i-2]<<2) + (data[i-2]<<1)) - data[i-4];
+#else
+				data[i] = residual[i] + 4*data[i-1] - 6*data[i-2] + 4*data[i-3] - data[i-4];
+#endif
+			break;
+		default:
+			FLAC__ASSERT(0);
+	}
+}
+
+#endif

src/audio/audio_file_formats/flac/libFLAC/float.c

@@ -0,0 +1,314 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "juce_FlacHeader.h"
+#if JUCE_USE_FLAC
+
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include "../assert.h"
+
+#include "include/private/float.h"
+
+#ifdef FLAC__INTEGER_ONLY_LIBRARY
+
+/* adjust for compilers that can't understand using LLU suffix for uint64_t literals */
+#ifdef _MSC_VER
+#define FLAC__U64L(x) x
+#else
+#define FLAC__U64L(x) x##LLU
+#endif
+
+const FLAC__fixedpoint FLAC__FP_ZERO = 0;
+const FLAC__fixedpoint FLAC__FP_ONE_HALF = 0x00008000;
+const FLAC__fixedpoint FLAC__FP_ONE = 0x00010000;
+const FLAC__fixedpoint FLAC__FP_LN2 = 45426;
+const FLAC__fixedpoint FLAC__FP_E = 178145;
+
+/* Lookup tables for Knuth's logarithm algorithm */
+#define LOG2_LOOKUP_PRECISION 16
+static const FLAC__uint32 log2_lookup[][LOG2_LOOKUP_PRECISION] = {
+	{
+		/*
+		 * 0 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00000001,
+		/* lg(4/3) = */ 0x00000000,
+		/* lg(8/7) = */ 0x00000000,
+		/* lg(16/15) = */ 0x00000000,
+		/* lg(32/31) = */ 0x00000000,
+		/* lg(64/63) = */ 0x00000000,
+		/* lg(128/127) = */ 0x00000000,
+		/* lg(256/255) = */ 0x00000000,
+		/* lg(512/511) = */ 0x00000000,
+		/* lg(1024/1023) = */ 0x00000000,
+		/* lg(2048/2047) = */ 0x00000000,
+		/* lg(4096/4095) = */ 0x00000000,
+		/* lg(8192/8191) = */ 0x00000000,
+		/* lg(16384/16383) = */ 0x00000000,
+		/* lg(32768/32767) = */ 0x00000000
+	},
+	{
+		/*
+		 * 4 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00000010,
+		/* lg(4/3) = */ 0x00000007,
+		/* lg(8/7) = */ 0x00000003,
+		/* lg(16/15) = */ 0x00000001,
+		/* lg(32/31) = */ 0x00000001,
+		/* lg(64/63) = */ 0x00000000,
+		/* lg(128/127) = */ 0x00000000,
+		/* lg(256/255) = */ 0x00000000,
+		/* lg(512/511) = */ 0x00000000,
+		/* lg(1024/1023) = */ 0x00000000,
+		/* lg(2048/2047) = */ 0x00000000,
+		/* lg(4096/4095) = */ 0x00000000,
+		/* lg(8192/8191) = */ 0x00000000,
+		/* lg(16384/16383) = */ 0x00000000,
+		/* lg(32768/32767) = */ 0x00000000
+	},
+	{
+		/*
+		 * 8 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00000100,
+		/* lg(4/3) = */ 0x0000006a,
+		/* lg(8/7) = */ 0x00000031,
+		/* lg(16/15) = */ 0x00000018,
+		/* lg(32/31) = */ 0x0000000c,
+		/* lg(64/63) = */ 0x00000006,
+		/* lg(128/127) = */ 0x00000003,
+		/* lg(256/255) = */ 0x00000001,
+		/* lg(512/511) = */ 0x00000001,
+		/* lg(1024/1023) = */ 0x00000000,
+		/* lg(2048/2047) = */ 0x00000000,
+		/* lg(4096/4095) = */ 0x00000000,
+		/* lg(8192/8191) = */ 0x00000000,
+		/* lg(16384/16383) = */ 0x00000000,
+		/* lg(32768/32767) = */ 0x00000000
+	},
+	{
+		/*
+		 * 12 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00001000,
+		/* lg(4/3) = */ 0x000006a4,
+		/* lg(8/7) = */ 0x00000315,
+		/* lg(16/15) = */ 0x0000017d,
+		/* lg(32/31) = */ 0x000000bc,
+		/* lg(64/63) = */ 0x0000005d,
+		/* lg(128/127) = */ 0x0000002e,
+		/* lg(256/255) = */ 0x00000017,
+		/* lg(512/511) = */ 0x0000000c,
+		/* lg(1024/1023) = */ 0x00000006,
+		/* lg(2048/2047) = */ 0x00000003,
+		/* lg(4096/4095) = */ 0x00000001,
+		/* lg(8192/8191) = */ 0x00000001,
+		/* lg(16384/16383) = */ 0x00000000,
+		/* lg(32768/32767) = */ 0x00000000
+	},
+	{
+		/*
+		 * 16 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00010000,
+		/* lg(4/3) = */ 0x00006a40,
+		/* lg(8/7) = */ 0x00003151,
+		/* lg(16/15) = */ 0x000017d6,
+		/* lg(32/31) = */ 0x00000bba,
+		/* lg(64/63) = */ 0x000005d1,
+		/* lg(128/127) = */ 0x000002e6,
+		/* lg(256/255) = */ 0x00000172,
+		/* lg(512/511) = */ 0x000000b9,
+		/* lg(1024/1023) = */ 0x0000005c,
+		/* lg(2048/2047) = */ 0x0000002e,
+		/* lg(4096/4095) = */ 0x00000017,
+		/* lg(8192/8191) = */ 0x0000000c,
+		/* lg(16384/16383) = */ 0x00000006,
+		/* lg(32768/32767) = */ 0x00000003
+	},
+	{
+		/*
+		 * 20 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x00100000,
+		/* lg(4/3) = */ 0x0006a3fe,
+		/* lg(8/7) = */ 0x00031513,
+		/* lg(16/15) = */ 0x00017d60,
+		/* lg(32/31) = */ 0x0000bb9d,
+		/* lg(64/63) = */ 0x00005d10,
+		/* lg(128/127) = */ 0x00002e59,
+		/* lg(256/255) = */ 0x00001721,
+		/* lg(512/511) = */ 0x00000b8e,
+		/* lg(1024/1023) = */ 0x000005c6,
+		/* lg(2048/2047) = */ 0x000002e3,
+		/* lg(4096/4095) = */ 0x00000171,
+		/* lg(8192/8191) = */ 0x000000b9,
+		/* lg(16384/16383) = */ 0x0000005c,
+		/* lg(32768/32767) = */ 0x0000002e
+	},
+	{
+		/*
+		 * 24 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x01000000,
+		/* lg(4/3) = */ 0x006a3fe6,
+		/* lg(8/7) = */ 0x00315130,
+		/* lg(16/15) = */ 0x0017d605,
+		/* lg(32/31) = */ 0x000bb9ca,
+		/* lg(64/63) = */ 0x0005d0fc,
+		/* lg(128/127) = */ 0x0002e58f,
+		/* lg(256/255) = */ 0x0001720e,
+		/* lg(512/511) = */ 0x0000b8d8,
+		/* lg(1024/1023) = */ 0x00005c61,
+		/* lg(2048/2047) = */ 0x00002e2d,
+		/* lg(4096/4095) = */ 0x00001716,
+		/* lg(8192/8191) = */ 0x00000b8b,
+		/* lg(16384/16383) = */ 0x000005c5,
+		/* lg(32768/32767) = */ 0x000002e3
+	},
+	{
+		/*
+		 * 28 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ 0x10000000,
+		/* lg(4/3) = */ 0x06a3fe5c,
+		/* lg(8/7) = */ 0x03151301,
+		/* lg(16/15) = */ 0x017d6049,
+		/* lg(32/31) = */ 0x00bb9ca6,
+		/* lg(64/63) = */ 0x005d0fba,
+		/* lg(128/127) = */ 0x002e58f7,
+		/* lg(256/255) = */ 0x001720da,
+		/* lg(512/511) = */ 0x000b8d87,
+		/* lg(1024/1023) = */ 0x0005c60b,
+		/* lg(2048/2047) = */ 0x0002e2d7,
+		/* lg(4096/4095) = */ 0x00017160,
+		/* lg(8192/8191) = */ 0x0000b8ad,
+		/* lg(16384/16383) = */ 0x00005c56,
+		/* lg(32768/32767) = */ 0x00002e2b
+	}
+};
+
+#if 0
+static const FLAC__uint64 log2_lookup_wide[] = {
+	{
+		/*
+		 * 32 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ FLAC__U64L(0x100000000),
+		/* lg(4/3) = */ FLAC__U64L(0x6a3fe5c6),
+		/* lg(8/7) = */ FLAC__U64L(0x31513015),
+		/* lg(16/15) = */ FLAC__U64L(0x17d60497),
+		/* lg(32/31) = */ FLAC__U64L(0x0bb9ca65),
+		/* lg(64/63) = */ FLAC__U64L(0x05d0fba2),
+		/* lg(128/127) = */ FLAC__U64L(0x02e58f74),
+		/* lg(256/255) = */ FLAC__U64L(0x01720d9c),
+		/* lg(512/511) = */ FLAC__U64L(0x00b8d875),
+		/* lg(1024/1023) = */ FLAC__U64L(0x005c60aa),
+		/* lg(2048/2047) = */ FLAC__U64L(0x002e2d72),
+		/* lg(4096/4095) = */ FLAC__U64L(0x00171600),
+		/* lg(8192/8191) = */ FLAC__U64L(0x000b8ad2),
+		/* lg(16384/16383) = */ FLAC__U64L(0x0005c55d),
+		/* lg(32768/32767) = */ FLAC__U64L(0x0002e2ac)
+	},
+	{
+		/*
+		 * 48 fraction bits
+		 */
+		/* undefined */ 0x00000000,
+		/* lg(2/1) = */ FLAC__U64L(0x1000000000000),
+		/* lg(4/3) = */ FLAC__U64L(0x6a3fe5c60429),
+		/* lg(8/7) = */ FLAC__U64L(0x315130157f7a),
+		/* lg(16/15) = */ FLAC__U64L(0x17d60496cfbb),
+		/* lg(32/31) = */ FLAC__U64L(0xbb9ca64ecac),
+		/* lg(64/63) = */ FLAC__U64L(0x5d0fba187cd),
+		/* lg(128/127) = */ FLAC__U64L(0x2e58f7441ee),
+		/* lg(256/255) = */ FLAC__U64L(0x1720d9c06a8),
+		/* lg(512/511) = */ FLAC__U64L(0xb8d8752173),
+		/* lg(1024/1023) = */ FLAC__U64L(0x5c60aa252e),
+		/* lg(2048/2047) = */ FLAC__U64L(0x2e2d71b0d8),
+		/* lg(4096/4095) = */ FLAC__U64L(0x1716001719),
+		/* lg(8192/8191) = */ FLAC__U64L(0xb8ad1de1b),
+		/* lg(16384/16383) = */ FLAC__U64L(0x5c55d640d),
+		/* lg(32768/32767) = */ FLAC__U64L(0x2e2abcf52)
+	}
+};
+#endif
+
+FLAC__uint32 FLAC__fixedpoint_log2(FLAC__uint32 x, unsigned fracbits, unsigned precision)
+{
+	const FLAC__uint32 ONE = (1u << fracbits);
+	const FLAC__uint32 *table = log2_lookup[fracbits >> 2];
+
+	FLAC__ASSERT(fracbits < 32);
+	FLAC__ASSERT((fracbits & 0x3) == 0);
+
+	if(x < ONE)
+		return 0;
+
+	if(precision > LOG2_LOOKUP_PRECISION)
+		precision = LOG2_LOOKUP_PRECISION;
+
+	/* Knuth's algorithm for computing logarithms, optimized for base-2 with lookup tables */
+	{
+		FLAC__uint32 y = 0;
+		FLAC__uint32 z = x >> 1, k = 1;
+		while (x > ONE && k < precision) {
+			if (x - z >= ONE) {
+				x -= z;
+				z = x >> k;
+				y += table[k];
+			}
+			else {
+				z >>= 1;
+				k++;
+			}
+		}
+		return y;
+	}
+}
+
+#endif /* defined FLAC__INTEGER_ONLY_LIBRARY */
+
+#endif

src/audio/audio_file_formats/flac/libFLAC/format.c

@@ -0,0 +1,604 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "juce_FlacHeader.h"
+#if JUCE_USE_FLAC
+
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include <stdio.h>
+#include <stdlib.h> /* for qsort() */
+#include <string.h> /* for memset() */
+#include "../assert.h"
+#include "../format.h"
+#include "include/private/format.h"
+
+#ifndef FLaC__INLINE
+#define FLaC__INLINE
+#endif
+
+#ifdef min
+#undef min
+#endif
+#define min(a,b) ((a)<(b)?(a):(b))
+
+/* adjust for compilers that can't understand using LLU suffix for uint64_t literals */
+#ifdef _MSC_VER
+#define FLAC__U64L(x) x
+#else
+#define FLAC__U64L(x) x##LLU
+#endif
+
+/* VERSION should come from configure */
+FLAC_API const char *FLAC__VERSION_STRING = VERSION
+
+
+
+
+;
+
+#if defined _MSC_VER || defined __BORLANDC__ || defined __MINW32__
+/* yet one more hack because of MSVC6: */
+FLAC_API const char *FLAC__VENDOR_STRING = "reference libFLAC 1.2.1 20070917";
+#else
+FLAC_API const char *FLAC__VENDOR_STRING = "reference libFLAC " VERSION " 20070917";
+#endif
+
+FLAC_API const FLAC__byte FLAC__STREAM_SYNC_STRING[4] = { 'f','L','a','C' };
+FLAC_API const unsigned FLAC__STREAM_SYNC = 0x664C6143;
+FLAC_API const unsigned FLAC__STREAM_SYNC_LEN = 32; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MIN_BLOCK_SIZE_LEN = 16; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MAX_BLOCK_SIZE_LEN = 16; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MIN_FRAME_SIZE_LEN = 24; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MAX_FRAME_SIZE_LEN = 24; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_SAMPLE_RATE_LEN = 20; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_CHANNELS_LEN = 3; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN = 5; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_TOTAL_SAMPLES_LEN = 36; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MD5SUM_LEN = 128; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_APPLICATION_ID_LEN = 32; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_SEEKPOINT_SAMPLE_NUMBER_LEN = 64; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_SEEKPOINT_STREAM_OFFSET_LEN = 64; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_SEEKPOINT_FRAME_SAMPLES_LEN = 16; /* bits */
+
+FLAC_API const FLAC__uint64 FLAC__STREAM_METADATA_SEEKPOINT_PLACEHOLDER = FLAC__U64L(0xffffffffffffffff);
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_VORBIS_COMMENT_ENTRY_LENGTH_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_VORBIS_COMMENT_NUM_COMMENTS_LEN = 32; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_INDEX_OFFSET_LEN = 64; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_INDEX_NUMBER_LEN = 8; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_INDEX_RESERVED_LEN = 3*8; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_OFFSET_LEN = 64; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_NUMBER_LEN = 8; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_ISRC_LEN = 12*8; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_TYPE_LEN = 1; /* bit */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_PRE_EMPHASIS_LEN = 1; /* bit */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_RESERVED_LEN = 6+13*8; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_NUM_INDICES_LEN = 8; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_MEDIA_CATALOG_NUMBER_LEN = 128*8; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_LEAD_IN_LEN = 64; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_IS_CD_LEN = 1; /* bit */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_RESERVED_LEN = 7+258*8; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_NUM_TRACKS_LEN = 8; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_TYPE_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_MIME_TYPE_LENGTH_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_DESCRIPTION_LENGTH_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_WIDTH_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_HEIGHT_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_DEPTH_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_COLORS_LEN = 32; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_DATA_LENGTH_LEN = 32; /* bits */
+
+FLAC_API const unsigned FLAC__STREAM_METADATA_IS_LAST_LEN = 1; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_TYPE_LEN = 7; /* bits */
+FLAC_API const unsigned FLAC__STREAM_METADATA_LENGTH_LEN = 24; /* bits */
+
+FLAC_API const unsigned FLAC__FRAME_HEADER_SYNC = 0x3ffe;
+FLAC_API const unsigned FLAC__FRAME_HEADER_SYNC_LEN = 14; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_RESERVED_LEN = 1; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_BLOCKING_STRATEGY_LEN = 1; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_BLOCK_SIZE_LEN = 4; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_SAMPLE_RATE_LEN = 4; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_CHANNEL_ASSIGNMENT_LEN = 4; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_BITS_PER_SAMPLE_LEN = 3; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_ZERO_PAD_LEN = 1; /* bits */
+FLAC_API const unsigned FLAC__FRAME_HEADER_CRC_LEN = 8; /* bits */
+
+FLAC_API const unsigned FLAC__FRAME_FOOTER_CRC_LEN = 16; /* bits */
+
+FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_TYPE_LEN = 2; /* bits */
+FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_ORDER_LEN = 4; /* bits */
+FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_PARAMETER_LEN = 4; /* bits */
+FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_PARAMETER_LEN = 5; /* bits */
+FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_RAW_LEN = 5; /* bits */
+
+FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_ESCAPE_PARAMETER = 15; /* == (1<<FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_PARAMETER_LEN)-1 */
+FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_ESCAPE_PARAMETER = 31; /* == (1<<FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_PARAMETER_LEN)-1 */
+
+FLAC_API const char * const FLAC__EntropyCodingMethodTypeString[] = {
+	"PARTITIONED_RICE",
+	"PARTITIONED_RICE2"
+};
+
+FLAC_API const unsigned FLAC__SUBFRAME_LPC_QLP_COEFF_PRECISION_LEN = 4; /* bits */
+FLAC_API const unsigned FLAC__SUBFRAME_LPC_QLP_SHIFT_LEN = 5; /* bits */
+
+FLAC_API const unsigned FLAC__SUBFRAME_ZERO_PAD_LEN = 1; /* bits */
+FLAC_API const unsigned FLAC__SUBFRAME_TYPE_LEN = 6; /* bits */
+FLAC_API const unsigned FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN = 1; /* bits */
+
+FLAC_API const unsigned FLAC__SUBFRAME_TYPE_CONSTANT_BYTE_ALIGNED_MASK = 0x00;
+FLAC_API const unsigned FLAC__SUBFRAME_TYPE_VERBATIM_BYTE_ALIGNED_MASK = 0x02;
+FLAC_API const unsigned FLAC__SUBFRAME_TYPE_FIXED_BYTE_ALIGNED_MASK = 0x10;
+FLAC_API const unsigned FLAC__SUBFRAME_TYPE_LPC_BYTE_ALIGNED_MASK = 0x40;
+
+FLAC_API const char * const FLAC__SubframeTypeString[] = {
+	"CONSTANT",
+	"VERBATIM",
+	"FIXED",
+	"LPC"
+};
+
+FLAC_API const char * const FLAC__ChannelAssignmentString[] = {
+	"INDEPENDENT",
+	"LEFT_SIDE",
+	"RIGHT_SIDE",
+	"MID_SIDE"
+};
+
+FLAC_API const char * const FLAC__FrameNumberTypeString[] = {
+	"FRAME_NUMBER_TYPE_FRAME_NUMBER",
+	"FRAME_NUMBER_TYPE_SAMPLE_NUMBER"
+};
+
+FLAC_API const char * const FLAC__MetadataTypeString[] = {
+	"STREAMINFO",
+	"PADDING",
+	"APPLICATION",
+	"SEEKTABLE",
+	"VORBIS_COMMENT",
+	"CUESHEET",
+	"PICTURE"
+};
+
+FLAC_API const char * const FLAC__StreamMetadata_Picture_TypeString[] = {
+	"Other",
+	"32x32 pixels 'file icon' (PNG only)",
+	"Other file icon",
+	"Cover (front)",
+	"Cover (back)",
+	"Leaflet page",
+	"Media (e.g. label side of CD)",
+	"Lead artist/lead performer/soloist",
+	"Artist/performer",
+	"Conductor",
+	"Band/Orchestra",
+	"Composer",
+	"Lyricist/text writer",
+	"Recording Location",
+	"During recording",
+	"During performance",
+	"Movie/video screen capture",
+	"A bright coloured fish",
+	"Illustration",
+	"Band/artist logotype",
+	"Publisher/Studio logotype"
+};
+
+FLAC_API FLAC__bool FLAC__format_sample_rate_is_valid(unsigned sample_rate)
+{
+	if(sample_rate == 0 || sample_rate > FLAC__MAX_SAMPLE_RATE) {
+		return false;
+	}
+	else
+		return true;
+}
+
+FLAC_API FLAC__bool FLAC__format_sample_rate_is_subset(unsigned sample_rate)
+{
+	if(
+		!FLAC__format_sample_rate_is_valid(sample_rate) ||
+		(
+			sample_rate >= (1u << 16) &&
+			!(sample_rate % 1000 == 0 || sample_rate % 10 == 0)
+		)
+	) {
+		return false;
+	}
+	else
+		return true;
+}
+
+/* @@@@ add to unit tests; it is already indirectly tested by the metadata_object tests */
+FLAC_API FLAC__bool FLAC__format_seektable_is_legal(const FLAC__StreamMetadata_SeekTable *seek_table)
+{
+	unsigned i;
+	FLAC__uint64 prev_sample_number = 0;
+	FLAC__bool got_prev = false;
+
+	FLAC__ASSERT(0 != seek_table);
+
+	for(i = 0; i < seek_table->num_points; i++) {
+		if(got_prev) {
+			if(
+				seek_table->points[i].sample_number != FLAC__STREAM_METADATA_SEEKPOINT_PLACEHOLDER &&
+				seek_table->points[i].sample_number <= prev_sample_number
+			)
+				return false;
+		}
+		prev_sample_number = seek_table->points[i].sample_number;
+		got_prev = true;
+	}
+
+	return true;
+}
+
+/* used as the sort predicate for qsort() */
+static int 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 FLaC__INLINE unsigned utf8len_(const FLAC__byte *utf8)
+{
+	FLAC__ASSERT(0 != utf8);
+	if ((utf8[0] & 0x80) == 0) {
+		return 1;
+	}
+	else if ((utf8[0] & 0xE0) == 0xC0 && (utf8[1] & 0xC0) == 0x80) {
+		if ((utf8[0] & 0xFE) == 0xC0) /* overlong sequence check */
+			return 0;
+		return 2;
+	}
+	else if ((utf8[0] & 0xF0) == 0xE0 && (utf8[1] & 0xC0) == 0x80 && (utf8[2] & 0xC0) == 0x80) {
+		if (utf8[0] == 0xE0 && (utf8[1] & 0xE0) == 0x80) /* overlong sequence check */
+			return 0;
+		/* illegal surrogates check (U+D800...U+DFFF and U+FFFE...U+FFFF) */
+		if (utf8[0] == 0xED && (utf8[1] & 0xE0) == 0xA0) /* D800-DFFF */
+			return 0;
+		if (utf8[0] == 0xEF && utf8[1] == 0xBF && (utf8[2] & 0xFE) == 0xBE) /* FFFE-FFFF */
+			return 0;
+		return 3;
+	}
+	else if ((utf8[0] & 0xF8) == 0xF0 && (utf8[1] & 0xC0) == 0x80 && (utf8[2] & 0xC0) == 0x80 && (utf8[3] & 0xC0) == 0x80) {
+		if (utf8[0] == 0xF0 && (utf8[1] & 0xF0) == 0x80) /* overlong sequence check */
+			return 0;
+		return 4;
+	}
+	else if ((utf8[0] & 0xFC) == 0xF8 && (utf8[1] & 0xC0) == 0x80 && (utf8[2] & 0xC0) == 0x80 && (utf8[3] & 0xC0) == 0x80 && (utf8[4] & 0xC0) == 0x80) {
+		if (utf8[0] == 0xF8 && (utf8[1] & 0xF8) == 0x80) /* overlong sequence check */
+			return 0;
+		return 5;
+	}
+	else if ((utf8[0] & 0xFE) == 0xFC && (utf8[1] & 0xC0) == 0x80 && (utf8[2] & 0xC0) == 0x80 && (utf8[3] & 0xC0) == 0x80 && (utf8[4] & 0xC0) == 0x80 && (utf8[5] & 0xC0) == 0x80) {
+		if (utf8[0] == 0xFC && (utf8[1] & 0xFC) == 0x80) /* overlong sequence check */
+			return 0;
+		return 6;
+	}
+	else {
+		return 0;
+	}
+}
+
+FLAC_API FLAC__bool FLAC__format_vorbiscomment_entry_name_is_legal(const char *name)
+{
+	char c;
+	for(c = *name; c; c = *(++name))
+		if(c < 0x20 || c == 0x3d || c > 0x7d)
+			return false;
+	return true;
+}
+
+FLAC_API FLAC__bool FLAC__format_vorbiscomment_entry_value_is_legal(const FLAC__byte *value, unsigned length)
+{
+	if(length == (unsigned)(-1)) {
+		while(*value) {
+			unsigned n = utf8len_(value);
+			if(n == 0)
+				return false;
+			value += n;
+		}
+	}
+	else {
+		const FLAC__byte *end = value + length;
+		while(value < end) {
+			unsigned n = utf8len_(value);
+			if(n == 0)
+				return false;
+			value += n;
+		}
+		if(value != end)
+			return false;
+	}
+	return true;
+}
+
+FLAC_API FLAC__bool FLAC__format_vorbiscomment_entry_is_legal(const FLAC__byte *entry, unsigned length)
+{
+	const FLAC__byte *s, *end;
+
+	for(s = entry, end = s + length; s < end && *s != '='; s++) {
+		if(*s < 0x20 || *s > 0x7D)
+			return false;
+	}
+	if(s == end)
+		return false;
+
+	s++; /* skip '=' */
+
+	while(s < end) {
+		unsigned n = utf8len_(s);
+		if(n == 0)
+			return false;
+		s += n;
+	}
+	if(s != end)
+		return false;
+
+	return true;
+}
+
+/* @@@@ add to unit tests; it is already indirectly tested by the metadata_object tests */
+FLAC_API FLAC__bool FLAC__format_cuesheet_is_legal(const FLAC__StreamMetadata_CueSheet *cue_sheet, FLAC__bool check_cd_da_subset, const char **violation)
+{
+	unsigned i, j;
+
+	if(check_cd_da_subset) {
+		if(cue_sheet->lead_in < 2 * 44100) {
+			if(violation) *violation = "CD-DA cue sheet must have a lead-in length of at least 2 seconds";
+			return false;
+		}
+		if(cue_sheet->lead_in % 588 != 0) {
+			if(violation) *violation = "CD-DA cue sheet lead-in length must be evenly divisible by 588 samples";
+			return false;
+		}
+	}
+
+	if(cue_sheet->num_tracks == 0) {
+		if(violation) *violation = "cue sheet must have at least one track (the lead-out)";
+		return false;
+	}
+
+	if(check_cd_da_subset && cue_sheet->tracks[cue_sheet->num_tracks-1].number != 170) {
+		if(violation) *violation = "CD-DA cue sheet must have a lead-out track number 170 (0xAA)";
+		return false;
+	}
+
+	for(i = 0; i < cue_sheet->num_tracks; i++) {
+		if(cue_sheet->tracks[i].number == 0) {
+			if(violation) *violation = "cue sheet may not have a track number 0";
+			return false;
+		}
+
+		if(check_cd_da_subset) {
+			if(!((cue_sheet->tracks[i].number >= 1 && cue_sheet->tracks[i].number <= 99) || cue_sheet->tracks[i].number == 170)) {
+				if(violation) *violation = "CD-DA cue sheet track number must be 1-99 or 170";
+				return false;
+			}
+		}
+
+		if(check_cd_da_subset && cue_sheet->tracks[i].offset % 588 != 0) {
+			if(violation) {
+				if(i == cue_sheet->num_tracks-1) /* the lead-out track... */
+					*violation = "CD-DA cue sheet lead-out offset must be evenly divisible by 588 samples";
+				else
+					*violation = "CD-DA cue sheet track offset must be evenly divisible by 588 samples";
+			}
+			return false;
+		}
+
+		if(i < cue_sheet->num_tracks - 1) {
+			if(cue_sheet->tracks[i].num_indices == 0) {
+				if(violation) *violation = "cue sheet track must have at least one index point";
+				return false;
+			}
+
+			if(cue_sheet->tracks[i].indices[0].number > 1) {
+				if(violation) *violation = "cue sheet track's first index number must be 0 or 1";
+				return false;
+			}
+		}
+
+		for(j = 0; j < cue_sheet->tracks[i].num_indices; j++) {
+			if(check_cd_da_subset && cue_sheet->tracks[i].indices[j].offset % 588 != 0) {
+				if(violation) *violation = "CD-DA cue sheet track index offset must be evenly divisible by 588 samples";
+				return false;
+			}
+
+			if(j > 0) {
+				if(cue_sheet->tracks[i].indices[j].number != cue_sheet->tracks[i].indices[j-1].number + 1) {
+					if(violation) *violation = "cue sheet track index numbers must increase by 1";
+					return false;
+				}
+			}
+		}
+	}
+
+	return true;
+}
+
+/* @@@@ add to unit tests; it is already indirectly tested by the metadata_object tests */
+FLAC_API FLAC__bool FLAC__format_picture_is_legal(const FLAC__StreamMetadata_Picture *picture, const char **violation)
+{
+	char *p;
+	FLAC__byte *b;
+
+	for(p = picture->mime_type; *p; p++) {
+		if(*p < 0x20 || *p > 0x7e) {
+			if(violation) *violation = "MIME type string must contain only printable ASCII characters (0x20-0x7e)";
+			return false;
+		}
+	}
+
+	for(b = picture->description; *b; ) {
+		unsigned n = utf8len_(b);
+		if(n == 0) {
+			if(violation) *violation = "description string must be valid UTF-8";
+			return false;
+		}
+		b += n;
+	}
+
+	return true;
+}
+
+/*
+ * These routines are private to libFLAC
+ */
+unsigned FLAC__format_get_max_rice_partition_order(unsigned blocksize, unsigned predictor_order)
+{
+	return
+		FLAC__format_get_max_rice_partition_order_from_blocksize_limited_max_and_predictor_order(
+			FLAC__format_get_max_rice_partition_order_from_blocksize(blocksize),
+			blocksize,
+			predictor_order
+		);
+}
+
+unsigned FLAC__format_get_max_rice_partition_order_from_blocksize(unsigned blocksize)
+{
+	unsigned max_rice_partition_order = 0;
+	while(!(blocksize & 1)) {
+		max_rice_partition_order++;
+		blocksize >>= 1;
+	}
+	return min(FLAC__MAX_RICE_PARTITION_ORDER, max_rice_partition_order);
+}
+
+unsigned FLAC__format_get_max_rice_partition_order_from_blocksize_limited_max_and_predictor_order(unsigned limit, unsigned blocksize, unsigned predictor_order)
+{
+	unsigned max_rice_partition_order = limit;
+
+	while(max_rice_partition_order > 0 && (blocksize >> max_rice_partition_order) <= predictor_order)
+		max_rice_partition_order--;
+
+	FLAC__ASSERT(
+		(max_rice_partition_order == 0 && blocksize >= predictor_order) ||
+		(max_rice_partition_order > 0 && blocksize >> max_rice_partition_order > predictor_order)
+	);
+
+	return max_rice_partition_order;
+}
+
+void FLAC__format_entropy_coding_method_partitioned_rice_contents_init(FLAC__EntropyCodingMethod_PartitionedRiceContents *object)
+{
+	FLAC__ASSERT(0 != object);
+
+	object->parameters = 0;
+	object->raw_bits = 0;
+	object->capacity_by_order = 0;
+}
+
+void FLAC__format_entropy_coding_method_partitioned_rice_contents_clear(FLAC__EntropyCodingMethod_PartitionedRiceContents *object)
+{
+	FLAC__ASSERT(0 != object);
+
+	if(0 != object->parameters)
+		free(object->parameters);
+	if(0 != object->raw_bits)
+		free(object->raw_bits);
+	FLAC__format_entropy_coding_method_partitioned_rice_contents_init(object);
+}
+
+FLAC__bool FLAC__format_entropy_coding_method_partitioned_rice_contents_ensure_size(FLAC__EntropyCodingMethod_PartitionedRiceContents *object, unsigned max_partition_order)
+{
+	FLAC__ASSERT(0 != object);
+
+	FLAC__ASSERT(object->capacity_by_order > 0 || (0 == object->parameters && 0 == object->raw_bits));
+
+	if(object->capacity_by_order < max_partition_order) {
+		if(0 == (object->parameters = (unsigned*)realloc(object->parameters, sizeof(unsigned)*(1 << max_partition_order))))
+			return false;
+		if(0 == (object->raw_bits = (unsigned*)realloc(object->raw_bits, sizeof(unsigned)*(1 << max_partition_order))))
+			return false;
+		memset(object->raw_bits, 0, sizeof(unsigned)*(1 << max_partition_order));
+		object->capacity_by_order = max_partition_order;
+	}
+
+	return true;
+}
+
+#endif

src/audio/audio_file_formats/flac/libFLAC/include/private/all.h

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

src/audio/audio_file_formats/flac/libFLAC/include/private/bitmath.h

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

src/audio/audio_file_formats/flac/libFLAC/include/private/bitreader.h

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

src/audio/audio_file_formats/flac/libFLAC/include/private/bitwriter.h

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

src/audio/audio_file_formats/flac/libFLAC/include/private/cpu.h

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

src/audio/audio_file_formats/flac/libFLAC/include/private/crc.h

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

src/audio/audio_file_formats/flac/libFLAC/include/private/fixed.h

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

src/audio/audio_file_formats/flac/libFLAC/include/private/float.h

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

src/audio/audio_file_formats/flac/libFLAC/include/private/format.h

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

src/audio/audio_file_formats/flac/libFLAC/include/private/lpc.h

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

src/audio/audio_file_formats/flac/libFLAC/include/private/md5.h

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

src/audio/audio_file_formats/flac/libFLAC/include/private/memory.h

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

src/audio/audio_file_formats/flac/libFLAC/include/private/metadata.h

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

src/audio/audio_file_formats/flac/libFLAC/include/private/stream_encoder_framing.h

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

src/audio/audio_file_formats/flac/libFLAC/include/private/window.h

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

src/audio/audio_file_formats/flac/libFLAC/include/protected/all.h

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

src/audio/audio_file_formats/flac/libFLAC/include/protected/stream_decoder.h

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

src/audio/audio_file_formats/flac/libFLAC/include/protected/stream_encoder.h

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

src/audio/audio_file_formats/flac/libFLAC/juce_FlacHeader.h

@@ -0,0 +1,42 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+// This file is included at the start of each FLAC .c file, just to do a few housekeeping
+// tasks..
+
+
+#include "../../../../../juce_Config.h"
+
+#define VERSION "1.2.1"
+
+#define FLAC__NO_DLL 1
+
+#ifdef _MSC_VER
+  #pragma warning (disable: 4267 4127 4244 4996 4100 4701 4702 4013 4133 4206 4312)
+#endif
+
+#if ! (defined (_WIN32) || defined (_WIN64) || defined (LINUX))
+ #define FLAC__SYS_DARWIN 1
+#endif

src/audio/audio_file_formats/flac/libFLAC/md5.c

@@ -0,0 +1,431 @@
+
+#include "juce_FlacHeader.h"
+#if JUCE_USE_FLAC
+
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include <stdlib.h>		/* for malloc() */
+#include <string.h>		/* for memcpy() */
+
+#include "include/private/md5.h"
+#include "../alloc.h"
+
+#ifndef FLaC__INLINE
+#define FLaC__INLINE
+#endif
+
+/*
+ * This code implements the MD5 message-digest algorithm.
+ * The algorithm is due to Ron Rivest.  This code was
+ * written by Colin Plumb in 1993, no copyright is claimed.
+ * This code is in the public domain; do with it what you wish.
+ *
+ * Equivalent code is available from RSA Data Security, Inc.
+ * This code has been tested against that, and is equivalent,
+ * except that you don't need to include two pages of legalese
+ * with every copy.
+ *
+ * To compute the message digest of a chunk of bytes, declare an
+ * MD5Context structure, pass it to MD5Init, call MD5Update as
+ * needed on buffers full of bytes, and then call MD5Final, which
+ * will fill a supplied 16-byte array with the digest.
+ *
+ * Changed so as no longer to depend on Colin Plumb's `usual.h' header
+ * definitions; now uses stuff from dpkg's config.h.
+ *  - Ian Jackson <ijackson@nyx.cs.du.edu>.
+ * Still in the public domain.
+ *
+ * Josh Coalson: made some changes to integrate with libFLAC.
+ * Still in the public domain.
+ */
+
+/* The four core functions - F1 is optimized somewhat */
+
+/* #define F1(x, y, z) (x & y | ~x & z) */
+#define F1(x, y, z) (z ^ (x & (y ^ z)))
+#define F2(x, y, z) F1(z, x, y)
+#define F3(x, y, z) (x ^ y ^ z)
+#define F4(x, y, z) (y ^ (x | ~z))
+
+/* This is the central step in the MD5 algorithm. */
+#define MD5STEP(f,w,x,y,z,in,s) \
+	 (w += f(x,y,z) + in, w = (w<<s | w>>(32-s)) + x)
+
+/*
+ * The core of the MD5 algorithm, this alters an existing MD5 hash to
+ * reflect the addition of 16 longwords of new data.  MD5Update blocks
+ * the data and converts bytes into longwords for this routine.
+ */
+static void FLAC__MD5Transform(FLAC__uint32 buf[4], FLAC__uint32 const in[16])
+{
+	register FLAC__uint32 a, b, c, d;
+
+	a = buf[0];
+	b = buf[1];
+	c = buf[2];
+	d = buf[3];
+
+	MD5STEP(F1, a, b, c, d, in[0] + 0xd76aa478, 7);
+	MD5STEP(F1, d, a, b, c, in[1] + 0xe8c7b756, 12);
+	MD5STEP(F1, c, d, a, b, in[2] + 0x242070db, 17);
+	MD5STEP(F1, b, c, d, a, in[3] + 0xc1bdceee, 22);
+	MD5STEP(F1, a, b, c, d, in[4] + 0xf57c0faf, 7);
+	MD5STEP(F1, d, a, b, c, in[5] + 0x4787c62a, 12);
+	MD5STEP(F1, c, d, a, b, in[6] + 0xa8304613, 17);
+	MD5STEP(F1, b, c, d, a, in[7] + 0xfd469501, 22);
+	MD5STEP(F1, a, b, c, d, in[8] + 0x698098d8, 7);
+	MD5STEP(F1, d, a, b, c, in[9] + 0x8b44f7af, 12);
+	MD5STEP(F1, c, d, a, b, in[10] + 0xffff5bb1, 17);
+	MD5STEP(F1, b, c, d, a, in[11] + 0x895cd7be, 22);
+	MD5STEP(F1, a, b, c, d, in[12] + 0x6b901122, 7);
+	MD5STEP(F1, d, a, b, c, in[13] + 0xfd987193, 12);
+	MD5STEP(F1, c, d, a, b, in[14] + 0xa679438e, 17);
+	MD5STEP(F1, b, c, d, a, in[15] + 0x49b40821, 22);
+
+	MD5STEP(F2, a, b, c, d, in[1] + 0xf61e2562, 5);
+	MD5STEP(F2, d, a, b, c, in[6] + 0xc040b340, 9);
+	MD5STEP(F2, c, d, a, b, in[11] + 0x265e5a51, 14);
+	MD5STEP(F2, b, c, d, a, in[0] + 0xe9b6c7aa, 20);
+	MD5STEP(F2, a, b, c, d, in[5] + 0xd62f105d, 5);
+	MD5STEP(F2, d, a, b, c, in[10] + 0x02441453, 9);
+	MD5STEP(F2, c, d, a, b, in[15] + 0xd8a1e681, 14);
+	MD5STEP(F2, b, c, d, a, in[4] + 0xe7d3fbc8, 20);
+	MD5STEP(F2, a, b, c, d, in[9] + 0x21e1cde6, 5);
+	MD5STEP(F2, d, a, b, c, in[14] + 0xc33707d6, 9);
+	MD5STEP(F2, c, d, a, b, in[3] + 0xf4d50d87, 14);
+	MD5STEP(F2, b, c, d, a, in[8] + 0x455a14ed, 20);
+	MD5STEP(F2, a, b, c, d, in[13] + 0xa9e3e905, 5);
+	MD5STEP(F2, d, a, b, c, in[2] + 0xfcefa3f8, 9);
+	MD5STEP(F2, c, d, a, b, in[7] + 0x676f02d9, 14);
+	MD5STEP(F2, b, c, d, a, in[12] + 0x8d2a4c8a, 20);
+
+	MD5STEP(F3, a, b, c, d, in[5] + 0xfffa3942, 4);
+	MD5STEP(F3, d, a, b, c, in[8] + 0x8771f681, 11);
+	MD5STEP(F3, c, d, a, b, in[11] + 0x6d9d6122, 16);
+	MD5STEP(F3, b, c, d, a, in[14] + 0xfde5380c, 23);
+	MD5STEP(F3, a, b, c, d, in[1] + 0xa4beea44, 4);
+	MD5STEP(F3, d, a, b, c, in[4] + 0x4bdecfa9, 11);
+	MD5STEP(F3, c, d, a, b, in[7] + 0xf6bb4b60, 16);
+	MD5STEP(F3, b, c, d, a, in[10] + 0xbebfbc70, 23);
+	MD5STEP(F3, a, b, c, d, in[13] + 0x289b7ec6, 4);
+	MD5STEP(F3, d, a, b, c, in[0] + 0xeaa127fa, 11);
+	MD5STEP(F3, c, d, a, b, in[3] + 0xd4ef3085, 16);
+	MD5STEP(F3, b, c, d, a, in[6] + 0x04881d05, 23);
+	MD5STEP(F3, a, b, c, d, in[9] + 0xd9d4d039, 4);
+	MD5STEP(F3, d, a, b, c, in[12] + 0xe6db99e5, 11);
+	MD5STEP(F3, c, d, a, b, in[15] + 0x1fa27cf8, 16);
+	MD5STEP(F3, b, c, d, a, in[2] + 0xc4ac5665, 23);
+
+	MD5STEP(F4, a, b, c, d, in[0] + 0xf4292244, 6);
+	MD5STEP(F4, d, a, b, c, in[7] + 0x432aff97, 10);
+	MD5STEP(F4, c, d, a, b, in[14] + 0xab9423a7, 15);
+	MD5STEP(F4, b, c, d, a, in[5] + 0xfc93a039, 21);
+	MD5STEP(F4, a, b, c, d, in[12] + 0x655b59c3, 6);
+	MD5STEP(F4, d, a, b, c, in[3] + 0x8f0ccc92, 10);
+	MD5STEP(F4, c, d, a, b, in[10] + 0xffeff47d, 15);
+	MD5STEP(F4, b, c, d, a, in[1] + 0x85845dd1, 21);
+	MD5STEP(F4, a, b, c, d, in[8] + 0x6fa87e4f, 6);
+	MD5STEP(F4, d, a, b, c, in[15] + 0xfe2ce6e0, 10);
+	MD5STEP(F4, c, d, a, b, in[6] + 0xa3014314, 15);
+	MD5STEP(F4, b, c, d, a, in[13] + 0x4e0811a1, 21);
+	MD5STEP(F4, a, b, c, d, in[4] + 0xf7537e82, 6);
+	MD5STEP(F4, d, a, b, c, in[11] + 0xbd3af235, 10);
+	MD5STEP(F4, c, d, a, b, in[2] + 0x2ad7d2bb, 15);
+	MD5STEP(F4, b, c, d, a, in[9] + 0xeb86d391, 21);
+
+	buf[0] += a;
+	buf[1] += b;
+	buf[2] += c;
+	buf[3] += d;
+}
+
+#if WORDS_BIGENDIAN
+//@@@@@@ OPT: use bswap/intrinsics
+static void byteSwap(FLAC__uint32 *buf, unsigned words)
+{
+	register FLAC__uint32 x;
+	do {
+		x = *buf;
+		x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff);
+		*buf++ = (x >> 16) | (x << 16);
+	} while (--words);
+}
+static void byteSwapX16(FLAC__uint32 *buf)
+{
+	register FLAC__uint32 x;
+
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf++ = (x >> 16) | (x << 16);
+	x = *buf; x = ((x << 8) & 0xff00ff00) | ((x >> 8) & 0x00ff00ff); *buf   = (x >> 16) | (x << 16);
+}
+#else
+#define byteSwap(buf, words)
+#define byteSwapX16(buf)
+#endif
+
+/*
+ * Update context to reflect the concatenation of another buffer full
+ * of bytes.
+ */
+static void FLAC__MD5Update(FLAC__MD5Context *ctx, FLAC__byte const *buf, unsigned len)
+{
+	FLAC__uint32 t;
+
+	/* Update byte count */
+
+	t = ctx->bytes[0];
+	if ((ctx->bytes[0] = t + len) < t)
+		ctx->bytes[1]++;	/* Carry from low to high */
+
+	t = 64 - (t & 0x3f);	/* Space available in ctx->in (at least 1) */
+	if (t > len) {
+		memcpy((FLAC__byte *)ctx->in + 64 - t, buf, len);
+		return;
+	}
+	/* First chunk is an odd size */
+	memcpy((FLAC__byte *)ctx->in + 64 - t, buf, t);
+	byteSwapX16(ctx->in);
+	FLAC__MD5Transform(ctx->buf, ctx->in);
+	buf += t;
+	len -= t;
+
+	/* Process data in 64-byte chunks */
+	while (len >= 64) {
+		memcpy(ctx->in, buf, 64);
+		byteSwapX16(ctx->in);
+		FLAC__MD5Transform(ctx->buf, ctx->in);
+		buf += 64;
+		len -= 64;
+	}
+
+	/* Handle any remaining bytes of data. */
+	memcpy(ctx->in, buf, len);
+}
+
+/*
+ * Start MD5 accumulation.  Set bit count to 0 and buffer to mysterious
+ * initialization constants.
+ */
+void FLAC__MD5Init(FLAC__MD5Context *ctx)
+{
+	ctx->buf[0] = 0x67452301;
+	ctx->buf[1] = 0xefcdab89;
+	ctx->buf[2] = 0x98badcfe;
+	ctx->buf[3] = 0x10325476;
+
+	ctx->bytes[0] = 0;
+	ctx->bytes[1] = 0;
+
+	ctx->internal_buf = 0;
+	ctx->capacity = 0;
+}
+
+/*
+ * Final wrapup - pad to 64-byte boundary with the bit pattern
+ * 1 0* (64-bit count of bits processed, MSB-first)
+ */
+void FLAC__MD5Final(FLAC__byte digest[16], FLAC__MD5Context *ctx)
+{
+	int count = ctx->bytes[0] & 0x3f;	/* Number of bytes in ctx->in */
+	FLAC__byte *p = (FLAC__byte *)ctx->in + count;
+
+	/* Set the first char of padding to 0x80.  There is always room. */
+	*p++ = 0x80;
+
+	/* Bytes of padding needed to make 56 bytes (-8..55) */
+	count = 56 - 1 - count;
+
+	if (count < 0) {	/* Padding forces an extra block */
+		memset(p, 0, count + 8);
+		byteSwapX16(ctx->in);
+		FLAC__MD5Transform(ctx->buf, ctx->in);
+		p = (FLAC__byte *)ctx->in;
+		count = 56;
+	}
+	memset(p, 0, count);
+	byteSwap(ctx->in, 14);
+
+	/* Append length in bits and transform */
+	ctx->in[14] = ctx->bytes[0] << 3;
+	ctx->in[15] = ctx->bytes[1] << 3 | ctx->bytes[0] >> 29;
+	FLAC__MD5Transform(ctx->buf, ctx->in);
+
+	byteSwap(ctx->buf, 4);
+	memcpy(digest, ctx->buf, 16);
+	memset(ctx, 0, sizeof(ctx));	/* In case it's sensitive */
+	if(0 != ctx->internal_buf) {
+		free(ctx->internal_buf);
+		ctx->internal_buf = 0;
+		ctx->capacity = 0;
+	}
+}
+
+/*
+ * Convert the incoming audio signal to a byte stream
+ */
+static void format_input_(FLAC__byte *buf, const FLAC__int32 * const signal[], unsigned channels, unsigned samples, unsigned bytes_per_sample)
+{
+	unsigned channel, sample;
+	register FLAC__int32 a_word;
+	register FLAC__byte *buf_ = buf;
+
+#if WORDS_BIGENDIAN
+#else
+	if(channels == 2 && bytes_per_sample == 2) {
+		FLAC__int16 *buf1_ = ((FLAC__int16*)buf_) + 1;
+		memcpy(buf_, signal[0], sizeof(FLAC__int32) * samples);
+		for(sample = 0; sample < samples; sample++, buf1_+=2)
+			*buf1_ = (FLAC__int16)signal[1][sample];
+	}
+	else if(channels == 1 && bytes_per_sample == 2) {
+		FLAC__int16 *buf1_ = (FLAC__int16*)buf_;
+		for(sample = 0; sample < samples; sample++)
+			*buf1_++ = (FLAC__int16)signal[0][sample];
+	}
+	else
+#endif
+	if(bytes_per_sample == 2) {
+		if(channels == 2) {
+			for(sample = 0; sample < samples; sample++) {
+				a_word = signal[0][sample];
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word;
+				a_word = signal[1][sample];
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word;
+			}
+		}
+		else if(channels == 1) {
+			for(sample = 0; sample < samples; sample++) {
+				a_word = signal[0][sample];
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word;
+			}
+		}
+		else {
+			for(sample = 0; sample < samples; sample++) {
+				for(channel = 0; channel < channels; channel++) {
+					a_word = signal[channel][sample];
+					*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+					*buf_++ = (FLAC__byte)a_word;
+				}
+			}
+		}
+	}
+	else if(bytes_per_sample == 3) {
+		if(channels == 2) {
+			for(sample = 0; sample < samples; sample++) {
+				a_word = signal[0][sample];
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word;
+				a_word = signal[1][sample];
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word;
+			}
+		}
+		else if(channels == 1) {
+			for(sample = 0; sample < samples; sample++) {
+				a_word = signal[0][sample];
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word;
+			}
+		}
+		else {
+			for(sample = 0; sample < samples; sample++) {
+				for(channel = 0; channel < channels; channel++) {
+					a_word = signal[channel][sample];
+					*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+					*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+					*buf_++ = (FLAC__byte)a_word;
+				}
+			}
+		}
+	}
+	else if(bytes_per_sample == 1) {
+		if(channels == 2) {
+			for(sample = 0; sample < samples; sample++) {
+				a_word = signal[0][sample];
+				*buf_++ = (FLAC__byte)a_word;
+				a_word = signal[1][sample];
+				*buf_++ = (FLAC__byte)a_word;
+			}
+		}
+		else if(channels == 1) {
+			for(sample = 0; sample < samples; sample++) {
+				a_word = signal[0][sample];
+				*buf_++ = (FLAC__byte)a_word;
+			}
+		}
+		else {
+			for(sample = 0; sample < samples; sample++) {
+				for(channel = 0; channel < channels; channel++) {
+					a_word = signal[channel][sample];
+					*buf_++ = (FLAC__byte)a_word;
+				}
+			}
+		}
+	}
+	else { /* bytes_per_sample == 4, maybe optimize more later */
+		for(sample = 0; sample < samples; sample++) {
+			for(channel = 0; channel < channels; channel++) {
+				a_word = signal[channel][sample];
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
+				*buf_++ = (FLAC__byte)a_word;
+			}
+		}
+	}
+}
+
+/*
+ * Convert the incoming audio signal to a byte stream and FLAC__MD5Update it.
+ */
+FLAC__bool FLAC__MD5Accumulate(FLAC__MD5Context *ctx, const FLAC__int32 * const signal[], unsigned channels, unsigned samples, unsigned bytes_per_sample)
+{
+	const size_t bytes_needed = (size_t)channels * (size_t)samples * (size_t)bytes_per_sample;
+
+	/* overflow check */
+	if((size_t)channels > SIZE_MAX / (size_t)bytes_per_sample)
+		return false;
+	if((size_t)channels * (size_t)bytes_per_sample > SIZE_MAX / (size_t)samples)
+		return false;
+
+	if(ctx->capacity < bytes_needed) {
+		FLAC__byte *tmp = (FLAC__byte*)realloc(ctx->internal_buf, bytes_needed);
+		if(0 == tmp) {
+			free(ctx->internal_buf);
+			if(0 == (ctx->internal_buf = (FLAC__byte*)safe_malloc_(bytes_needed)))
+				return false;
+		}
+		ctx->internal_buf = tmp;
+		ctx->capacity = bytes_needed;
+	}
+
+	format_input_(ctx->internal_buf, signal, channels, samples, bytes_per_sample);
+
+	FLAC__MD5Update(ctx, ctx->internal_buf, bytes_needed);
+
+	return true;
+}
+
+#endif

src/audio/audio_file_formats/flac/libFLAC/memory.c

@@ -0,0 +1,227 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "juce_FlacHeader.h"
+#if JUCE_USE_FLAC
+
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include "include/private/memory.h"
+#include "../assert.h"
+#include "../alloc.h"
+
+void *FLAC__memory_alloc_aligned(size_t bytes, void **aligned_address)
+{
+	void *x;
+
+	FLAC__ASSERT(0 != aligned_address);
+
+#ifdef FLAC__ALIGN_MALLOC_DATA
+	/* align on 32-byte (256-bit) boundary */
+	x = safe_malloc_add_2op_(bytes, /*+*/31);
+#ifdef SIZEOF_VOIDP
+#if SIZEOF_VOIDP == 4
+		/* could do  *aligned_address = x + ((unsigned) (32 - (((unsigned)x) & 31))) & 31; */
+		*aligned_address = (void*)(((unsigned)x + 31) & -32);
+#elif SIZEOF_VOIDP == 8
+		*aligned_address = (void*)(((FLAC__uint64)x + 31) & (FLAC__uint64)(-((FLAC__int64)32)));
+#else
+# error  Unsupported sizeof(void*)
+#endif
+#else
+	/* there's got to be a better way to do this right for all archs */
+	if(sizeof(void*) == sizeof(unsigned))
+		*aligned_address = (void*)(((unsigned)x + 31) & -32);
+	else if(sizeof(void*) == sizeof(FLAC__uint64))
+		*aligned_address = (void*)(((FLAC__uint64)x + 31) & (FLAC__uint64)(-((FLAC__int64)32)));
+	else
+		return 0;
+#endif
+#else
+	x = safe_malloc_(bytes);
+	*aligned_address = x;
+#endif
+	return x;
+}
+
+FLAC__bool FLAC__memory_alloc_aligned_int32_array(unsigned elements, FLAC__int32 **unaligned_pointer, FLAC__int32 **aligned_pointer)
+{
+	FLAC__int32 *pu; /* unaligned pointer */
+	union { /* union needed to comply with C99 pointer aliasing rules */
+		FLAC__int32 *pa; /* aligned pointer */
+		void        *pv; /* aligned pointer alias */
+	} u;
+
+	FLAC__ASSERT(elements > 0);
+	FLAC__ASSERT(0 != unaligned_pointer);
+	FLAC__ASSERT(0 != aligned_pointer);
+	FLAC__ASSERT(unaligned_pointer != aligned_pointer);
+
+	if((size_t)elements > SIZE_MAX / sizeof(*pu)) /* overflow check */
+		return false;
+
+	pu = (FLAC__int32*)FLAC__memory_alloc_aligned(sizeof(*pu) * (size_t)elements, &u.pv);
+	if(0 == pu) {
+		return false;
+	}
+	else {
+		if(*unaligned_pointer != 0)
+			free(*unaligned_pointer);
+		*unaligned_pointer = pu;
+		*aligned_pointer = u.pa;
+		return true;
+	}
+}
+
+FLAC__bool FLAC__memory_alloc_aligned_uint32_array(unsigned elements, FLAC__uint32 **unaligned_pointer, FLAC__uint32 **aligned_pointer)
+{
+	FLAC__uint32 *pu; /* unaligned pointer */
+	union { /* union needed to comply with C99 pointer aliasing rules */
+		FLAC__uint32 *pa; /* aligned pointer */
+		void         *pv; /* aligned pointer alias */
+	} u;
+
+	FLAC__ASSERT(elements > 0);
+	FLAC__ASSERT(0 != unaligned_pointer);
+	FLAC__ASSERT(0 != aligned_pointer);
+	FLAC__ASSERT(unaligned_pointer != aligned_pointer);
+
+	if((size_t)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(unsigned elements, FLAC__uint64 **unaligned_pointer, FLAC__uint64 **aligned_pointer)
+{
+	FLAC__uint64 *pu; /* unaligned pointer */
+	union { /* union needed to comply with C99 pointer aliasing rules */
+		FLAC__uint64 *pa; /* aligned pointer */
+		void         *pv; /* aligned pointer alias */
+	} u;
+
+	FLAC__ASSERT(elements > 0);
+	FLAC__ASSERT(0 != unaligned_pointer);
+	FLAC__ASSERT(0 != aligned_pointer);
+	FLAC__ASSERT(unaligned_pointer != aligned_pointer);
+
+	if((size_t)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(unsigned elements, unsigned **unaligned_pointer, unsigned **aligned_pointer)
+{
+	unsigned *pu; /* unaligned pointer */
+	union { /* union needed to comply with C99 pointer aliasing rules */
+		unsigned *pa; /* aligned pointer */
+		void     *pv; /* aligned pointer alias */
+	} u;
+
+	FLAC__ASSERT(elements > 0);
+	FLAC__ASSERT(0 != unaligned_pointer);
+	FLAC__ASSERT(0 != aligned_pointer);
+	FLAC__ASSERT(unaligned_pointer != aligned_pointer);
+
+	if((size_t)elements > SIZE_MAX / sizeof(*pu)) /* overflow check */
+		return false;
+
+	pu = (unsigned*)FLAC__memory_alloc_aligned(sizeof(*pu) * elements, &u.pv);
+	if(0 == pu) {
+		return false;
+	}
+	else {
+		if(*unaligned_pointer != 0)
+			free(*unaligned_pointer);
+		*unaligned_pointer = pu;
+		*aligned_pointer = u.pa;
+		return true;
+	}
+}
+
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+
+FLAC__bool FLAC__memory_alloc_aligned_real_array(unsigned elements, FLAC__real **unaligned_pointer, FLAC__real **aligned_pointer)
+{
+	FLAC__real *pu; /* unaligned pointer */
+	union { /* union needed to comply with C99 pointer aliasing rules */
+		FLAC__real *pa; /* aligned pointer */
+		void       *pv; /* aligned pointer alias */
+	} u;
+
+	FLAC__ASSERT(elements > 0);
+	FLAC__ASSERT(0 != unaligned_pointer);
+	FLAC__ASSERT(0 != aligned_pointer);
+	FLAC__ASSERT(unaligned_pointer != aligned_pointer);
+
+	if((size_t)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
+
+#endif

src/audio/audio_file_formats/flac/libFLAC/stream_encoder_framing.c

@@ -0,0 +1,559 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2000,2001,2002,2003,2004,2005,2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "juce_FlacHeader.h"
+#if JUCE_USE_FLAC
+
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include <stdio.h>
+#include <string.h> /* for strlen() */
+#include "include/private/stream_encoder_framing.h"
+#include "include/private/crc.h"
+#include "../assert.h"
+
+#ifdef max
+#undef max
+#endif
+#define max(x,y) ((x)>(y)?(x):(y))
+
+static FLAC__bool add_entropy_coding_method_(FLAC__BitWriter *bw, const FLAC__EntropyCodingMethod *method);
+static FLAC__bool add_residual_partitioned_rice_(FLAC__BitWriter *bw, const FLAC__int32 residual[], const unsigned residual_samples, const unsigned predictor_order, const unsigned rice_parameters[], const unsigned raw_bits[], const unsigned partition_order, const FLAC__bool is_extended);
+
+FLAC__bool FLAC__add_metadata_block(const FLAC__StreamMetadata *metadata, FLAC__BitWriter *bw)
+{
+	unsigned i, j;
+	const unsigned vendor_string_length = (unsigned)strlen(FLAC__VENDOR_STRING);
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->is_last, FLAC__STREAM_METADATA_IS_LAST_LEN))
+		return false;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->type, FLAC__STREAM_METADATA_TYPE_LEN))
+		return false;
+
+	/*
+	 * First, for VORBIS_COMMENTs, adjust the length to reflect our vendor string
+	 */
+	i = metadata->length;
+	if(metadata->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) {
+		FLAC__ASSERT(metadata->data.vorbis_comment.vendor_string.length == 0 || 0 != metadata->data.vorbis_comment.vendor_string.entry);
+		i -= metadata->data.vorbis_comment.vendor_string.length;
+		i += vendor_string_length;
+	}
+	FLAC__ASSERT(i < (1u << FLAC__STREAM_METADATA_LENGTH_LEN));
+	if(!FLAC__bitwriter_write_raw_uint32(bw, i, FLAC__STREAM_METADATA_LENGTH_LEN))
+		return false;
+
+	switch(metadata->type) {
+		case FLAC__METADATA_TYPE_STREAMINFO:
+			FLAC__ASSERT(metadata->data.stream_info.min_blocksize < (1u << FLAC__STREAM_METADATA_STREAMINFO_MIN_BLOCK_SIZE_LEN));
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.min_blocksize, FLAC__STREAM_METADATA_STREAMINFO_MIN_BLOCK_SIZE_LEN))
+				return false;
+			FLAC__ASSERT(metadata->data.stream_info.max_blocksize < (1u << FLAC__STREAM_METADATA_STREAMINFO_MAX_BLOCK_SIZE_LEN));
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.max_blocksize, FLAC__STREAM_METADATA_STREAMINFO_MAX_BLOCK_SIZE_LEN))
+				return false;
+			FLAC__ASSERT(metadata->data.stream_info.min_framesize < (1u << FLAC__STREAM_METADATA_STREAMINFO_MIN_FRAME_SIZE_LEN));
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.min_framesize, FLAC__STREAM_METADATA_STREAMINFO_MIN_FRAME_SIZE_LEN))
+				return false;
+			FLAC__ASSERT(metadata->data.stream_info.max_framesize < (1u << FLAC__STREAM_METADATA_STREAMINFO_MAX_FRAME_SIZE_LEN));
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.max_framesize, FLAC__STREAM_METADATA_STREAMINFO_MAX_FRAME_SIZE_LEN))
+				return false;
+			FLAC__ASSERT(FLAC__format_sample_rate_is_valid(metadata->data.stream_info.sample_rate));
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.sample_rate, FLAC__STREAM_METADATA_STREAMINFO_SAMPLE_RATE_LEN))
+				return false;
+			FLAC__ASSERT(metadata->data.stream_info.channels > 0);
+			FLAC__ASSERT(metadata->data.stream_info.channels <= (1u << FLAC__STREAM_METADATA_STREAMINFO_CHANNELS_LEN));
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.channels-1, FLAC__STREAM_METADATA_STREAMINFO_CHANNELS_LEN))
+				return false;
+			FLAC__ASSERT(metadata->data.stream_info.bits_per_sample > 0);
+			FLAC__ASSERT(metadata->data.stream_info.bits_per_sample <= (1u << FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN));
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.bits_per_sample-1, FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN))
+				return false;
+			if(!FLAC__bitwriter_write_raw_uint64(bw, metadata->data.stream_info.total_samples, FLAC__STREAM_METADATA_STREAMINFO_TOTAL_SAMPLES_LEN))
+				return false;
+			if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.stream_info.md5sum, 16))
+				return false;
+			break;
+		case FLAC__METADATA_TYPE_PADDING:
+			if(!FLAC__bitwriter_write_zeroes(bw, metadata->length * 8))
+				return false;
+			break;
+		case FLAC__METADATA_TYPE_APPLICATION:
+			if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.application.id, FLAC__STREAM_METADATA_APPLICATION_ID_LEN / 8))
+				return false;
+			if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.application.data, metadata->length - (FLAC__STREAM_METADATA_APPLICATION_ID_LEN / 8)))
+				return false;
+			break;
+		case FLAC__METADATA_TYPE_SEEKTABLE:
+			for(i = 0; i < metadata->data.seek_table.num_points; i++) {
+				if(!FLAC__bitwriter_write_raw_uint64(bw, metadata->data.seek_table.points[i].sample_number, FLAC__STREAM_METADATA_SEEKPOINT_SAMPLE_NUMBER_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint64(bw, metadata->data.seek_table.points[i].stream_offset, FLAC__STREAM_METADATA_SEEKPOINT_STREAM_OFFSET_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.seek_table.points[i].frame_samples, FLAC__STREAM_METADATA_SEEKPOINT_FRAME_SAMPLES_LEN))
+					return false;
+			}
+			break;
+		case FLAC__METADATA_TYPE_VORBIS_COMMENT:
+			if(!FLAC__bitwriter_write_raw_uint32_little_endian(bw, vendor_string_length))
+				return false;
+			if(!FLAC__bitwriter_write_byte_block(bw, (const FLAC__byte*)FLAC__VENDOR_STRING, vendor_string_length))
+				return false;
+			if(!FLAC__bitwriter_write_raw_uint32_little_endian(bw, metadata->data.vorbis_comment.num_comments))
+				return false;
+			for(i = 0; i < metadata->data.vorbis_comment.num_comments; i++) {
+				if(!FLAC__bitwriter_write_raw_uint32_little_endian(bw, metadata->data.vorbis_comment.comments[i].length))
+					return false;
+				if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.vorbis_comment.comments[i].entry, metadata->data.vorbis_comment.comments[i].length))
+					return false;
+			}
+			break;
+		case FLAC__METADATA_TYPE_CUESHEET:
+			FLAC__ASSERT(FLAC__STREAM_METADATA_CUESHEET_MEDIA_CATALOG_NUMBER_LEN % 8 == 0);
+			if(!FLAC__bitwriter_write_byte_block(bw, (const FLAC__byte*)metadata->data.cue_sheet.media_catalog_number, FLAC__STREAM_METADATA_CUESHEET_MEDIA_CATALOG_NUMBER_LEN/8))
+				return false;
+			if(!FLAC__bitwriter_write_raw_uint64(bw, metadata->data.cue_sheet.lead_in, FLAC__STREAM_METADATA_CUESHEET_LEAD_IN_LEN))
+				return false;
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.cue_sheet.is_cd? 1 : 0, FLAC__STREAM_METADATA_CUESHEET_IS_CD_LEN))
+				return false;
+			if(!FLAC__bitwriter_write_zeroes(bw, FLAC__STREAM_METADATA_CUESHEET_RESERVED_LEN))
+				return false;
+			if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.cue_sheet.num_tracks, FLAC__STREAM_METADATA_CUESHEET_NUM_TRACKS_LEN))
+				return false;
+			for(i = 0; i < metadata->data.cue_sheet.num_tracks; i++) {
+				const FLAC__StreamMetadata_CueSheet_Track *track = metadata->data.cue_sheet.tracks + i;
+
+				if(!FLAC__bitwriter_write_raw_uint64(bw, track->offset, FLAC__STREAM_METADATA_CUESHEET_TRACK_OFFSET_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, track->number, FLAC__STREAM_METADATA_CUESHEET_TRACK_NUMBER_LEN))
+					return false;
+				FLAC__ASSERT(FLAC__STREAM_METADATA_CUESHEET_TRACK_ISRC_LEN % 8 == 0);
+				if(!FLAC__bitwriter_write_byte_block(bw, (const FLAC__byte*)track->isrc, FLAC__STREAM_METADATA_CUESHEET_TRACK_ISRC_LEN/8))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, track->type, FLAC__STREAM_METADATA_CUESHEET_TRACK_TYPE_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, track->pre_emphasis, FLAC__STREAM_METADATA_CUESHEET_TRACK_PRE_EMPHASIS_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_zeroes(bw, FLAC__STREAM_METADATA_CUESHEET_TRACK_RESERVED_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, track->num_indices, FLAC__STREAM_METADATA_CUESHEET_TRACK_NUM_INDICES_LEN))
+					return false;
+				for(j = 0; j < track->num_indices; j++) {
+					const FLAC__StreamMetadata_CueSheet_Index *index = track->indices + j;
+
+					if(!FLAC__bitwriter_write_raw_uint64(bw, index->offset, FLAC__STREAM_METADATA_CUESHEET_INDEX_OFFSET_LEN))
+						return false;
+					if(!FLAC__bitwriter_write_raw_uint32(bw, index->number, FLAC__STREAM_METADATA_CUESHEET_INDEX_NUMBER_LEN))
+						return false;
+					if(!FLAC__bitwriter_write_zeroes(bw, FLAC__STREAM_METADATA_CUESHEET_INDEX_RESERVED_LEN))
+						return false;
+				}
+			}
+			break;
+		case FLAC__METADATA_TYPE_PICTURE:
+			{
+				size_t len;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.type, FLAC__STREAM_METADATA_PICTURE_TYPE_LEN))
+					return false;
+				len = strlen(metadata->data.picture.mime_type);
+				if(!FLAC__bitwriter_write_raw_uint32(bw, len, FLAC__STREAM_METADATA_PICTURE_MIME_TYPE_LENGTH_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_byte_block(bw, (const FLAC__byte*)metadata->data.picture.mime_type, len))
+					return false;
+				len = strlen((const char *)metadata->data.picture.description);
+				if(!FLAC__bitwriter_write_raw_uint32(bw, len, FLAC__STREAM_METADATA_PICTURE_DESCRIPTION_LENGTH_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.picture.description, len))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.width, FLAC__STREAM_METADATA_PICTURE_WIDTH_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.height, FLAC__STREAM_METADATA_PICTURE_HEIGHT_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.depth, FLAC__STREAM_METADATA_PICTURE_DEPTH_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.colors, FLAC__STREAM_METADATA_PICTURE_COLORS_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.data_length, FLAC__STREAM_METADATA_PICTURE_DATA_LENGTH_LEN))
+					return false;
+				if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.picture.data, metadata->data.picture.data_length))
+					return false;
+			}
+			break;
+		default:
+			if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.unknown.data, metadata->length))
+				return false;
+			break;
+	}
+
+	FLAC__ASSERT(FLAC__bitwriter_is_byte_aligned(bw));
+	return true;
+}
+
+FLAC__bool FLAC__frame_add_header(const FLAC__FrameHeader *header, FLAC__BitWriter *bw)
+{
+	unsigned u, blocksize_hint, sample_rate_hint;
+	FLAC__byte crc;
+
+	FLAC__ASSERT(FLAC__bitwriter_is_byte_aligned(bw));
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, FLAC__FRAME_HEADER_SYNC, FLAC__FRAME_HEADER_SYNC_LEN))
+		return false;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, 0, FLAC__FRAME_HEADER_RESERVED_LEN))
+		return false;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, (header->number_type == FLAC__FRAME_NUMBER_TYPE_FRAME_NUMBER)? 0 : 1, FLAC__FRAME_HEADER_BLOCKING_STRATEGY_LEN))
+		return false;
+
+	FLAC__ASSERT(header->blocksize > 0 && header->blocksize <= FLAC__MAX_BLOCK_SIZE);
+	/* when this assertion holds true, any legal blocksize can be expressed in the frame header */
+	FLAC__ASSERT(FLAC__MAX_BLOCK_SIZE <= 65535u);
+	blocksize_hint = 0;
+	switch(header->blocksize) {
+		case   192: u = 1; break;
+		case   576: u = 2; break;
+		case  1152: u = 3; break;
+		case  2304: u = 4; break;
+		case  4608: u = 5; break;
+		case   256: u = 8; break;
+		case   512: u = 9; break;
+		case  1024: u = 10; break;
+		case  2048: u = 11; break;
+		case  4096: u = 12; break;
+		case  8192: u = 13; break;
+		case 16384: u = 14; break;
+		case 32768: u = 15; break;
+		default:
+			if(header->blocksize <= 0x100)
+				blocksize_hint = u = 6;
+			else
+				blocksize_hint = u = 7;
+			break;
+	}
+	if(!FLAC__bitwriter_write_raw_uint32(bw, u, FLAC__FRAME_HEADER_BLOCK_SIZE_LEN))
+		return false;
+
+	FLAC__ASSERT(FLAC__format_sample_rate_is_valid(header->sample_rate));
+	sample_rate_hint = 0;
+	switch(header->sample_rate) {
+		case  88200: u = 1; break;
+		case 176400: u = 2; break;
+		case 192000: u = 3; break;
+		case   8000: u = 4; break;
+		case  16000: u = 5; break;
+		case  22050: u = 6; break;
+		case  24000: u = 7; break;
+		case  32000: u = 8; break;
+		case  44100: u = 9; break;
+		case  48000: u = 10; break;
+		case  96000: u = 11; break;
+		default:
+			if(header->sample_rate <= 255000 && header->sample_rate % 1000 == 0)
+				sample_rate_hint = u = 12;
+			else if(header->sample_rate % 10 == 0)
+				sample_rate_hint = u = 14;
+			else if(header->sample_rate <= 0xffff)
+				sample_rate_hint = u = 13;
+			else
+				u = 0;
+			break;
+	}
+	if(!FLAC__bitwriter_write_raw_uint32(bw, u, FLAC__FRAME_HEADER_SAMPLE_RATE_LEN))
+		return false;
+
+	FLAC__ASSERT(header->channels > 0 && header->channels <= (1u << FLAC__STREAM_METADATA_STREAMINFO_CHANNELS_LEN) && header->channels <= FLAC__MAX_CHANNELS);
+	switch(header->channel_assignment) {
+		case FLAC__CHANNEL_ASSIGNMENT_INDEPENDENT:
+			u = header->channels - 1;
+			break;
+		case FLAC__CHANNEL_ASSIGNMENT_LEFT_SIDE:
+			FLAC__ASSERT(header->channels == 2);
+			u = 8;
+			break;
+		case FLAC__CHANNEL_ASSIGNMENT_RIGHT_SIDE:
+			FLAC__ASSERT(header->channels == 2);
+			u = 9;
+			break;
+		case FLAC__CHANNEL_ASSIGNMENT_MID_SIDE:
+			FLAC__ASSERT(header->channels == 2);
+			u = 10;
+			break;
+		default:
+			FLAC__ASSERT(0);
+	}
+	if(!FLAC__bitwriter_write_raw_uint32(bw, u, FLAC__FRAME_HEADER_CHANNEL_ASSIGNMENT_LEN))
+		return false;
+
+	FLAC__ASSERT(header->bits_per_sample > 0 && header->bits_per_sample <= (1u << FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN));
+	switch(header->bits_per_sample) {
+		case 8 : u = 1; break;
+		case 12: u = 2; break;
+		case 16: u = 4; break;
+		case 20: u = 5; break;
+		case 24: u = 6; break;
+		default: u = 0; break;
+	}
+	if(!FLAC__bitwriter_write_raw_uint32(bw, u, FLAC__FRAME_HEADER_BITS_PER_SAMPLE_LEN))
+		return false;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, 0, FLAC__FRAME_HEADER_ZERO_PAD_LEN))
+		return false;
+
+	if(header->number_type == FLAC__FRAME_NUMBER_TYPE_FRAME_NUMBER) {
+		if(!FLAC__bitwriter_write_utf8_uint32(bw, header->number.frame_number))
+			return false;
+	}
+	else {
+		if(!FLAC__bitwriter_write_utf8_uint64(bw, header->number.sample_number))
+			return false;
+	}
+
+	if(blocksize_hint)
+		if(!FLAC__bitwriter_write_raw_uint32(bw, header->blocksize-1, (blocksize_hint==6)? 8:16))
+			return false;
+
+	switch(sample_rate_hint) {
+		case 12:
+			if(!FLAC__bitwriter_write_raw_uint32(bw, header->sample_rate / 1000, 8))
+				return false;
+			break;
+		case 13:
+			if(!FLAC__bitwriter_write_raw_uint32(bw, header->sample_rate, 16))
+				return false;
+			break;
+		case 14:
+			if(!FLAC__bitwriter_write_raw_uint32(bw, header->sample_rate / 10, 16))
+				return false;
+			break;
+	}
+
+	/* write the CRC */
+	if(!FLAC__bitwriter_get_write_crc8(bw, &crc))
+		return false;
+	if(!FLAC__bitwriter_write_raw_uint32(bw, crc, FLAC__FRAME_HEADER_CRC_LEN))
+		return false;
+
+	return true;
+}
+
+FLAC__bool FLAC__subframe_add_constant(const FLAC__Subframe_Constant *subframe, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw)
+{
+	FLAC__bool ok;
+
+	ok =
+		FLAC__bitwriter_write_raw_uint32(bw, FLAC__SUBFRAME_TYPE_CONSTANT_BYTE_ALIGNED_MASK | (wasted_bits? 1:0), FLAC__SUBFRAME_ZERO_PAD_LEN + FLAC__SUBFRAME_TYPE_LEN + FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN) &&
+		(wasted_bits? FLAC__bitwriter_write_unary_unsigned(bw, wasted_bits-1) : true) &&
+		FLAC__bitwriter_write_raw_int32(bw, subframe->value, subframe_bps)
+	;
+
+	return ok;
+}
+
+FLAC__bool FLAC__subframe_add_fixed(const FLAC__Subframe_Fixed *subframe, unsigned residual_samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw)
+{
+	unsigned i;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, FLAC__SUBFRAME_TYPE_FIXED_BYTE_ALIGNED_MASK | (subframe->order<<1) | (wasted_bits? 1:0), FLAC__SUBFRAME_ZERO_PAD_LEN + FLAC__SUBFRAME_TYPE_LEN + FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN))
+		return false;
+	if(wasted_bits)
+		if(!FLAC__bitwriter_write_unary_unsigned(bw, wasted_bits-1))
+			return false;
+
+	for(i = 0; i < subframe->order; i++)
+		if(!FLAC__bitwriter_write_raw_int32(bw, subframe->warmup[i], subframe_bps))
+			return false;
+
+	if(!add_entropy_coding_method_(bw, &subframe->entropy_coding_method))
+		return false;
+	switch(subframe->entropy_coding_method.type) {
+		case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE:
+		case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2:
+			if(!add_residual_partitioned_rice_(
+				bw,
+				subframe->residual,
+				residual_samples,
+				subframe->order,
+				subframe->entropy_coding_method.data.partitioned_rice.contents->parameters,
+				subframe->entropy_coding_method.data.partitioned_rice.contents->raw_bits,
+				subframe->entropy_coding_method.data.partitioned_rice.order,
+				/*is_extended=*/subframe->entropy_coding_method.type == FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2
+			))
+				return false;
+			break;
+		default:
+			FLAC__ASSERT(0);
+	}
+
+	return true;
+}
+
+FLAC__bool FLAC__subframe_add_lpc(const FLAC__Subframe_LPC *subframe, unsigned residual_samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw)
+{
+	unsigned i;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, FLAC__SUBFRAME_TYPE_LPC_BYTE_ALIGNED_MASK | ((subframe->order-1)<<1) | (wasted_bits? 1:0), FLAC__SUBFRAME_ZERO_PAD_LEN + FLAC__SUBFRAME_TYPE_LEN + FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN))
+		return false;
+	if(wasted_bits)
+		if(!FLAC__bitwriter_write_unary_unsigned(bw, wasted_bits-1))
+			return false;
+
+	for(i = 0; i < subframe->order; i++)
+		if(!FLAC__bitwriter_write_raw_int32(bw, subframe->warmup[i], subframe_bps))
+			return false;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, subframe->qlp_coeff_precision-1, FLAC__SUBFRAME_LPC_QLP_COEFF_PRECISION_LEN))
+		return false;
+	if(!FLAC__bitwriter_write_raw_int32(bw, subframe->quantization_level, FLAC__SUBFRAME_LPC_QLP_SHIFT_LEN))
+		return false;
+	for(i = 0; i < subframe->order; i++)
+		if(!FLAC__bitwriter_write_raw_int32(bw, subframe->qlp_coeff[i], subframe->qlp_coeff_precision))
+			return false;
+
+	if(!add_entropy_coding_method_(bw, &subframe->entropy_coding_method))
+		return false;
+	switch(subframe->entropy_coding_method.type) {
+		case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE:
+		case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2:
+			if(!add_residual_partitioned_rice_(
+				bw,
+				subframe->residual,
+				residual_samples,
+				subframe->order,
+				subframe->entropy_coding_method.data.partitioned_rice.contents->parameters,
+				subframe->entropy_coding_method.data.partitioned_rice.contents->raw_bits,
+				subframe->entropy_coding_method.data.partitioned_rice.order,
+				/*is_extended=*/subframe->entropy_coding_method.type == FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2
+			))
+				return false;
+			break;
+		default:
+			FLAC__ASSERT(0);
+	}
+
+	return true;
+}
+
+FLAC__bool FLAC__subframe_add_verbatim(const FLAC__Subframe_Verbatim *subframe, unsigned samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw)
+{
+	unsigned i;
+	const FLAC__int32 *signal = subframe->data;
+
+	if(!FLAC__bitwriter_write_raw_uint32(bw, FLAC__SUBFRAME_TYPE_VERBATIM_BYTE_ALIGNED_MASK | (wasted_bits? 1:0), FLAC__SUBFRAME_ZERO_PAD_LEN + FLAC__SUBFRAME_TYPE_LEN + FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN))
+		return false;
+	if(wasted_bits)
+		if(!FLAC__bitwriter_write_unary_unsigned(bw, wasted_bits-1))
+			return false;
+
+	for(i = 0; i < samples; i++)
+		if(!FLAC__bitwriter_write_raw_int32(bw, signal[i], subframe_bps))
+			return false;
+
+	return true;
+}
+
+FLAC__bool add_entropy_coding_method_(FLAC__BitWriter *bw, const FLAC__EntropyCodingMethod *method)
+{
+	if(!FLAC__bitwriter_write_raw_uint32(bw, method->type, FLAC__ENTROPY_CODING_METHOD_TYPE_LEN))
+		return false;
+	switch(method->type) {
+		case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE:
+		case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2:
+			if(!FLAC__bitwriter_write_raw_uint32(bw, method->data.partitioned_rice.order, FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_ORDER_LEN))
+				return false;
+			break;
+		default:
+			FLAC__ASSERT(0);
+	}
+	return true;
+}
+
+FLAC__bool add_residual_partitioned_rice_(FLAC__BitWriter *bw, const FLAC__int32 residual[], const unsigned residual_samples, const unsigned predictor_order, const unsigned rice_parameters[], const unsigned raw_bits[], const unsigned partition_order, const FLAC__bool is_extended)
+{
+	const unsigned plen = is_extended? FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_PARAMETER_LEN : FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_PARAMETER_LEN;
+	const unsigned pesc = is_extended? FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_ESCAPE_PARAMETER : FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_ESCAPE_PARAMETER;
+
+	if(partition_order == 0) {
+		unsigned i;
+
+		if(raw_bits[0] == 0) {
+			if(!FLAC__bitwriter_write_raw_uint32(bw, rice_parameters[0], plen))
+				return false;
+			if(!FLAC__bitwriter_write_rice_signed_block(bw, residual, residual_samples, rice_parameters[0]))
+				return false;
+		}
+		else {
+			FLAC__ASSERT(rice_parameters[0] == 0);
+			if(!FLAC__bitwriter_write_raw_uint32(bw, pesc, plen))
+				return false;
+			if(!FLAC__bitwriter_write_raw_uint32(bw, raw_bits[0], FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_RAW_LEN))
+				return false;
+			for(i = 0; i < residual_samples; i++) {
+				if(!FLAC__bitwriter_write_raw_int32(bw, residual[i], raw_bits[0]))
+					return false;
+			}
+		}
+		return true;
+	}
+	else {
+		unsigned i, j, k = 0, k_last = 0;
+		unsigned partition_samples;
+		const unsigned default_partition_samples = (residual_samples+predictor_order) >> partition_order;
+		for(i = 0; i < (1u<<partition_order); i++) {
+			partition_samples = default_partition_samples;
+			if(i == 0)
+				partition_samples -= predictor_order;
+			k += partition_samples;
+			if(raw_bits[i] == 0) {
+				if(!FLAC__bitwriter_write_raw_uint32(bw, rice_parameters[i], plen))
+					return false;
+				if(!FLAC__bitwriter_write_rice_signed_block(bw, residual+k_last, k-k_last, rice_parameters[i]))
+					return false;
+			}
+			else {
+				if(!FLAC__bitwriter_write_raw_uint32(bw, pesc, plen))
+					return false;
+				if(!FLAC__bitwriter_write_raw_uint32(bw, raw_bits[i], FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_RAW_LEN))
+					return false;
+				for(j = k_last; j < k; j++) {
+					if(!FLAC__bitwriter_write_raw_int32(bw, residual[j], raw_bits[i]))
+						return false;
+				}
+			}
+			k_last = k;
+		}
+		return true;
+	}
+}
+
+#endif

src/audio/audio_file_formats/flac/libFLAC/window_flac.c

@@ -0,0 +1,230 @@
+/* libFLAC - Free Lossless Audio Codec library
+ * Copyright (C) 2006,2007  Josh Coalson
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ *
+ * - Redistributions of source code must retain the above copyright
+ * notice, this list of conditions and the following disclaimer.
+ *
+ * - Redistributions in binary form must reproduce the above copyright
+ * notice, this list of conditions and the following disclaimer in the
+ * documentation and/or other materials provided with the distribution.
+ *
+ * - Neither the name of the Xiph.org Foundation nor the names of its
+ * contributors may be used to endorse or promote products derived from
+ * this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+ * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
+ * LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
+ * A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
+ * CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+ * EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
+ * PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
+ * LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
+ * NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
+ * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "juce_FlacHeader.h"
+#if JUCE_USE_FLAC
+
+#if HAVE_CONFIG_H
+#  include <config.h>
+#endif
+
+#include <math.h>
+#include "../assert.h"
+#include "../format.h"
+#include "include/private/window.h"
+
+#ifndef FLAC__INTEGER_ONLY_LIBRARY
+
+#ifndef M_PI
+/* math.h in VC++ doesn't seem to have this (how Microsoft is that?) */
+#define M_PI 3.14159265358979323846
+#endif
+
+
+void FLAC__window_bartlett(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	if (L & 1) {
+		for (n = 0; n <= N/2; n++)
+			window[n] = 2.0f * n / (float)N;
+		for (; n <= N; n++)
+			window[n] = 2.0f - 2.0f * n / (float)N;
+	}
+	else {
+		for (n = 0; n <= L/2-1; n++)
+			window[n] = 2.0f * n / (float)N;
+		for (; n <= N; n++)
+			window[n] = 2.0f - 2.0f * (N-n) / (float)N;
+	}
+}
+
+void FLAC__window_bartlett_hann(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = (FLAC__real)(0.62f - 0.48f * fabs((float)n/(float)N+0.5f) + 0.38f * cos(2.0f * M_PI * ((float)n/(float)N+0.5f)));
+}
+
+void FLAC__window_blackman(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = (FLAC__real)(0.42f - 0.5f * cos(2.0f * M_PI * n / N) + 0.08f * cos(4.0f * M_PI * n / N));
+}
+
+/* 4-term -92dB side-lobe */
+void FLAC__window_blackman_harris_4term_92db_sidelobe(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n <= N; n++)
+		window[n] = (FLAC__real)(0.35875f - 0.48829f * cos(2.0f * M_PI * n / N) + 0.14128f * cos(4.0f * M_PI * n / N) - 0.01168f * cos(6.0f * M_PI * n / N));
+}
+
+void FLAC__window_connes(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	const double N2 = (double)N / 2.;
+	FLAC__int32 n;
+
+	for (n = 0; n <= N; n++) {
+		double k = ((double)n - N2) / N2;
+		k = 1.0f - k * k;
+		window[n] = (FLAC__real)(k * k);
+	}
+}
+
+void FLAC__window_flattop(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = (FLAC__real)(1.0f - 1.93f * cos(2.0f * M_PI * n / N) + 1.29f * cos(4.0f * M_PI * n / N) - 0.388f * cos(6.0f * M_PI * n / N) + 0.0322f * cos(8.0f * M_PI * n / N));
+}
+
+void FLAC__window_gauss(FLAC__real *window, const FLAC__int32 L, const FLAC__real stddev)
+{
+	const FLAC__int32 N = L - 1;
+	const double N2 = (double)N / 2.;
+	FLAC__int32 n;
+
+	for (n = 0; n <= N; n++) {
+		const double k = ((double)n - N2) / (stddev * N2);
+		window[n] = (FLAC__real)exp(-0.5f * k * k);
+	}
+}
+
+void FLAC__window_hamming(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = (FLAC__real)(0.54f - 0.46f * cos(2.0f * M_PI * n / N));
+}
+
+void FLAC__window_hann(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = (FLAC__real)(0.5f - 0.5f * cos(2.0f * M_PI * n / N));
+}
+
+void FLAC__window_kaiser_bessel(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = (FLAC__real)(0.402f - 0.498f * cos(2.0f * M_PI * n / N) + 0.098f * cos(4.0f * M_PI * n / N) - 0.001f * cos(6.0f * M_PI * n / N));
+}
+
+void FLAC__window_nuttall(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = (FLAC__real)(0.3635819f - 0.4891775f*cos(2.0f*M_PI*n/N) + 0.1365995f*cos(4.0f*M_PI*n/N) - 0.0106411f*cos(6.0f*M_PI*n/N));
+}
+
+void FLAC__window_rectangle(FLAC__real *window, const FLAC__int32 L)
+{
+	FLAC__int32 n;
+
+	for (n = 0; n < L; n++)
+		window[n] = 1.0f;
+}
+
+void FLAC__window_triangle(FLAC__real *window, const FLAC__int32 L)
+{
+	FLAC__int32 n;
+
+	if (L & 1) {
+		for (n = 1; n <= L+1/2; n++)
+			window[n-1] = 2.0f * n / ((float)L + 1.0f);
+		for (; n <= L; n++)
+			window[n-1] = - (float)(2 * (L - n + 1)) / ((float)L + 1.0f);
+	}
+	else {
+		for (n = 1; n <= L/2; n++)
+			window[n-1] = 2.0f * n / (float)L;
+		for (; n <= L; n++)
+			window[n-1] = ((float)(2 * (L - n)) + 1.0f) / (float)L;
+	}
+}
+
+void FLAC__window_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p)
+{
+	if (p <= 0.0)
+		FLAC__window_rectangle(window, L);
+	else if (p >= 1.0)
+		FLAC__window_hann(window, L);
+	else {
+		const FLAC__int32 Np = (FLAC__int32)(p / 2.0f * L) - 1;
+		FLAC__int32 n;
+		/* start with rectangle... */
+		FLAC__window_rectangle(window, L);
+		/* ...replace ends with hann */
+		if (Np > 0) {
+			for (n = 0; n <= Np; n++) {
+				window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * n / Np));
+				window[L-Np-1+n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * (n+Np) / Np));
+			}
+		}
+	}
+}
+
+void FLAC__window_welch(FLAC__real *window, const FLAC__int32 L)
+{
+	const FLAC__int32 N = L - 1;
+	const double N2 = (double)N / 2.;
+	FLAC__int32 n;
+
+	for (n = 0; n <= N; n++) {
+		const double k = ((double)n - N2) / N2;
+		window[n] = (FLAC__real)(1.0f - k * k);
+	}
+}
+
+#endif /* !defined FLAC__INTEGER_ONLY_LIBRARY */
+
+#endif

src/audio/audio_file_formats/flac/ordinals.h

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

src/audio/audio_file_formats/juce_AiffAudioFormat.cpp

@@ -0,0 +1,813 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#include "../../core/juce_StandardHeader.h"
+
+BEGIN_JUCE_NAMESPACE
+
+#include "juce_AiffAudioFormat.h"
+#include "../../io/streams/juce_BufferedInputStream.h"
+#include "../../core/juce_PlatformUtilities.h"
+#include "../../text/juce_LocalisedStrings.h"
+
+
+#undef chunkName
+#define chunkName(a) (int)littleEndianInt(a)
+
+//==============================================================================
+#define aiffFormatName                          TRANS("AIFF file")
+static const tchar* const aiffExtensions[] =    { T(".aiff"), T(".aif"), 0 };
+
+
+//==============================================================================
+class AiffAudioFormatReader  : public AudioFormatReader
+{
+public:
+    int bytesPerFrame;
+    int64 dataChunkStart;
+    bool littleEndian;
+
+    //==============================================================================
+    AiffAudioFormatReader (InputStream* in)
+        : AudioFormatReader (in, aiffFormatName)
+    {
+        if (input->readInt() == chunkName ("FORM"))
+        {
+            const int len = input->readIntBigEndian();
+            const int64 end = input->getPosition() + len;
+
+            const int nextType = input->readInt();
+            if (nextType == chunkName ("AIFF") || nextType == chunkName ("AIFC"))
+            {
+                bool hasGotVer = false;
+                bool hasGotData = false;
+                bool hasGotType = false;
+
+                while (input->getPosition() < end)
+                {
+                    const int type = input->readInt();
+                    const uint32 length = (uint32) input->readIntBigEndian();
+                    const int64 chunkEnd = input->getPosition() + length;
+
+                    if (type == chunkName ("FVER"))
+                    {
+                        hasGotVer = true;
+
+                        const int ver = input->readIntBigEndian();
+                        if (ver != 0 && ver != (int)0xa2805140)
+                            break;
+                    }
+                    else if (type == chunkName ("COMM"))
+                    {
+                        hasGotType = true;
+
+                        numChannels = (unsigned int)input->readShortBigEndian();
+                        lengthInSamples = input->readIntBigEndian();
+                        bitsPerSample = input->readShortBigEndian();
+                        bytesPerFrame = (numChannels * bitsPerSample) >> 3;
+
+                        unsigned char sampleRateBytes[10];
+                        input->read (sampleRateBytes, 10);
+                        const int byte0 = sampleRateBytes[0];
+
+                        if ((byte0 & 0x80) != 0
+                             || byte0 <= 0x3F || byte0 > 0x40
+                             || (byte0 == 0x40 && sampleRateBytes[1] > 0x1C))
+                            break;
+
+                        unsigned int sampRate = bigEndianInt ((char*) sampleRateBytes + 2);
+                        sampRate >>= (16414 - bigEndianShort ((char*) sampleRateBytes));
+                        sampleRate = (int)sampRate;
+
+                        if (length <= 18)
+                        {
+                            // some types don't have a chunk large enough to include a compression
+                            // type, so assume it's just big-endian pcm
+                            littleEndian = false;
+                        }
+                        else
+                        {
+                            const int compType = input->readInt();
+
+                            if (compType == chunkName ("NONE") || compType == chunkName ("twos"))
+                            {
+                                littleEndian = false;
+                            }
+                            else if (compType == chunkName ("sowt"))
+                            {
+                                littleEndian = true;
+                            }
+                            else
+                            {
+                                sampleRate = 0;
+                                break;
+                            }
+                        }
+                    }
+                    else if (type == chunkName ("SSND"))
+                    {
+                        hasGotData = true;
+
+                        const int offset = input->readIntBigEndian();
+                        dataChunkStart = input->getPosition() + 4 + offset;
+                        lengthInSamples = (bytesPerFrame > 0) ? jmin (lengthInSamples, (int64) (length / bytesPerFrame)) : 0;
+                    }
+                    else if ((hasGotVer && hasGotData && hasGotType)
+                              || chunkEnd < input->getPosition()
+                              || input->isExhausted())
+                    {
+                        break;
+                    }
+
+                    input->setPosition (chunkEnd);
+                }
+            }
+        }
+    }
+
+    ~AiffAudioFormatReader()
+    {
+    }
+
+    //==============================================================================
+    bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                      int64 startSampleInFile, int numSamples)
+    {
+        numSamples = (int) jmin ((int64) numSamples, lengthInSamples - startSampleInFile);
+
+        if (numSamples <= 0)
+            return true;
+
+        input->setPosition (dataChunkStart + startSampleInFile * bytesPerFrame);
+
+        const int tempBufSize = 480 * 3 * 4; // (keep this a multiple of 3)
+        char tempBuffer [tempBufSize];
+
+        while (numSamples > 0)
+        {
+            int* left = destSamples[0];
+            if (left != 0)
+                left += startOffsetInDestBuffer;
+
+            int* right = numDestChannels > 1 ? destSamples[1] : 0;
+            if (right != 0)
+                right += startOffsetInDestBuffer;
+
+            const int numThisTime = jmin (tempBufSize / bytesPerFrame, numSamples);
+            const int bytesRead = input->read (tempBuffer, numThisTime * bytesPerFrame);
+
+            if (bytesRead < numThisTime * bytesPerFrame)
+                zeromem (tempBuffer + bytesRead, numThisTime * bytesPerFrame - bytesRead);
+
+            if (bitsPerSample == 16)
+            {
+                if (littleEndian)
+                {
+                    const short* src = (const short*) tempBuffer;
+
+                    if (numChannels > 1)
+                    {
+                        if (left == 0)
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                *right++ = (int) swapIfBigEndian ((unsigned short) *src++) << 16;
+                                ++src;
+                            }
+                        }
+                        else if (right == 0)
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                ++src;
+                                *left++ = (int) swapIfBigEndian ((unsigned short) *src++) << 16;
+                            }
+                        }
+                        else
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                *left++ = (int) swapIfBigEndian ((unsigned short) *src++) << 16;
+                                *right++ = (int) swapIfBigEndian ((unsigned short) *src++) << 16;
+                            }
+                        }
+                    }
+                    else
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *left++ = (int) swapIfBigEndian ((unsigned short) *src++) << 16;
+                        }
+                    }
+                }
+                else
+                {
+                    const char* src = (const char*) tempBuffer;
+
+                    if (numChannels > 1)
+                    {
+                        if (left == 0)
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                *right++ = bigEndianShort (src) << 16;
+                                src += 4;
+                            }
+                        }
+                        else if (right == 0)
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                src += 2;
+                                *left++ = bigEndianShort (src) << 16;
+                                src += 2;
+                            }
+                        }
+                        else
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                *left++ = bigEndianShort (src) << 16;
+                                src += 2;
+                                *right++ = bigEndianShort (src) << 16;
+                                src += 2;
+                            }
+                        }
+                    }
+                    else
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *left++ = bigEndianShort (src) << 16;
+                            src += 2;
+                        }
+                    }
+                }
+            }
+            else if (bitsPerSample == 24)
+            {
+                const char* src = (const char*)tempBuffer;
+
+                if (littleEndian)
+                {
+                    if (numChannels > 1)
+                    {
+                        if (left == 0)
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                *right++ = littleEndian24Bit (src) << 8;
+                                src += 6;
+                            }
+                        }
+                        else if (right == 0)
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                src += 3;
+                                *left++ = littleEndian24Bit (src) << 8;
+                                src += 3;
+                            }
+                        }
+                        else
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                *left++ = littleEndian24Bit (src) << 8;
+                                src += 3;
+                                *right++ = littleEndian24Bit (src) << 8;
+                                src += 3;
+                            }
+                        }
+                    }
+                    else
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *left++ = littleEndian24Bit (src) << 8;
+                            src += 3;
+                        }
+                    }
+                }
+                else
+                {
+                    if (numChannels > 1)
+                    {
+                        if (left == 0)
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                *right++ = bigEndian24Bit (src) << 8;
+                                src += 6;
+                            }
+                        }
+                        else if (right == 0)
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                src += 3;
+                                *left++ = bigEndian24Bit (src) << 8;
+                                src += 3;
+                            }
+                        }
+                        else
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                *left++ = bigEndian24Bit (src) << 8;
+                                src += 3;
+                                *right++ = bigEndian24Bit (src) << 8;
+                                src += 3;
+                            }
+                        }
+                    }
+                    else
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *left++ = bigEndian24Bit (src) << 8;
+                            src += 3;
+                        }
+                    }
+                }
+            }
+            else if (bitsPerSample == 32)
+            {
+                const unsigned int* src = (const unsigned int*) tempBuffer;
+                unsigned int* l = (unsigned int*) left;
+                unsigned int* r = (unsigned int*) right;
+
+                if (littleEndian)
+                {
+                    if (numChannels > 1)
+                    {
+                        if (l == 0)
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                ++src;
+                                *r++ = swapIfBigEndian (*src++);
+                            }
+                        }
+                        else if (r == 0)
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                *l++ = swapIfBigEndian (*src++);
+                                ++src;
+                            }
+                        }
+                        else
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                *l++ = swapIfBigEndian (*src++);
+                                *r++ = swapIfBigEndian (*src++);
+                            }
+                        }
+                    }
+                    else
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *l++ = swapIfBigEndian (*src++);
+                        }
+                    }
+                }
+                else
+                {
+                    if (numChannels > 1)
+                    {
+                        if (l == 0)
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                ++src;
+                                *r++ = swapIfLittleEndian (*src++);
+                            }
+                        }
+                        else if (r == 0)
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                *l++ = swapIfLittleEndian (*src++);
+                                ++src;
+                            }
+                        }
+                        else
+                        {
+                            for (int i = numThisTime; --i >= 0;)
+                            {
+                                *l++ = swapIfLittleEndian (*src++);
+                                *r++ = swapIfLittleEndian (*src++);
+                            }
+                        }
+                    }
+                    else
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *l++ = swapIfLittleEndian (*src++);
+                        }
+                    }
+                }
+
+                left = (int*) l;
+                right = (int*) r;
+            }
+            else if (bitsPerSample == 8)
+            {
+                const char* src = (const char*) tempBuffer;
+
+                if (numChannels > 1)
+                {
+                    if (left == 0)
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *right++ = ((int) *src++) << 24;
+                            ++src;
+                        }
+                    }
+                    else if (right == 0)
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            ++src;
+                            *left++ = ((int) *src++) << 24;
+                        }
+                    }
+                    else
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *left++ = ((int) *src++) << 24;
+                            *right++ = ((int) *src++) << 24;
+                        }
+                    }
+                }
+                else
+                {
+                    for (int i = numThisTime; --i >= 0;)
+                    {
+                        *left++ = ((int) *src++) << 24;
+                    }
+                }
+            }
+
+            startOffsetInDestBuffer += numThisTime;
+            numSamples -= numThisTime;
+        }
+
+        if (numSamples > 0)
+        {
+            for (int i = numDestChannels; --i >= 0;)
+                if (destSamples[i] != 0)
+                    zeromem (destSamples[i] + startOffsetInDestBuffer,
+                             sizeof (int) * numSamples);
+        }
+
+        return true;
+    }
+
+    juce_UseDebuggingNewOperator
+
+private:
+    AiffAudioFormatReader (const AiffAudioFormatReader&);
+    const AiffAudioFormatReader& operator= (const AiffAudioFormatReader&);
+};
+
+//==============================================================================
+class AiffAudioFormatWriter  : public AudioFormatWriter
+{
+    MemoryBlock tempBlock;
+    uint32 lengthInSamples, bytesWritten;
+    int64 headerPosition;
+    bool writeFailed;
+
+    AiffAudioFormatWriter (const AiffAudioFormatWriter&);
+    const AiffAudioFormatWriter& operator= (const AiffAudioFormatWriter&);
+
+    void writeHeader()
+    {
+        const bool couldSeekOk = output->setPosition (headerPosition);
+        (void) couldSeekOk;
+
+        // if this fails, you've given it an output stream that can't seek! It needs
+        // to be able to seek back to write the header
+        jassert (couldSeekOk);
+
+        const int headerLen = 54;
+        int audioBytes = lengthInSamples * ((bitsPerSample * numChannels) / 8);
+        audioBytes += (audioBytes & 1);
+
+        output->writeInt (chunkName ("FORM"));
+        output->writeIntBigEndian (headerLen + audioBytes - 8);
+        output->writeInt (chunkName ("AIFF"));
+        output->writeInt (chunkName ("COMM"));
+        output->writeIntBigEndian (18);
+        output->writeShortBigEndian ((short) numChannels);
+        output->writeIntBigEndian (lengthInSamples);
+        output->writeShortBigEndian ((short) bitsPerSample);
+
+        uint8 sampleRateBytes[10];
+        zeromem (sampleRateBytes, 10);
+
+        if (sampleRate <= 1)
+        {
+            sampleRateBytes[0] = 0x3f;
+            sampleRateBytes[1] = 0xff;
+            sampleRateBytes[2] = 0x80;
+        }
+        else
+        {
+            int mask = 0x40000000;
+            sampleRateBytes[0] = 0x40;
+
+            if (sampleRate >= mask)
+            {
+                jassertfalse
+                sampleRateBytes[1] = 0x1d;
+            }
+            else
+            {
+                int n = (int) sampleRate;
+
+                int i;
+                for (i = 0; i <= 32 ; ++i)
+                {
+                    if ((n & mask) != 0)
+                        break;
+
+                    mask >>= 1;
+                }
+
+                n = n << (i + 1);
+
+                sampleRateBytes[1] = (uint8) (29 - i);
+                sampleRateBytes[2] = (uint8) ((n >> 24) & 0xff);
+                sampleRateBytes[3] = (uint8) ((n >> 16) & 0xff);
+                sampleRateBytes[4] = (uint8) ((n >>  8) & 0xff);
+                sampleRateBytes[5] = (uint8) (n & 0xff);
+            }
+        }
+
+        output->write (sampleRateBytes, 10);
+
+        output->writeInt (chunkName ("SSND"));
+        output->writeIntBigEndian (audioBytes + 8);
+        output->writeInt (0);
+        output->writeInt (0);
+
+        jassert (output->getPosition() == headerLen);
+    }
+
+public:
+    //==============================================================================
+    AiffAudioFormatWriter (OutputStream* out,
+                           const double sampleRate_,
+                           const unsigned int chans,
+                           const int bits)
+        : AudioFormatWriter (out,
+                             aiffFormatName,
+                             sampleRate_,
+                             chans,
+                             bits),
+          lengthInSamples (0),
+          bytesWritten (0),
+          writeFailed (false)
+    {
+        headerPosition = out->getPosition();
+        writeHeader();
+    }
+
+    ~AiffAudioFormatWriter()
+    {
+        if ((bytesWritten & 1) != 0)
+            output->writeByte (0);
+
+        writeHeader();
+    }
+
+    //==============================================================================
+    bool write (const int** data, int numSamples)
+    {
+        if (writeFailed)
+            return false;
+
+        const int bytes = numChannels * numSamples * bitsPerSample / 8;
+        tempBlock.ensureSize (bytes, false);
+        char* buffer = (char*) tempBlock.getData();
+
+        const int* left = data[0];
+        const int* right = data[1];
+        if (right == 0)
+            right = left;
+
+        if (bitsPerSample == 16)
+        {
+            short* b = (short*) buffer;
+
+            if (numChannels > 1)
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    *b++ = (short) swapIfLittleEndian ((unsigned short) (*left++ >> 16));
+                    *b++ = (short) swapIfLittleEndian ((unsigned short) (*right++ >> 16));
+                }
+            }
+            else
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    *b++ = (short) swapIfLittleEndian ((unsigned short) (*left++ >> 16));
+                }
+            }
+        }
+        else if (bitsPerSample == 24)
+        {
+            char* b = (char*) buffer;
+
+            if (numChannels > 1)
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    bigEndian24BitToChars (*left++ >> 8, b);
+                    b += 3;
+                    bigEndian24BitToChars (*right++ >> 8, b);
+                    b += 3;
+                }
+            }
+            else
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    bigEndian24BitToChars (*left++ >> 8, b);
+                    b += 3;
+                }
+            }
+        }
+        else if (bitsPerSample == 32)
+        {
+            unsigned int* b = (unsigned int*) buffer;
+
+            if (numChannels > 1)
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    *b++ = swapIfLittleEndian ((unsigned int) *left++);
+                    *b++ = swapIfLittleEndian ((unsigned int) *right++);
+                }
+            }
+            else
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    *b++ = swapIfLittleEndian ((unsigned int) *left++);
+                }
+            }
+        }
+        else if (bitsPerSample == 8)
+        {
+            char* b = (char*)buffer;
+
+            if (numChannels > 1)
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    *b++ = (char) (*left++ >> 24);
+                    *b++ = (char) (*right++ >> 24);
+                }
+            }
+            else
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    *b++ = (char) (*left++ >> 24);
+                }
+            }
+        }
+
+        if (bytesWritten + bytes >= (uint32) 0xfff00000
+             || ! output->write (buffer, bytes))
+        {
+            // failed to write to disk, so let's try writing the header.
+            // If it's just run out of disk space, then if it does manage
+            // to write the header, we'll still have a useable file..
+            writeHeader();
+            writeFailed = true;
+            return false;
+        }
+        else
+        {
+            bytesWritten += bytes;
+            lengthInSamples += numSamples;
+
+            return true;
+        }
+    }
+
+    juce_UseDebuggingNewOperator
+};
+
+//==============================================================================
+AiffAudioFormat::AiffAudioFormat()
+    : AudioFormat (aiffFormatName, (const tchar**) aiffExtensions)
+{
+}
+
+AiffAudioFormat::~AiffAudioFormat()
+{
+}
+
+const Array <int> AiffAudioFormat::getPossibleSampleRates()
+{
+    const int rates[] = { 22050, 32000, 44100, 48000, 88200, 96000, 176400, 192000, 0 };
+    return Array <int> (rates);
+}
+
+const Array <int> AiffAudioFormat::getPossibleBitDepths()
+{
+    const int depths[] = { 8, 16, 24, 0 };
+    return Array <int> (depths);
+}
+
+bool AiffAudioFormat::canDoStereo()
+{
+    return true;
+}
+
+bool AiffAudioFormat::canDoMono()
+{
+    return true;
+}
+
+#if JUCE_MAC
+bool AiffAudioFormat::canHandleFile (const File& f)
+{
+    if (AudioFormat::canHandleFile (f))
+        return true;
+
+    const OSType type = PlatformUtilities::getTypeOfFile (f.getFullPathName());
+    return type == 'AIFF' || type == 'AIFC'
+        || type == 'aiff' || type == 'aifc';
+}
+#endif
+
+AudioFormatReader* AiffAudioFormat::createReaderFor (InputStream* sourceStream,
+                                                     const bool deleteStreamIfOpeningFails)
+{
+    AiffAudioFormatReader* w = new AiffAudioFormatReader (sourceStream);
+
+    if (w->sampleRate == 0)
+    {
+        if (! deleteStreamIfOpeningFails)
+            w->input = 0;
+
+        deleteAndZero (w);
+    }
+
+    return w;
+}
+
+AudioFormatWriter* AiffAudioFormat::createWriterFor (OutputStream* out,
+                                                     double sampleRate,
+                                                     unsigned int chans,
+                                                     int bitsPerSample,
+                                                     const StringPairArray& /*metadataValues*/,
+                                                     int /*qualityOptionIndex*/)
+{
+    if (getPossibleBitDepths().contains (bitsPerSample))
+    {
+        return new AiffAudioFormatWriter (out,
+                                          sampleRate,
+                                          chans,
+                                          bitsPerSample);
+    }
+
+    return 0;
+}
+
+END_JUCE_NAMESPACE

src/audio/audio_file_formats/juce_AiffAudioFormat.h

@@ -0,0 +1,74 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#ifndef __JUCE_AIFFAUDIOFORMAT_JUCEHEADER__
+#define __JUCE_AIFFAUDIOFORMAT_JUCEHEADER__
+
+#include "juce_AudioFormat.h"
+
+
+//==============================================================================
+/**
+    Reads and Writes AIFF format audio files.
+
+    @see AudioFormat
+*/
+class JUCE_API  AiffAudioFormat  : public AudioFormat
+{
+public:
+    //==============================================================================
+    /** Creates an format object. */
+    AiffAudioFormat();
+
+    /** Destructor. */
+    ~AiffAudioFormat();
+
+    //==============================================================================
+    const Array <int> getPossibleSampleRates();
+    const Array <int> getPossibleBitDepths();
+    bool canDoStereo();
+    bool canDoMono();
+#if JUCE_MAC
+    bool canHandleFile (const File& fileToTest);
+#endif
+
+    //==============================================================================
+    AudioFormatReader* createReaderFor (InputStream* sourceStream,
+                                        const bool deleteStreamIfOpeningFails);
+
+    AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
+                                        double sampleRateToUse,
+                                        unsigned int numberOfChannels,
+                                        int bitsPerSample,
+                                        const StringPairArray& metadataValues,
+                                        int qualityOptionIndex);
+
+
+    //==============================================================================
+    juce_UseDebuggingNewOperator
+};
+
+
+#endif   // __JUCE_AIFFAUDIOFORMAT_JUCEHEADER__

src/audio/audio_file_formats/juce_AudioCDBurner.h

@@ -0,0 +1,107 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#ifndef __JUCE_AUDIOCDBURNER_JUCEHEADER__
+#define __JUCE_AUDIOCDBURNER_JUCEHEADER__
+
+#include "juce_AudioFormatReader.h"
+#include "../audio_sources/juce_AudioSource.h"
+
+
+//==============================================================================
+/**
+*/
+class AudioCDBurner
+{
+public:
+    //==============================================================================
+    /** Returns a list of available optical drives.
+
+        Use openDevice() to open one of the items from this list.
+    */
+    static const StringArray findAvailableDevices();
+
+    /** Tries to open one of the optical drives.
+
+        The deviceIndex is an index into the array returned by findAvailableDevices().
+    */
+    static AudioCDBurner* openDevice (const int deviceIndex);
+
+    /** Destructor. */
+    ~AudioCDBurner();
+
+    //==============================================================================
+    /** Returns true if there's a writable disk in the drive.
+    */
+    bool isDiskPresent() const;
+
+    /** Returns the number of free blocks on the disk.
+
+        There are 75 blocks per second, at 44100Hz.
+    */
+    int getNumAvailableAudioBlocks() const;
+
+    /** Adds a track to be written.
+
+        The source passed-in here will be kept by this object, and it will
+        be used and deleted at some point in the future, either during the
+        burn() method or when this AudioCDBurner object is deleted. Your caller
+        method shouldn't keep a reference to it or use it again after passing
+        it in here.
+    */
+    bool addAudioTrack (AudioSource* source, int numSamples);
+
+    /**
+
+        Return true to cancel the current burn operation
+    */
+    class BurnProgressListener
+    {
+    public:
+        BurnProgressListener() throw() {}
+        virtual ~BurnProgressListener() {}
+
+        /** Called at intervals to report on the progress of the AudioCDBurner.
+
+            To cancel the burn, return true from this.
+        */
+        virtual bool audioCDBurnProgress (float proportionComplete) = 0;
+    };
+
+    const String burn (BurnProgressListener* listener,
+                       const bool ejectDiscAfterwards,
+                       const bool peformFakeBurnForTesting);
+
+    //==============================================================================
+    juce_UseDebuggingNewOperator
+
+private:
+    AudioCDBurner (const int deviceIndex);
+
+    void* internal;
+};
+
+
+#endif   // __JUCE_AUDIOCDBURNER_JUCEHEADER__

src/audio/audio_file_formats/juce_AudioCDReader.cpp

@@ -0,0 +1,245 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#include "../../core/juce_StandardHeader.h"
+
+BEGIN_JUCE_NAMESPACE
+
+#if JUCE_MAC
+
+//==============================================================================
+// Mac version doesn't need any native code because it's all done with files..
+// Windows + Linux versions are in the platform-dependent code sections.
+
+#include "juce_AudioCDReader.h"
+#include "juce_AiffAudioFormat.h"
+#include "../../io/files/juce_FileInputStream.h"
+#include "../../io/streams/juce_BufferedInputStream.h"
+
+
+static void findCDs (OwnedArray<File>& cds)
+{
+    File volumes ("/Volumes");
+    volumes.findChildFiles (cds, File::findDirectories, false);
+
+    for (int i = cds.size(); --i >= 0;)
+        if (! cds[i]->getChildFile (".TOC.plist").exists())
+            cds.remove (i);
+}
+
+const StringArray AudioCDReader::getAvailableCDNames()
+{
+    OwnedArray<File> cds;
+    findCDs (cds);
+
+    StringArray names;
+
+    for (int i = 0; i < cds.size(); ++i)
+        names.add (cds[i]->getFileName());
+
+    return names;
+}
+
+AudioCDReader* AudioCDReader::createReaderForCD (const int index)
+{
+    OwnedArray<File> cds;
+    findCDs (cds);
+
+    if (cds[index] != 0)
+        return new AudioCDReader (*cds[index]);
+    else
+        return 0;
+}
+
+AudioCDReader::AudioCDReader (const File& volume)
+   : AudioFormatReader (0, "CD Audio"),
+     volumeDir (volume),
+     currentReaderTrack (-1),
+     reader (0)
+{
+     sampleRate = 44100.0;
+     bitsPerSample = 16;
+     numChannels = 2;
+     usesFloatingPointData = false;
+
+     refreshTrackLengths();
+}
+
+AudioCDReader::~AudioCDReader()
+{
+    if (reader != 0)
+        delete reader;
+}
+
+static int getTrackNumber (const File& file)
+{
+    return file.getFileName()
+               .initialSectionContainingOnly (T("0123456789"))
+               .getIntValue();
+}
+
+int AudioCDReader::compareElements (const File* const first, const File* const second) throw()
+{
+    const int firstTrack  = getTrackNumber (*first);
+    const int secondTrack = getTrackNumber (*second);
+
+    jassert (firstTrack > 0 && secondTrack > 0);
+
+    return firstTrack - secondTrack;
+}
+
+void AudioCDReader::refreshTrackLengths()
+{
+    tracks.clear();
+    trackStartSamples.clear();
+    volumeDir.findChildFiles (tracks, File::findFiles | File::ignoreHiddenFiles, false, T("*.aiff"));
+
+    tracks.sort (*this);
+
+    AiffAudioFormat format;
+    int sample = 0;
+
+    for (int i = 0; i < tracks.size(); ++i)
+    {
+        trackStartSamples.add (sample);
+
+        FileInputStream* const in = tracks[i]->createInputStream();
+
+        if (in != 0)
+        {
+            AudioFormatReader* const r = format.createReaderFor (in, true);
+
+            if (r != 0)
+            {
+                sample += r->lengthInSamples;
+                delete r;
+            }
+        }
+    }
+
+    trackStartSamples.add (sample);
+    lengthInSamples = sample;
+}
+
+bool AudioCDReader::readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                                 int64 startSampleInFile, int numSamples)
+{
+    while (numSamples > 0)
+    {
+        int track = -1;
+
+        for (int i = 0; i < trackStartSamples.size() - 1; ++i)
+        {
+            if (startSampleInFile < trackStartSamples.getUnchecked (i + 1))
+            {
+                track = i;
+                break;
+            }
+        }
+
+        if (track < 0)
+            return false;
+
+        if (track != currentReaderTrack)
+        {
+            deleteAndZero (reader);
+
+            if (tracks [track] != 0)
+            {
+                FileInputStream* const in = tracks [track]->createInputStream();
+
+                if (in != 0)
+                {
+                    BufferedInputStream* const bin = new BufferedInputStream (in, 65536, true);
+
+                    AiffAudioFormat format;
+                    reader = format.createReaderFor (bin, true);
+
+                    if (reader == 0)
+                        currentReaderTrack = -1;
+                    else
+                        currentReaderTrack = track;
+                }
+            }
+        }
+
+        if (reader == 0)
+            return false;
+
+        const int startPos = (int) (startSampleInFile - trackStartSamples.getUnchecked (track));
+        const int numAvailable = (int) jmin ((int64) numSamples, reader->lengthInSamples - startPos);
+
+        reader->readSamples (destSamples, numDestChannels, startOffsetInDestBuffer, startPos, numAvailable);
+
+        numSamples -= numAvailable;
+        startSampleInFile += numAvailable;
+    }
+
+    return true;
+}
+
+bool AudioCDReader::isCDStillPresent() const
+{
+    return volumeDir.exists();
+}
+
+int AudioCDReader::getNumTracks() const
+{
+    return tracks.size();
+}
+
+int AudioCDReader::getPositionOfTrackStart (int trackNum) const
+{
+    return trackStartSamples [trackNum];
+}
+
+bool AudioCDReader::isTrackAudio (int trackNum) const
+{
+    return tracks [trackNum] != 0;
+}
+
+void AudioCDReader::enableIndexScanning (bool b)
+{
+    // any way to do this on a Mac??
+}
+
+int AudioCDReader::getLastIndex() const
+{
+    return 0;
+}
+
+const Array <int> AudioCDReader::findIndexesInTrack (const int trackNumber)
+{
+    return Array <int>();
+}
+
+int AudioCDReader::getCDDBId()
+{
+    return 0; //xxx
+}
+
+#endif
+
+END_JUCE_NAMESPACE

src/audio/audio_file_formats/juce_AudioCDReader.h

@@ -0,0 +1,180 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#ifndef __JUCE_AUDIOCDREADER_JUCEHEADER__
+#define __JUCE_AUDIOCDREADER_JUCEHEADER__
+
+#include "juce_AudioFormatReader.h"
+#include "../../text/juce_StringArray.h"
+#if JUCE_MAC
+#include "../../io/files/juce_File.h"
+#endif
+
+//==============================================================================
+/**
+    A type of AudioFormatReader that reads from an audio CD.
+
+    One of these can be used to read a CD as if it's one big audio stream. Use the
+    getPositionOfTrackStart() method to find where the individual tracks are
+    within the stream.
+
+    @see AudioFormatReader
+*/
+class JUCE_API  AudioCDReader  : public AudioFormatReader
+{
+public:
+    //==============================================================================
+    /** Returns a list of names of Audio CDs currently available for reading.
+
+        If there's a CD drive but no CD in it, this might return an empty list, or
+        possibly a device that can be opened but which has no tracks, depending
+        on the platform.
+
+        @see createReaderForCD
+    */
+    static const StringArray getAvailableCDNames();
+
+    /** Tries to create an AudioFormatReader that can read from an Audio CD.
+
+        @param index    the index of one of the available CDs - use getAvailableCDNames()
+                        to find out how many there are.
+        @returns        a new AudioCDReader object, or 0 if it couldn't be created. The
+                        caller will be responsible for deleting the object returned.
+    */
+    static AudioCDReader* createReaderForCD (const int index);
+
+    //==============================================================================
+    /** Destructor. */
+    ~AudioCDReader();
+
+    /** Implementation of the AudioFormatReader method. */
+    bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                      int64 startSampleInFile, int numSamples);
+
+    /** Checks whether the CD has been removed from the drive.
+    */
+    bool isCDStillPresent() const;
+
+    /** Returns the total number of tracks (audio + data).
+    */
+    int getNumTracks() const;
+
+    /** Finds the sample offset of the start of a track.
+
+        @param trackNum     the track number, where 0 is the first track.
+    */
+    int getPositionOfTrackStart (int trackNum) const;
+
+    /** Returns true if a given track is an audio track.
+
+        @param trackNum     the track number, where 0 is the first track.
+    */
+    bool isTrackAudio (int trackNum) const;
+
+    /** Refreshes the object's table of contents.
+
+        If the disc has been ejected and a different one put in since this
+        object was created, this will cause it to update its idea of how many tracks
+        there are, etc.
+    */
+    void refreshTrackLengths();
+
+    /** Enables scanning for indexes within tracks.
+
+        @see getLastIndex
+    */
+    void enableIndexScanning (bool enabled);
+
+    /** Returns the index number found during the last read() call.
+
+        Index scanning is turned off by default - turn it on with enableIndexScanning().
+
+        Then when the read() method is called, if it comes across an index within that
+        block, the index number is stored and returned by this method.
+
+        Some devices might not support indexes, of course.
+
+        (If you don't know what CD indexes are, it's unlikely you'll ever need them).
+
+        @see enableIndexScanning
+    */
+    int getLastIndex() const;
+
+    /** Scans a track to find the position of any indexes within it.
+
+        @param trackNumber  the track to look in, where 0 is the first track on the disc
+        @returns    an array of sample positions of any index points found (not including
+                    the index that marks the start of the track)
+    */
+    const Array <int> findIndexesInTrack (const int trackNumber);
+
+    /** Returns the CDDB id number for the CD.
+
+        It's not a great way of identifying a disc, but it's traditional.
+    */
+    int getCDDBId();
+
+    /** Tries to eject the disk.
+
+        Of course this might not be possible, if some other process is using it.
+    */
+    void ejectDisk();
+
+    //==============================================================================
+    juce_UseDebuggingNewOperator
+
+private:
+
+#if JUCE_MAC
+    File volumeDir;
+    OwnedArray<File> tracks;
+    Array <int> trackStartSamples;
+    int currentReaderTrack;
+    AudioFormatReader* reader;
+    AudioCDReader (const File& volume);
+public:
+    static int compareElements (const File* const, const File* const) throw();
+private:
+
+#elif JUCE_WIN32
+    int numTracks;
+    int trackStarts[100];
+    bool audioTracks [100];
+    void* handle;
+    bool indexingEnabled;
+    int lastIndex, firstFrameInBuffer, samplesInBuffer;
+    MemoryBlock buffer;
+    AudioCDReader (void* handle);
+    int getIndexAt (int samplePos);
+
+#elif JUCE_LINUX
+    AudioCDReader();
+#endif
+
+    AudioCDReader (const AudioCDReader&);
+    const AudioCDReader& operator= (const AudioCDReader&);
+};
+
+#endif   // __JUCE_AUDIOCDREADER_JUCEHEADER__

src/audio/audio_file_formats/juce_AudioFormat.cpp

@@ -0,0 +1,557 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#include "../../core/juce_StandardHeader.h"
+
+BEGIN_JUCE_NAMESPACE
+
+#include "juce_AudioFormat.h"
+#include "../dsp/juce_AudioSampleBuffer.h"
+
+
+//==============================================================================
+AudioFormatReader::AudioFormatReader (InputStream* const in,
+                                      const String& formatName_)
+    : sampleRate (0),
+      bitsPerSample (0),
+      lengthInSamples (0),
+      numChannels (0),
+      usesFloatingPointData (false),
+      input (in),
+      formatName (formatName_)
+{
+}
+
+AudioFormatReader::~AudioFormatReader()
+{
+    delete input;
+}
+
+bool AudioFormatReader::read (int** destSamples,
+                              int numDestChannels,
+                              int64 startSampleInSource,
+                              int numSamplesToRead,
+                              const bool fillLeftoverChannelsWithCopies)
+{
+    jassert (numDestChannels > 0); // you have to actually give this some channels to work with!
+
+    int startOffsetInDestBuffer = 0;
+
+    if (startSampleInSource < 0)
+    {
+        const int silence = (int) jmin (-startSampleInSource, (int64) numSamplesToRead);
+
+        for (int i = numDestChannels; --i >= 0;)
+            if (destSamples[i] != 0)
+                zeromem (destSamples[i], sizeof (int) * silence);
+
+        startOffsetInDestBuffer += silence;
+        numSamplesToRead -= silence;
+        startSampleInSource = 0;
+    }
+
+    if (numSamplesToRead <= 0)
+        return true;
+
+    if (! readSamples (destSamples, jmin (numChannels, numDestChannels), startOffsetInDestBuffer,
+                       startSampleInSource, numSamplesToRead))
+        return false;
+
+    if (numDestChannels > (int) numChannels)
+    {
+        if (fillLeftoverChannelsWithCopies)
+        {
+            int* lastFullChannel = destSamples[0];
+
+            for (int i = numDestChannels; --i > 0;)
+            {
+                if (destSamples[i] != 0)
+                {
+                    lastFullChannel = destSamples[i];
+                    break;
+                }
+            }
+
+            if (lastFullChannel != 0)
+                for (int i = numChannels; i < numDestChannels; ++i)
+                    if (destSamples[i] != 0)
+                        memcpy (destSamples[i], lastFullChannel, sizeof (int) * numSamplesToRead);
+        }
+        else
+        {
+            for (int i = numChannels; i < numDestChannels; ++i)
+                if (destSamples[i] != 0)
+                    zeromem (destSamples[i], sizeof (int) * numSamplesToRead);
+        }
+    }
+
+    return true;
+}
+
+static void findMaxMin (const float* src, const int num,
+                        float& maxVal, float& minVal)
+{
+    float mn = src[0];
+    float mx = mn;
+
+    for (int i = 1; i < num; ++i)
+    {
+        const float s = src[i];
+        if (s > mx)
+            mx = s;
+        if (s < mn)
+            mn = s;
+    }
+
+    maxVal = mx;
+    minVal = mn;
+}
+
+void AudioFormatReader::readMaxLevels (int64 startSampleInFile,
+                                       int64 numSamples,
+                                       float& lowestLeft, float& highestLeft,
+                                       float& lowestRight, float& highestRight)
+{
+    if (numSamples <= 0)
+    {
+        lowestLeft = 0;
+        lowestRight = 0;
+        highestLeft = 0;
+        highestRight = 0;
+        return;
+    }
+
+    const int bufferSize = (int) jmin (numSamples, (int64) 4096);
+    MemoryBlock tempSpace (bufferSize * sizeof (int) * 2 + 64);
+
+    int* tempBuffer[3];
+    tempBuffer[0] = (int*) tempSpace.getData();
+    tempBuffer[1] = ((int*) tempSpace.getData()) + bufferSize;
+    tempBuffer[2] = 0;
+
+    if (usesFloatingPointData)
+    {
+        float lmin = 1.0e6;
+        float lmax = -lmin;
+        float rmin = lmin;
+        float rmax = lmax;
+
+        while (numSamples > 0)
+        {
+            const int numToDo = (int) jmin (numSamples, (int64) bufferSize);
+            read ((int**) tempBuffer, 2, startSampleInFile, numToDo, false);
+
+            numSamples -= numToDo;
+
+            float bufmin, bufmax;
+            findMaxMin ((float*) tempBuffer[0], numToDo, bufmax, bufmin);
+            lmin = jmin (lmin, bufmin);
+            lmax = jmax (lmax, bufmax);
+
+            if (numChannels > 1)
+            {
+                findMaxMin ((float*) tempBuffer[1], numToDo, bufmax, bufmin);
+                rmin = jmin (rmin, bufmin);
+                rmax = jmax (rmax, bufmax);
+            }
+        }
+
+        if (numChannels <= 1)
+        {
+            rmax = lmax;
+            rmin = lmin;
+        }
+
+        lowestLeft = lmin;
+        highestLeft = lmax;
+        lowestRight = rmin;
+        highestRight = rmax;
+    }
+    else
+    {
+        int lmax = INT_MIN;
+        int lmin = INT_MAX;
+        int rmax = INT_MIN;
+        int rmin = INT_MAX;
+
+        while (numSamples > 0)
+        {
+            const int numToDo = (int) jmin (numSamples, (int64) bufferSize);
+            read ((int**) tempBuffer, 2, startSampleInFile, numToDo, false);
+
+            numSamples -= numToDo;
+
+            for (int j = numChannels; --j >= 0;)
+            {
+                int bufMax = INT_MIN;
+                int bufMin = INT_MAX;
+
+                const int* const b = tempBuffer[j];
+
+                for (int i = 0; i < numToDo; ++i)
+                {
+                    const int samp = b[i];
+
+                    if (samp < bufMin)
+                        bufMin = samp;
+
+                    if (samp > bufMax)
+                        bufMax = samp;
+                }
+
+                if (j == 0)
+                {
+                    lmax = jmax (lmax, bufMax);
+                    lmin = jmin (lmin, bufMin);
+                }
+                else
+                {
+                    rmax = jmax (rmax, bufMax);
+                    rmin = jmin (rmin, bufMin);
+                }
+            }
+        }
+
+        if (numChannels <= 1)
+        {
+            rmax = lmax;
+            rmin = lmin;
+        }
+
+        lowestLeft = lmin / (float)INT_MAX;
+        highestLeft = lmax / (float)INT_MAX;
+        lowestRight = rmin / (float)INT_MAX;
+        highestRight = rmax / (float)INT_MAX;
+    }
+}
+
+int64 AudioFormatReader::searchForLevel (int64 startSample,
+                                         int64 numSamplesToSearch,
+                                         const double magnitudeRangeMinimum,
+                                         const double magnitudeRangeMaximum,
+                                         const int minimumConsecutiveSamples)
+{
+    if (numSamplesToSearch == 0)
+        return -1;
+
+    const int bufferSize = 4096;
+    MemoryBlock tempSpace (bufferSize * sizeof (int) * 2 + 64);
+
+    int* tempBuffer[3];
+    tempBuffer[0] = (int*) tempSpace.getData();
+    tempBuffer[1] = ((int*) tempSpace.getData()) + bufferSize;
+    tempBuffer[2] = 0;
+
+    int consecutive = 0;
+    int64 firstMatchPos = -1;
+
+    jassert (magnitudeRangeMaximum > magnitudeRangeMinimum);
+
+    const double doubleMin = jlimit (0.0, (double) INT_MAX, magnitudeRangeMinimum * INT_MAX);
+    const double doubleMax = jlimit (doubleMin, (double) INT_MAX, magnitudeRangeMaximum * INT_MAX);
+    const int intMagnitudeRangeMinimum = roundDoubleToInt (doubleMin);
+    const int intMagnitudeRangeMaximum = roundDoubleToInt (doubleMax);
+
+    while (numSamplesToSearch != 0)
+    {
+        const int numThisTime = (int) jmin (abs64 (numSamplesToSearch), (int64) bufferSize);
+        int64 bufferStart = startSample;
+
+        if (numSamplesToSearch < 0)
+            bufferStart -= numThisTime;
+
+        if (bufferStart >= (int) lengthInSamples)
+            break;
+
+        read ((int**) tempBuffer, 2, bufferStart, numThisTime, false);
+
+        int num = numThisTime;
+        while (--num >= 0)
+        {
+            if (numSamplesToSearch < 0)
+                --startSample;
+
+            bool matches = false;
+            const int index = (int) (startSample - bufferStart);
+
+            if (usesFloatingPointData)
+            {
+                const float sample1 = fabsf (((float*) tempBuffer[0]) [index]);
+
+                if (sample1 >= magnitudeRangeMinimum
+                     && sample1 <= magnitudeRangeMaximum)
+                {
+                    matches = true;
+                }
+                else if (numChannels > 1)
+                {
+                    const float sample2 = fabsf (((float*) tempBuffer[1]) [index]);
+
+                    matches = (sample2 >= magnitudeRangeMinimum
+                                 && sample2 <= magnitudeRangeMaximum);
+                }
+            }
+            else
+            {
+                const int sample1 = abs (tempBuffer[0] [index]);
+
+                if (sample1 >= intMagnitudeRangeMinimum
+                     && sample1 <= intMagnitudeRangeMaximum)
+                {
+                    matches = true;
+                }
+                else if (numChannels > 1)
+                {
+                    const int sample2 = abs (tempBuffer[1][index]);
+
+                    matches = (sample2 >= intMagnitudeRangeMinimum
+                                 && sample2 <= intMagnitudeRangeMaximum);
+                }
+            }
+
+            if (matches)
+            {
+                if (firstMatchPos < 0)
+                    firstMatchPos = startSample;
+
+                if (++consecutive >= minimumConsecutiveSamples)
+                {
+                    if (firstMatchPos < 0 || firstMatchPos >= lengthInSamples)
+                        return -1;
+
+                    return firstMatchPos;
+                }
+            }
+            else
+            {
+                consecutive = 0;
+                firstMatchPos = -1;
+            }
+
+            if (numSamplesToSearch > 0)
+                ++startSample;
+        }
+
+        if (numSamplesToSearch > 0)
+            numSamplesToSearch -= numThisTime;
+        else
+            numSamplesToSearch += numThisTime;
+    }
+
+    return -1;
+}
+
+//==============================================================================
+AudioFormatWriter::AudioFormatWriter (OutputStream* const out,
+                                      const String& formatName_,
+                                      const double rate,
+                                      const unsigned int numChannels_,
+                                      const unsigned int bitsPerSample_)
+  : sampleRate (rate),
+    numChannels (numChannels_),
+    bitsPerSample (bitsPerSample_),
+    usesFloatingPointData (false),
+    output (out),
+    formatName (formatName_)
+{
+}
+
+AudioFormatWriter::~AudioFormatWriter()
+{
+    delete output;
+}
+
+bool AudioFormatWriter::writeFromAudioReader (AudioFormatReader& reader,
+                                              int64 startSample,
+                                              int64 numSamplesToRead)
+{
+    const int bufferSize = 16384;
+    const int maxChans = 128;
+    AudioSampleBuffer tempBuffer (reader.numChannels, bufferSize);
+    int* buffers [maxChans];
+
+    for (int i = maxChans; --i >= 0;)
+        buffers[i] = 0;
+
+    if (numSamplesToRead < 0)
+        numSamplesToRead = reader.lengthInSamples;
+
+    while (numSamplesToRead > 0)
+    {
+        const int numToDo = (int) jmin (numSamplesToRead, (int64) bufferSize);
+
+        for (int i = tempBuffer.getNumChannels(); --i >= 0;)
+            buffers[i] = (int*) tempBuffer.getSampleData (i, 0);
+
+        if (! reader.read (buffers, maxChans, startSample, numToDo, false))
+            return false;
+
+        if (reader.usesFloatingPointData != isFloatingPoint())
+        {
+            int** bufferChan = buffers;
+
+            while (*bufferChan != 0)
+            {
+                int* b = *bufferChan++;
+
+                if (isFloatingPoint())
+                {
+                    // int -> float
+                    const double factor = 1.0 / INT_MAX;
+
+                    for (int i = 0; i < numToDo; ++i)
+                        ((float*)b)[i] = (float) (factor * b[i]);
+                }
+                else
+                {
+                    // float -> int
+                    for (int i = 0; i < numToDo; ++i)
+                    {
+                        const double samp = *(const float*) b;
+
+                        if (samp <= -1.0)
+                            *b++ = INT_MIN;
+                        else if (samp >= 1.0)
+                            *b++ = INT_MAX;
+                        else
+                            *b++ = roundDoubleToInt (INT_MAX * samp);
+                    }
+                }
+            }
+        }
+
+        if (! write ((const int**) buffers, numToDo))
+            return false;
+
+        numSamplesToRead -= numToDo;
+        startSample += numToDo;
+    }
+
+    return true;
+}
+
+bool AudioFormatWriter::writeFromAudioSource (AudioSource& source,
+                                              int numSamplesToRead,
+                                              const int samplesPerBlock)
+{
+    const int maxChans = 128;
+    AudioSampleBuffer tempBuffer (getNumChannels(), samplesPerBlock);
+    int* buffers [maxChans];
+
+    while (numSamplesToRead > 0)
+    {
+        const int numToDo = jmin (numSamplesToRead, samplesPerBlock);
+
+        AudioSourceChannelInfo info;
+        info.buffer = &tempBuffer;
+        info.startSample = 0;
+        info.numSamples = numToDo;
+        info.clearActiveBufferRegion();
+
+        source.getNextAudioBlock (info);
+
+        int i;
+        for (i = maxChans; --i >= 0;)
+            buffers[i] = 0;
+
+        for (i = tempBuffer.getNumChannels(); --i >= 0;)
+            buffers[i] = (int*) tempBuffer.getSampleData (i, 0);
+
+        if (! isFloatingPoint())
+        {
+            int** bufferChan = buffers;
+
+            while (*bufferChan != 0)
+            {
+                int* b = *bufferChan++;
+
+                // float -> int
+                for (int j = numToDo; --j >= 0;)
+                {
+                    const double samp = *(const float*) b;
+
+                    if (samp <= -1.0)
+                        *b++ = INT_MIN;
+                    else if (samp >= 1.0)
+                        *b++ = INT_MAX;
+                    else
+                        *b++ = roundDoubleToInt (INT_MAX * samp);
+                }
+            }
+        }
+
+        if (! write ((const int**) buffers, numToDo))
+            return false;
+
+        numSamplesToRead -= numToDo;
+    }
+
+    return true;
+}
+
+//==============================================================================
+AudioFormat::AudioFormat (const String& name,
+                          const tchar** const extensions)
+  : formatName (name),
+    fileExtensions (extensions)
+{
+}
+
+AudioFormat::~AudioFormat()
+{
+}
+
+//==============================================================================
+const String& AudioFormat::getFormatName() const
+{
+    return formatName;
+}
+
+const StringArray& AudioFormat::getFileExtensions() const
+{
+    return fileExtensions;
+}
+
+bool AudioFormat::canHandleFile (const File& f)
+{
+    for (int i = 0; i < fileExtensions.size(); ++i)
+        if (f.hasFileExtension (fileExtensions[i]))
+            return true;
+
+    return false;
+}
+
+bool AudioFormat::isCompressed()
+{
+    return false;
+}
+
+const StringArray AudioFormat::getQualityOptions()
+{
+    return StringArray();
+}
+
+
+END_JUCE_NAMESPACE

src/audio/audio_file_formats/juce_AudioFormat.h

@@ -0,0 +1,172 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#ifndef __JUCE_AUDIOFORMAT_JUCEHEADER__
+#define __JUCE_AUDIOFORMAT_JUCEHEADER__
+
+#include "juce_AudioFormatReader.h"
+#include "juce_AudioFormatWriter.h"
+
+
+//==============================================================================
+/**
+    Subclasses of AudioFormat are used to read and write different audio
+    file formats.
+
+    @see AudioFormatReader, AudioFormatWriter, WavAudioFormat, AiffAudioFormat
+*/
+class JUCE_API  AudioFormat
+{
+public:
+    //==============================================================================
+    /** Destructor. */
+    virtual ~AudioFormat();
+
+    //==============================================================================
+    /** Returns the name of this format.
+
+        e.g. "WAV file" or "AIFF file"
+    */
+    const String& getFormatName() const;
+
+    /** Returns all the file extensions that might apply to a file of this format.
+
+        The first item will be the one that's preferred when creating a new file.
+
+        So for a wav file this might just return ".wav"; for an AIFF file it might
+        return two items, ".aif" and ".aiff"
+    */
+    const StringArray& getFileExtensions() const;
+
+    //==============================================================================
+    /** Returns true if this the given file can be read by this format.
+
+        Subclasses shouldn't do too much work here, just check the extension or
+        file type. The base class implementation just checks the file's extension
+        against one of the ones that was registered in the constructor.
+    */
+    virtual bool canHandleFile (const File& fileToTest);
+
+    /** Returns a set of sample rates that the format can read and write. */
+    virtual const Array <int> getPossibleSampleRates() = 0;
+
+    /** Returns a set of bit depths that the format can read and write. */
+    virtual const Array <int> getPossibleBitDepths() = 0;
+
+    /** Returns true if the format can do 2-channel audio. */
+    virtual bool canDoStereo() = 0;
+
+    /** Returns true if the format can do 1-channel audio. */
+    virtual bool canDoMono() = 0;
+
+    /** Returns true if the format uses compressed data. */
+    virtual bool isCompressed();
+
+    /** Returns a list of different qualities that can be used when writing.
+
+        Non-compressed formats will just return an empty array, but for something
+        like Ogg-Vorbis or MP3, it might return a list of bit-rates, etc.
+
+        When calling createWriterFor(), an index from this array is passed in to
+        tell the format which option is required.
+    */
+    virtual const StringArray getQualityOptions();
+
+    //==============================================================================
+    /** Tries to create an object that can read from a stream containing audio
+        data in this format.
+
+        The reader object that is returned can be used to read from the stream, and
+        should then be deleted by the caller.
+
+        @param sourceStream                 the stream to read from - the AudioFormatReader object
+                                            that is returned will delete this stream when it no longer
+                                            needs it.
+        @param deleteStreamIfOpeningFails   if no reader can be created, this determines whether this method
+                                            should delete the stream object that was passed-in. (If a valid
+                                            reader is returned, it will always be in charge of deleting the
+                                            stream, so this parameter is ignored)
+        @see AudioFormatReader
+    */
+    virtual AudioFormatReader* createReaderFor (InputStream* sourceStream,
+                                                const bool deleteStreamIfOpeningFails) = 0;
+
+    /** Tries to create an object that can write to a stream with this audio format.
+
+        The writer object that is returned can be used to write to the stream, and
+        should then be deleted by the caller.
+
+        If the stream can't be created for some reason (e.g. the parameters passed in
+        here aren't suitable), this will return 0.
+
+        @param streamToWriteTo      the stream that the data will go to - this will be
+                                    deleted by the AudioFormatWriter object when it's no longer
+                                    needed. If no AudioFormatWriter can be created by this method,
+                                    the stream will NOT be deleted, so that the caller can re-use it
+                                    to try to open a different format, etc
+        @param sampleRateToUse      the sample rate for the file, which must be one of the ones
+                                    returned by getPossibleSampleRates()
+        @param numberOfChannels     the number of channels - this must be either 1 or 2, and
+                                    the choice will depend on the results of canDoMono() and
+                                    canDoStereo()
+        @param bitsPerSample        the bits per sample to use - this must be one of the values
+                                    returned by getPossibleBitDepths()
+        @param metadataValues       a set of metadata values that the writer should try to write
+                                    to the stream. Exactly what these are depends on the format,
+                                    and the subclass doesn't actually have to do anything with
+                                    them if it doesn't want to. Have a look at the specific format
+                                    implementation classes to see possible values that can be
+                                    used
+        @param qualityOptionIndex   the index of one of compression qualities returned by the
+                                    getQualityOptions() method. If there aren't any quality options
+                                    for this format, just pass 0 in this parameter, as it'll be
+                                    ignored
+        @see AudioFormatWriter
+    */
+    virtual AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
+                                                double sampleRateToUse,
+                                                unsigned int numberOfChannels,
+                                                int bitsPerSample,
+                                                const StringPairArray& metadataValues,
+                                                int qualityOptionIndex) = 0;
+
+protected:
+    /** Creates an AudioFormat object.
+
+        @param formatName       this sets the value that will be returned by getFormatName()
+        @param fileExtensions   a zero-terminated list of file extensions - this is what will
+                                be returned by getFileExtension()
+    */
+    AudioFormat (const String& formatName,
+                 const tchar** const fileExtensions);
+
+private:
+    //==============================================================================
+    String formatName;
+    StringArray fileExtensions;
+};
+
+
+#endif   // __JUCE_AUDIOFORMAT_JUCEHEADER__

src/audio/audio_file_formats/juce_AudioFormatManager.cpp

@@ -0,0 +1,225 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#include "../../core/juce_StandardHeader.h"
+
+BEGIN_JUCE_NAMESPACE
+
+#include "juce_AudioFormatManager.h"
+#include "juce_AiffAudioFormat.h"
+#include "juce_WavAudioFormat.h"
+#include "juce_FlacAudioFormat.h"
+#include "juce_OggVorbisAudioFormat.h"
+#include "../../io/files/juce_FileInputStream.h"
+
+
+//==============================================================================
+AudioFormatManager::AudioFormatManager()
+    : knownFormats (4),
+      defaultFormatIndex (0)
+{
+}
+
+AudioFormatManager::~AudioFormatManager()
+{
+    clearFormats();
+    clearSingletonInstance();
+}
+
+juce_ImplementSingleton (AudioFormatManager);
+
+
+//==============================================================================
+void AudioFormatManager::registerFormat (AudioFormat* newFormat,
+                                         const bool makeThisTheDefaultFormat)
+{
+    jassert (newFormat != 0);
+
+    if (newFormat != 0)
+    {
+#ifdef JUCE_DEBUG
+        for (int i = getNumKnownFormats(); --i >= 0;)
+        {
+            if (getKnownFormat (i)->getFormatName() == newFormat->getFormatName())
+            {
+                jassertfalse // trying to add the same format twice!
+            }
+        }
+#endif
+
+        if (makeThisTheDefaultFormat)
+            defaultFormatIndex = knownFormats.size();
+
+        knownFormats.add (newFormat);
+    }
+}
+
+void AudioFormatManager::registerBasicFormats()
+{
+#if JUCE_MAC
+    registerFormat (new AiffAudioFormat(), true);
+    registerFormat (new WavAudioFormat(), false);
+#else
+    registerFormat (new WavAudioFormat(), true);
+    registerFormat (new AiffAudioFormat(), false);
+#endif
+
+#if JUCE_USE_FLAC
+    registerFormat (new FlacAudioFormat(), false);
+#endif
+
+#if JUCE_USE_OGGVORBIS
+    registerFormat (new OggVorbisAudioFormat(), false);
+#endif
+}
+
+void AudioFormatManager::clearFormats()
+{
+    for (int i = getNumKnownFormats(); --i >= 0;)
+    {
+        AudioFormat* const af = getKnownFormat(i);
+        delete af;
+    }
+
+    knownFormats.clear();
+    defaultFormatIndex = 0;
+}
+
+int AudioFormatManager::getNumKnownFormats() const
+{
+    return knownFormats.size();
+}
+
+AudioFormat* AudioFormatManager::getKnownFormat (const int index) const
+{
+    return (AudioFormat*) knownFormats [index];
+}
+
+AudioFormat* AudioFormatManager::getDefaultFormat() const
+{
+    return getKnownFormat (defaultFormatIndex);
+}
+
+AudioFormat* AudioFormatManager::findFormatForFileExtension (const String& fileExtension) const
+{
+    String e (fileExtension);
+    if (! e.startsWithChar (T('.')))
+        e = T(".") + e;
+
+    for (int i = 0; i < getNumKnownFormats(); ++i)
+        if (getKnownFormat(i)->getFileExtensions().contains (e, true))
+            return getKnownFormat(i);
+
+    return 0;
+}
+
+const String AudioFormatManager::getWildcardForAllFormats() const
+{
+    StringArray allExtensions;
+
+    int i;
+    for (i = 0; i < getNumKnownFormats(); ++i)
+        allExtensions.addArray (getKnownFormat (i)->getFileExtensions());
+
+    allExtensions.trim();
+    allExtensions.removeEmptyStrings();
+
+    String s;
+    for (i = 0; i < allExtensions.size(); ++i)
+    {
+        s << T('*');
+
+        if (! allExtensions[i].startsWithChar (T('.')))
+            s << T('.');
+
+        s << allExtensions[i];
+
+        if (i < allExtensions.size() - 1)
+            s << T(';');
+    }
+
+    return s;
+}
+
+//==============================================================================
+AudioFormatReader* AudioFormatManager::createReaderFor (const File& file)
+{
+    // you need to actually register some formats before the manager can
+    // use them to open a file!
+    jassert (knownFormats.size() > 0);
+
+    for (int i = 0; i < getNumKnownFormats(); ++i)
+    {
+        AudioFormat* const af = getKnownFormat(i);
+
+        if (af->canHandleFile (file))
+        {
+            InputStream* const in = file.createInputStream();
+
+            if (in != 0)
+            {
+                AudioFormatReader* const r = af->createReaderFor (in, true);
+
+                if (r != 0)
+                    return r;
+            }
+        }
+    }
+
+    return 0;
+}
+
+AudioFormatReader* AudioFormatManager::createReaderFor (InputStream* in)
+{
+    // you need to actually register some formats before the manager can
+    // use them to open a file!
+    jassert (knownFormats.size() > 0);
+
+    if (in != 0)
+    {
+        const int64 originalStreamPos = in->getPosition();
+
+        for (int i = 0; i < getNumKnownFormats(); ++i)
+        {
+            AudioFormatReader* const r = getKnownFormat(i)->createReaderFor (in, false);
+
+            if (r != 0)
+                return r;
+
+            in->setPosition (originalStreamPos);
+
+            // the stream that is passed-in must be capable of being repositioned so
+            // that all the formats can have a go at opening it.
+            jassert (in->getPosition() == originalStreamPos);
+        }
+
+        delete in;
+    }
+
+    return 0;
+}
+
+
+END_JUCE_NAMESPACE

src/audio/audio_file_formats/juce_AudioFormatManager.h

@@ -0,0 +1,144 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#ifndef __JUCE_AUDIOFORMATMANAGER_JUCEHEADER__
+#define __JUCE_AUDIOFORMATMANAGER_JUCEHEADER__
+
+#include "juce_AudioFormat.h"
+#include "../../core/juce_Singleton.h"
+
+
+//==============================================================================
+/**
+    A class for keeping a list of available audio formats, and for deciding which
+    one to use to open a given file.
+
+    You can either use this class as a singleton object, or create instances of it
+    yourself. Once created, use its registerFormat() method to tell it which
+    formats it should use.
+
+    @see AudioFormat
+*/
+class JUCE_API  AudioFormatManager
+{
+public:
+    //==============================================================================
+    /** Creates an empty format manager.
+
+        Before it'll be any use, you'll need to call registerFormat() with all the
+        formats you want it to be able to recognise.
+    */
+    AudioFormatManager();
+
+    /** Destructor. */
+    ~AudioFormatManager();
+
+    juce_DeclareSingleton (AudioFormatManager, false);
+
+    //==============================================================================
+    /** Adds a format to the manager's list of available file types.
+
+        The object passed-in will be deleted by this object, so don't keep a pointer
+        to it!
+
+        If makeThisTheDefaultFormat is true, then the getDefaultFormat() method will
+        return this one when called.
+    */
+    void registerFormat (AudioFormat* newFormat,
+                         const bool makeThisTheDefaultFormat);
+
+    /** Handy method to make it easy to register the formats that come with Juce.
+
+        Currently, this will add WAV and AIFF to the list.
+    */
+    void registerBasicFormats();
+
+    /** Clears the list of known formats. */
+    void clearFormats();
+
+    /** Returns the number of currently registered file formats. */
+    int getNumKnownFormats() const;
+
+    /** Returns one of the registered file formats. */
+    AudioFormat* getKnownFormat (const int index) const;
+
+    /** Looks for which of the known formats is listed as being for a given file
+        extension.
+
+        The extension may have a dot before it, so e.g. ".wav" or "wav" are both ok.
+    */
+    AudioFormat* findFormatForFileExtension (const String& fileExtension) const;
+
+    /** Returns the format which has been set as the default one.
+
+        You can set a format as being the default when it is registered. It's useful
+        when you want to write to a file, because the best format may change between
+        platforms, e.g. AIFF is preferred on the Mac, WAV on Windows.
+
+        If none has been set as the default, this method will just return the first
+        one in the list.
+    */
+    AudioFormat* getDefaultFormat() const;
+
+    /** Returns a set of wildcards for file-matching that contains the extensions for
+        all known formats.
+
+        E.g. if might return "*.wav;*.aiff" if it just knows about wavs and aiffs.
+    */
+    const String getWildcardForAllFormats() const;
+
+    //==============================================================================
+    /** Searches through the known formats to try to create a suitable reader for
+        this file.
+
+        If none of the registered formats can open the file, it'll return 0. If it
+        returns a reader, it's the caller's responsibility to delete the reader.
+    */
+    AudioFormatReader* createReaderFor (const File& audioFile);
+
+    /** Searches through the known formats to try to create a suitable reader for
+        this stream.
+
+        The stream object that is passed-in will be deleted by this method or by the
+        reader that is returned, so the caller should not keep any references to it.
+
+        The stream that is passed-in must be capable of being repositioned so
+        that all the formats can have a go at opening it.
+
+        If none of the registered formats can open the stream, it'll return 0. If it
+        returns a reader, it's the caller's responsibility to delete the reader.
+    */
+    AudioFormatReader* createReaderFor (InputStream* audioFileStream);
+
+    //==============================================================================
+    juce_UseDebuggingNewOperator
+
+private:
+    VoidArray knownFormats;
+    int defaultFormatIndex;
+};
+
+
+#endif   // __JUCE_AUDIOFORMATMANAGER_JUCEHEADER__

src/audio/audio_file_formats/juce_AudioFormatReader.h

@@ -0,0 +1,229 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#ifndef __JUCE_AUDIOFORMATREADER_JUCEHEADER__
+#define __JUCE_AUDIOFORMATREADER_JUCEHEADER__
+
+#include "../../io/streams/juce_InputStream.h"
+#include "../../text/juce_StringPairArray.h"
+class AudioFormat;
+
+
+//==============================================================================
+/**
+    Reads samples from an audio file stream.
+
+    A subclass that reads a specific type of audio format will be created by
+    an AudioFormat object.
+
+    @see AudioFormat, AudioFormatWriter
+*/
+class JUCE_API  AudioFormatReader
+{
+protected:
+    //==============================================================================
+    /** Creates an AudioFormatReader object.
+
+        @param sourceStream     the stream to read from - this will be deleted
+                                by this object when it is no longer needed. (Some
+                                specialised readers might not use this parameter and
+                                can leave it as 0).
+        @param formatName       the description that will be returned by the getFormatName()
+                                method
+    */
+    AudioFormatReader (InputStream* const sourceStream,
+                       const String& formatName);
+
+public:
+    /** Destructor. */
+    virtual ~AudioFormatReader();
+
+    //==============================================================================
+    /** Returns a description of what type of format this is.
+
+        E.g. "AIFF"
+    */
+    const String getFormatName() const throw()      { return formatName; }
+
+    //==============================================================================
+    /** Reads samples from the stream.
+
+        @param destSamples          an array of buffers into which the sample data for each
+                                    channel will be written.
+                                    If the format is fixed-point, each channel will be written
+                                    as an array of 32-bit signed integers using the full
+                                    range -0x80000000 to 0x7fffffff, regardless of the source's
+                                    bit-depth. If it is a floating-point format, you should cast
+                                    the resulting array to a (float**) to get the values (in the
+                                    range -1.0 to 1.0 or beyond)
+                                    If the format is stereo, then destSamples[0] is the left channel
+                                    data, and destSamples[1] is the right channel.
+                                    The numDestChannels parameter indicates how many pointers this array
+                                    contains, but some of these pointers can be null if you don't want to
+                                    read data for some of the channels
+        @param numDestChannels      the number of array elements in the destChannels array
+        @param startSampleInSource  the position in the audio file or stream at which the samples
+                                    should be read, as a number of samples from the start of the
+                                    stream. It's ok for this to be beyond the start or end of the
+                                    available data - any samples that are out-of-range will be returned
+                                    as zeros.
+        @param numSamplesToRead     the number of samples to read. If this is greater than the number
+                                    of samples that the file or stream contains. the result will be padded
+                                    with zeros
+        @param fillLeftoverChannelsWithCopies   if true, this indicates that if there's no source data available
+                                    for some of the channels that you pass in, then they should be filled with
+                                    copies of valid source channels.
+                                    E.g. if you're reading a mono file and you pass 2 channels to this method, then
+                                    if fillLeftoverChannelsWithCopies is true, both destination channels will be filled
+                                    with the same data from the file's single channel. If fillLeftoverChannelsWithCopies
+                                    was false, then only the first channel would be filled with the file's contents, and
+                                    the second would be cleared. If there are many channels, e.g. you try to read 4 channels
+                                    from a stereo file, then the last 3 would all end up with copies of the same data.
+        @returns                    true if the operation succeeded, false if there was an error. Note
+                                    that reading sections of data beyond the extent of the stream isn't an
+                                    error - the reader should just return zeros for these regions
+        @see readMaxLevels
+    */
+    bool read (int** destSamples,
+               int numDestChannels,
+               int64 startSampleInSource,
+               int numSamplesToRead,
+               const bool fillLeftoverChannelsWithCopies);
+
+    /** Finds the highest and lowest sample levels from a section of the audio stream.
+
+        This will read a block of samples from the stream, and measure the
+        highest and lowest sample levels from the channels in that section, returning
+        these as normalised floating-point levels.
+
+        @param startSample          the offset into the audio stream to start reading from. It's
+                                    ok for this to be beyond the start or end of the stream.
+        @param numSamples           how many samples to read
+        @param lowestLeft           on return, this is the lowest absolute sample from the left channel
+        @param highestLeft          on return, this is the highest absolute sample from the left channel
+        @param lowestRight          on return, this is the lowest absolute sample from the right
+                                    channel (if there is one)
+        @param highestRight         on return, this is the highest absolute sample from the right
+                                    channel (if there is one)
+        @see read
+    */
+    virtual void readMaxLevels (int64 startSample,
+                                int64 numSamples,
+                                float& lowestLeft,
+                                float& highestLeft,
+                                float& lowestRight,
+                                float& highestRight);
+
+    /** Scans the source looking for a sample whose magnitude is in a specified range.
+
+        This will read from the source, either forwards or backwards between two sample
+        positions, until it finds a sample whose magnitude lies between two specified levels.
+
+        If it finds a suitable sample, it returns its position; if not, it will return -1.
+
+        There's also a minimumConsecutiveSamples setting to help avoid spikes or zero-crossing
+        points when you're searching for a continuous range of samples
+
+        @param startSample              the first sample to look at
+        @param numSamplesToSearch       the number of samples to scan. If this value is negative,
+                                        the search will go backwards
+        @param magnitudeRangeMinimum    the lowest magnitude (inclusive) that is considered a hit, from 0 to 1.0
+        @param magnitudeRangeMaximum    the highest magnitude (inclusive) that is considered a hit, from 0 to 1.0
+        @param minimumConsecutiveSamples    if this is > 0, the method will only look for a sequence
+                                            of this many consecutive samples, all of which lie
+                                            within the target range. When it finds such a sequence,
+                                            it returns the position of the first in-range sample
+                                            it found (i.e. the earliest one if scanning forwards, the
+                                            latest one if scanning backwards)
+    */
+    int64 searchForLevel (int64 startSample,
+                          int64 numSamplesToSearch,
+                          const double magnitudeRangeMinimum,
+                          const double magnitudeRangeMaximum,
+                          const int minimumConsecutiveSamples);
+
+    //==============================================================================
+    /** The sample-rate of the stream. */
+    double sampleRate;
+
+    /** The number of bits per sample, e.g. 16, 24, 32. */
+    unsigned int bitsPerSample;
+
+    /** The total number of samples in the audio stream. */
+    int64 lengthInSamples;
+
+    /** The total number of channels in the audio stream. */
+    unsigned int numChannels;
+
+    /** Indicates whether the data is floating-point or fixed. */
+    bool usesFloatingPointData;
+
+    /** A set of metadata values that the reader has pulled out of the stream.
+
+        Exactly what these values are depends on the format, so you can
+        check out the format implementation code to see what kind of stuff
+        they understand.
+    */
+    StringPairArray metadataValues;
+
+    /** The input stream, for use by subclasses. */
+    InputStream* input;
+
+
+    //==============================================================================
+    /** Subclasses must implement this method to perform the low-level read operation.
+
+        Callers should use read() instead of calling this directly.
+
+        @param destSamples  the array of destination buffers to fill. Some of these
+                            pointers may be null
+        @param numDestChannels  the number of items in the destSamples array. This
+                            value is guaranteed not to be greater than the number of
+                            channels that this reader object contains
+        @param startOffsetInDestBuffer      the number of samples from the start of the
+                            dest data at which to begin writing
+        @param startSampleInFile    the number of samples into the source data at which
+                            to begin reading. This value is guaranteed to be >= 0.
+        @param numSamples   the number of samples to read
+    */
+    virtual bool readSamples (int** destSamples,
+                              int numDestChannels,
+                              int startOffsetInDestBuffer,
+                              int64 startSampleInFile,
+                              int numSamples) = 0;
+
+
+    //==============================================================================
+    juce_UseDebuggingNewOperator
+
+private:
+    String formatName;
+
+    AudioFormatReader (const AudioFormatReader&);
+    const AudioFormatReader& operator= (const AudioFormatReader&);
+};
+
+
+#endif   // __JUCE_AUDIOFORMATREADER_JUCEHEADER__

src/audio/audio_file_formats/juce_AudioFormatWriter.h

@@ -0,0 +1,168 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#ifndef __JUCE_AUDIOFORMATWRITER_JUCEHEADER__
+#define __JUCE_AUDIOFORMATWRITER_JUCEHEADER__
+
+#include "../../io/files/juce_FileOutputStream.h"
+#include "juce_AudioFormatReader.h"
+#include "../audio_sources/juce_AudioSource.h"
+
+
+//==============================================================================
+/**
+    Writes samples to an audio file stream.
+
+    A subclass that writes a specific type of audio format will be created by
+    an AudioFormat object.
+
+    After creating one of these with the AudioFormat::createWriterFor() method
+    you can call its write() method to store the samples, and then delete it.
+
+    @see AudioFormat, AudioFormatReader
+*/
+class JUCE_API  AudioFormatWriter
+{
+protected:
+    //==============================================================================
+    /** Creates an AudioFormatWriter object.
+
+        @param destStream       the stream to write to - this will be deleted
+                                by this object when it is no longer needed
+        @param formatName       the description that will be returned by the getFormatName()
+                                method
+        @param sampleRate       the sample rate to use - the base class just stores
+                                this value, it doesn't do anything with it
+        @param numberOfChannels the number of channels to write - the base class just stores
+                                this value, it doesn't do anything with it
+        @param bitsPerSample    the bit depth of the stream - the base class just stores
+                                this value, it doesn't do anything with it
+    */
+    AudioFormatWriter (OutputStream* const destStream,
+                       const String& formatName,
+                       const double sampleRate,
+                       const unsigned int numberOfChannels,
+                       const unsigned int bitsPerSample);
+
+public:
+    /** Destructor. */
+    virtual ~AudioFormatWriter();
+
+    //==============================================================================
+    /** Returns a description of what type of format this is.
+
+        E.g. "AIFF file"
+    */
+    const String getFormatName() const throw()      { return formatName; }
+
+    //==============================================================================
+    /** Writes a set of samples to the audio stream.
+
+        Note that if you're trying to write the contents of an AudioSampleBuffer, you
+        can use AudioSampleBuffer::writeToAudioWriter().
+
+        @param samplesToWrite   an array of arrays containing the sample data for
+                                each channel to write. This is a zero-terminated
+                                array of arrays, and can contain a different number
+                                of channels than the actual stream uses, and the
+                                writer should do its best to cope with this.
+                                If the format is fixed-point, each channel will be formatted
+                                as an array of signed integers using the full 32-bit
+                                range -0x80000000 to 0x7fffffff, regardless of the source's
+                                bit-depth. If it is a floating-point format, you should treat
+                                the arrays as arrays of floats, and just cast it to an (int**)
+                                to pass it into the method.
+        @param numSamples       the number of samples to write
+    */
+    virtual bool write (const int** samplesToWrite,
+                        int numSamples) = 0;
+
+    //==============================================================================
+    /** Reads a section of samples from an AudioFormatReader, and writes these to
+        the output.
+
+        This will take care of any floating-point conversion that's required to convert
+        between the two formats. It won't deal with sample-rate conversion, though.
+
+        If numSamplesToRead < 0, it will write the entire length of the reader.
+
+        @returns false if it can't read or write properly during the operation
+    */
+    bool writeFromAudioReader (AudioFormatReader& reader,
+                               int64 startSample,
+                               int64 numSamplesToRead);
+
+    /** Reads some samples from an AudioSource, and writes these to the output.
+
+        The source must already have been initialised with the AudioSource::prepareToPlay() method
+
+        @param source               the source to read from
+        @param numSamplesToRead     total number of samples to read and write
+        @param samplesPerBlock      the maximum number of samples to fetch from the source
+        @returns false if it can't read or write properly during the operation
+    */
+    bool writeFromAudioSource (AudioSource& source,
+                               int numSamplesToRead,
+                               const int samplesPerBlock = 2048);
+
+    //==============================================================================
+    /** Returns the sample rate being used. */
+    double getSampleRate() const throw()        { return sampleRate; }
+
+    /** Returns the number of channels being written. */
+    int getNumChannels() const throw()          { return numChannels; }
+
+    /** Returns the bit-depth of the data being written. */
+    int getBitsPerSample() const throw()        { return bitsPerSample; }
+
+    /** Returns true if it's a floating-point format, false if it's fixed-point. */
+    bool isFloatingPoint() const throw()        { return usesFloatingPointData; }
+
+
+    //==============================================================================
+    juce_UseDebuggingNewOperator
+
+protected:
+    /** The sample rate of the stream. */
+    double sampleRate;
+
+    /** The number of channels being written to the stream. */
+    unsigned int numChannels;
+
+    /** The bit depth of the file. */
+    unsigned int bitsPerSample;
+
+    /** True if it's a floating-point format, false if it's fixed-point. */
+    bool usesFloatingPointData;
+
+    /** The output stream for Use by subclasses. */
+    OutputStream* output;
+
+private:
+    String formatName;
+};
+
+
+#endif   // __JUCE_AUDIOFORMATWRITER_JUCEHEADER__

src/audio/audio_file_formats/juce_AudioSubsectionReader.cpp

@@ -0,0 +1,96 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#include "../../core/juce_StandardHeader.h"
+
+BEGIN_JUCE_NAMESPACE
+
+#include "juce_AudioSubsectionReader.h"
+
+
+//==============================================================================
+AudioSubsectionReader::AudioSubsectionReader (AudioFormatReader* const source_,
+                                              const int64 startSample_,
+                                              const int64 length_,
+                                              const bool deleteSourceWhenDeleted_)
+   : AudioFormatReader (0, source_->getFormatName()),
+     source (source_),
+     startSample (startSample_),
+     deleteSourceWhenDeleted (deleteSourceWhenDeleted_)
+{
+    length = jmin (jmax ((int64) 0, source->lengthInSamples - startSample), length_);
+
+    sampleRate = source->sampleRate;
+    bitsPerSample = source->bitsPerSample;
+    lengthInSamples = length;
+    numChannels = source->numChannels;
+    usesFloatingPointData = source->usesFloatingPointData;
+}
+
+AudioSubsectionReader::~AudioSubsectionReader()
+{
+    if (deleteSourceWhenDeleted)
+        delete source;
+}
+
+//==============================================================================
+bool AudioSubsectionReader::readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                                         int64 startSampleInFile, int numSamples)
+{
+    if (startSampleInFile + numSamples > length)
+    {
+        for (int i = numDestChannels; --i >= 0;)
+            if (destSamples[i] != 0)
+                zeromem (destSamples[i], sizeof (int) * numSamples);
+
+        numSamples = jmin (numSamples, (int) (length - startSampleInFile));
+
+        if (numSamples <= 0)
+            return true;
+    }
+
+    return source->readSamples (destSamples, numDestChannels, startOffsetInDestBuffer,
+                                startSampleInFile + startSample, numSamples);
+}
+
+void AudioSubsectionReader::readMaxLevels (int64 startSampleInFile,
+                                           int64 numSamples,
+                                           float& lowestLeft,
+                                           float& highestLeft,
+                                           float& lowestRight,
+                                           float& highestRight)
+{
+    startSampleInFile = jmax ((int64) 0, startSampleInFile);
+    numSamples = jmax ((int64) 0, jmin (numSamples, length - startSampleInFile));
+
+    source->readMaxLevels (startSampleInFile + startSample,
+                           numSamples,
+                           lowestLeft,
+                           highestLeft,
+                           lowestRight,
+                           highestRight);
+}
+
+END_JUCE_NAMESPACE

src/audio/audio_file_formats/juce_AudioSubsectionReader.h

@@ -0,0 +1,93 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#ifndef __JUCE_AUDIOSUBSECTIONREADER_JUCEHEADER__
+#define __JUCE_AUDIOSUBSECTIONREADER_JUCEHEADER__
+
+#include "juce_AudioFormatReader.h"
+
+
+//==============================================================================
+/**
+    This class is used to wrap an AudioFormatReader and only read from a
+    subsection of the file.
+
+    So if you have a reader which can read a 1000 sample file, you could wrap it
+    in one of these to only access, e.g. samples 100 to 200, and any samples
+    outside that will come back as 0. Accessing sample 0 from this reader will
+    actually read the first sample from the other's subsection, which might
+    be at a non-zero position.
+
+    @see AudioFormatReader
+*/
+class JUCE_API  AudioSubsectionReader  : public AudioFormatReader
+{
+public:
+    //==============================================================================
+    /** Creates a AudioSubsectionReader for a given data source.
+
+        @param sourceReader             the source reader from which we'll be taking data
+        @param subsectionStartSample    the sample within the source reader which will be
+                                        mapped onto sample 0 for this reader.
+        @param subsectionLength         the number of samples from the source that will
+                                        make up the subsection. If this reader is asked for
+                                        any samples beyond this region, it will return zero.
+        @param deleteSourceWhenDeleted  if true, the sourceReader object will be deleted when
+                                        this object is deleted.
+    */
+    AudioSubsectionReader (AudioFormatReader* const sourceReader,
+                           const int64 subsectionStartSample,
+                           const int64 subsectionLength,
+                           const bool deleteSourceWhenDeleted);
+
+    /** Destructor. */
+    ~AudioSubsectionReader();
+
+
+    //==============================================================================
+    bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                      int64 startSampleInFile, int numSamples);
+
+    void readMaxLevels (int64 startSample,
+                        int64 numSamples,
+                        float& lowestLeft,
+                        float& highestLeft,
+                        float& lowestRight,
+                        float& highestRight);
+
+
+    //==============================================================================
+    juce_UseDebuggingNewOperator
+
+private:
+    AudioFormatReader* const source;
+    int64 startSample, length;
+    const bool deleteSourceWhenDeleted;
+
+    AudioSubsectionReader (const AudioSubsectionReader&);
+    const AudioSubsectionReader& operator= (const AudioSubsectionReader&);
+};
+
+#endif   // __JUCE_AUDIOSUBSECTIONREADER_JUCEHEADER__

src/audio/audio_file_formats/juce_AudioThumbnail.cpp

@@ -0,0 +1,535 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#include "../../core/juce_StandardHeader.h"
+
+BEGIN_JUCE_NAMESPACE
+
+#include "juce_AudioThumbnail.h"
+#include "juce_AudioThumbnailCache.h"
+
+const int timeBeforeDeletingReader = 2000;
+
+
+//==============================================================================
+struct AudioThumbnailDataFormat
+{
+    char thumbnailMagic[4];
+    int samplesPerThumbSample;
+    int64 totalSamples;         // source samples
+    int64 numFinishedSamples;   // source samples
+    int numThumbnailSamples;
+    int numChannels;
+    int sampleRate;
+    char future[16];
+    char data[1];
+};
+
+#if JUCE_BIG_ENDIAN
+ static void swap (int& n)   { n = (int) swapByteOrder ((uint32) n); }
+ static void swap (int64& n) { n = (int64) swapByteOrder ((uint64) n); }
+#endif
+
+static void swapEndiannessIfNeeded (AudioThumbnailDataFormat* const d)
+{
+    (void) d;
+
+#if JUCE_BIG_ENDIAN
+    swap (d->samplesPerThumbSample);
+    swap (d->totalSamples);
+    swap (d->numFinishedSamples);
+    swap (d->numThumbnailSamples);
+    swap (d->numChannels);
+    swap (d->sampleRate);
+#endif
+}
+
+//==============================================================================
+AudioThumbnail::AudioThumbnail (const int orginalSamplesPerThumbnailSample_,
+                                AudioFormatManager& formatManagerToUse_,
+                                AudioThumbnailCache& cacheToUse)
+    : formatManagerToUse (formatManagerToUse_),
+      cache (cacheToUse),
+      source (0),
+      reader (0),
+      orginalSamplesPerThumbnailSample (orginalSamplesPerThumbnailSample_)
+{
+    clear();
+}
+
+AudioThumbnail::~AudioThumbnail()
+{
+    cache.removeThumbnail (this);
+
+    const ScopedLock sl (readerLock);
+    deleteAndZero (reader);
+
+    delete source;
+}
+
+void AudioThumbnail::setSource (InputSource* const newSource)
+{
+    cache.removeThumbnail (this);
+    timerCallback(); // stops the timer and deletes the reader
+
+    delete source;
+    source = newSource;
+
+    clear();
+
+    if (newSource != 0
+          && ! (cache.loadThumb (*this, newSource->hashCode())
+                 && isFullyLoaded()))
+    {
+        {
+            const ScopedLock sl (readerLock);
+            reader = createReader();
+        }
+
+        if (reader != 0)
+        {
+            initialiseFromAudioFile (*reader);
+            cache.addThumbnail (this);
+        }
+    }
+
+    sendChangeMessage (this);
+}
+
+bool AudioThumbnail::useTimeSlice()
+{
+    const ScopedLock sl (readerLock);
+
+    if (isFullyLoaded())
+    {
+        if (reader != 0)
+            startTimer (timeBeforeDeletingReader);
+
+        cache.removeThumbnail (this);
+        return false;
+    }
+
+    if (reader == 0)
+        reader = createReader();
+
+    if (reader != 0)
+    {
+        readNextBlockFromAudioFile (*reader);
+        stopTimer();
+
+        sendChangeMessage (this);
+
+        const bool justFinished = isFullyLoaded();
+
+        if (justFinished)
+            cache.storeThumb (*this, source->hashCode());
+
+        return ! justFinished;
+    }
+
+    return false;
+}
+
+AudioFormatReader* AudioThumbnail::createReader() const
+{
+    if (source != 0)
+    {
+        InputStream* const audioFileStream = source->createInputStream();
+
+        if (audioFileStream != 0)
+            return formatManagerToUse.createReaderFor (audioFileStream);
+    }
+
+    return 0;
+}
+
+void AudioThumbnail::timerCallback()
+{
+    stopTimer();
+
+    const ScopedLock sl (readerLock);
+    deleteAndZero (reader);
+}
+
+void AudioThumbnail::clear()
+{
+    data.setSize (sizeof (AudioThumbnailDataFormat) + 3);
+
+    AudioThumbnailDataFormat* const d = (AudioThumbnailDataFormat*) data.getData();
+
+    d->thumbnailMagic[0] = 'j';
+    d->thumbnailMagic[1] = 'a';
+    d->thumbnailMagic[2] = 't';
+    d->thumbnailMagic[3] = 'm';
+
+    d->samplesPerThumbSample = orginalSamplesPerThumbnailSample;
+    d->totalSamples = 0;
+    d->numFinishedSamples = 0;
+    d->numThumbnailSamples = 0;
+    d->numChannels = 0;
+    d->sampleRate = 0;
+
+    numSamplesCached = 0;
+    cacheNeedsRefilling = true;
+}
+
+void AudioThumbnail::loadFrom (InputStream& input)
+{
+    data.setSize (0);
+    input.readIntoMemoryBlock (data);
+
+    AudioThumbnailDataFormat* const d = (AudioThumbnailDataFormat*) data.getData();
+    swapEndiannessIfNeeded (d);
+
+    if (! (d->thumbnailMagic[0] == 'j'
+             && d->thumbnailMagic[1] == 'a'
+             && d->thumbnailMagic[2] == 't'
+             && d->thumbnailMagic[3] == 'm'))
+    {
+        clear();
+    }
+
+    numSamplesCached = 0;
+    cacheNeedsRefilling = true;
+}
+
+void AudioThumbnail::saveTo (OutputStream& output) const
+{
+    AudioThumbnailDataFormat* const d = (AudioThumbnailDataFormat*) data.getData();
+    swapEndiannessIfNeeded (d);
+    output.write (data.getData(), data.getSize());
+    swapEndiannessIfNeeded (d);
+}
+
+bool AudioThumbnail::initialiseFromAudioFile (AudioFormatReader& reader)
+{
+    AudioThumbnailDataFormat* d = (AudioThumbnailDataFormat*) data.getData();
+
+    d->totalSamples = reader.lengthInSamples;
+    d->numChannels = jmin (2, reader.numChannels);
+    d->numFinishedSamples = 0;
+    d->sampleRate = roundDoubleToInt (reader.sampleRate);
+    d->numThumbnailSamples = (int) (d->totalSamples / d->samplesPerThumbSample) + 1;
+
+    data.setSize (sizeof (AudioThumbnailDataFormat) + 3 + d->numThumbnailSamples * d->numChannels * 2);
+
+    d = (AudioThumbnailDataFormat*) data.getData();
+    zeromem (&(d->data[0]), d->numThumbnailSamples * d->numChannels * 2);
+
+    return d->totalSamples > 0;
+}
+
+bool AudioThumbnail::readNextBlockFromAudioFile (AudioFormatReader& reader)
+{
+    AudioThumbnailDataFormat* const d = (AudioThumbnailDataFormat*) data.getData();
+
+    if (d->numFinishedSamples < d->totalSamples)
+    {
+        const int numToDo = (int) jmin ((int64) 65536, d->totalSamples - d->numFinishedSamples);
+
+        generateSection (reader,
+                         d->numFinishedSamples,
+                         numToDo);
+
+        d->numFinishedSamples += numToDo;
+    }
+
+    cacheNeedsRefilling = true;
+    return (d->numFinishedSamples < d->totalSamples);
+}
+
+int AudioThumbnail::getNumChannels() const throw()
+{
+    const AudioThumbnailDataFormat* const d = (const AudioThumbnailDataFormat*) data.getData();
+    jassert (d != 0);
+
+    return d->numChannels;
+}
+
+double AudioThumbnail::getTotalLength() const throw()
+{
+    const AudioThumbnailDataFormat* const d = (const AudioThumbnailDataFormat*) data.getData();
+    jassert (d != 0);
+
+    if (d->sampleRate > 0)
+        return d->totalSamples / (double)d->sampleRate;
+    else
+        return 0.0;
+}
+
+void AudioThumbnail::generateSection (AudioFormatReader& reader,
+                                      int64 startSample,
+                                      int numSamples)
+{
+    AudioThumbnailDataFormat* const d = (AudioThumbnailDataFormat*) data.getData();
+    jassert (d != 0);
+
+    int firstDataPos = (int) (startSample / d->samplesPerThumbSample);
+    int lastDataPos = (int) ((startSample + numSamples) / d->samplesPerThumbSample);
+
+    char* l = getChannelData (0);
+    char* r = getChannelData (1);
+
+    for (int i = firstDataPos; i < lastDataPos; ++i)
+    {
+        const int sourceStart = i * d->samplesPerThumbSample;
+        const int sourceEnd = sourceStart + d->samplesPerThumbSample;
+
+        float lowestLeft, highestLeft, lowestRight, highestRight;
+
+        reader.readMaxLevels (sourceStart,
+                              sourceEnd - sourceStart,
+                              lowestLeft,
+                              highestLeft,
+                              lowestRight,
+                              highestRight);
+
+        int n = i * 2;
+
+        if (r != 0)
+        {
+            l [n]   = (char) jlimit (-128.0f, 127.0f, lowestLeft * 127.0f);
+            r [n++] = (char) jlimit (-128.0f, 127.0f, lowestRight * 127.0f);
+            l [n]   = (char) jlimit (-128.0f, 127.0f, highestLeft * 127.0f);
+            r [n++] = (char) jlimit (-128.0f, 127.0f, highestRight * 127.0f);
+        }
+        else
+        {
+            l [n++] = (char) jlimit (-128.0f, 127.0f, lowestLeft * 127.0f);
+            l [n++] = (char) jlimit (-128.0f, 127.0f, highestLeft * 127.0f);
+        }
+    }
+}
+
+char* AudioThumbnail::getChannelData (int channel) const
+{
+    AudioThumbnailDataFormat* const d = (AudioThumbnailDataFormat*) data.getData();
+    jassert (d != 0);
+
+    if (channel >= 0 && channel < d->numChannels)
+        return d->data + (channel * 2 * d->numThumbnailSamples);
+
+    return 0;
+}
+
+bool AudioThumbnail::isFullyLoaded() const throw()
+{
+    const AudioThumbnailDataFormat* const d = (const AudioThumbnailDataFormat*) data.getData();
+    jassert (d != 0);
+
+    return d->numFinishedSamples >= d->totalSamples;
+}
+
+void AudioThumbnail::refillCache (const int numSamples,
+                                  double startTime,
+                                  const double timePerPixel)
+{
+    const AudioThumbnailDataFormat* const d = (const AudioThumbnailDataFormat*) data.getData();
+    jassert (d != 0);
+
+    if (numSamples <= 0
+         || timePerPixel <= 0.0
+         || d->sampleRate <= 0)
+    {
+        numSamplesCached = 0;
+        cacheNeedsRefilling = true;
+        return;
+    }
+
+    if (numSamples == numSamplesCached
+        && numChannelsCached == d->numChannels
+        && startTime == cachedStart
+        && timePerPixel == cachedTimePerPixel
+        && ! cacheNeedsRefilling)
+    {
+        return;
+    }
+
+    numSamplesCached = numSamples;
+    numChannelsCached = d->numChannels;
+    cachedStart = startTime;
+    cachedTimePerPixel = timePerPixel;
+
+    cachedLevels.ensureSize (2 * numChannelsCached * numSamples);
+
+    const bool needExtraDetail = (timePerPixel * d->sampleRate <= d->samplesPerThumbSample);
+
+    const ScopedLock sl (readerLock);
+
+    cacheNeedsRefilling = false;
+
+    if (needExtraDetail && reader == 0)
+        reader = createReader();
+
+    if (reader != 0 && timePerPixel * d->sampleRate <= d->samplesPerThumbSample)
+    {
+        startTimer (timeBeforeDeletingReader);
+
+        char* cacheData = (char*) cachedLevels.getData();
+        int sample = roundDoubleToInt (startTime * d->sampleRate);
+
+        for (int i = numSamples; --i >= 0;)
+        {
+            const int nextSample = roundDoubleToInt ((startTime + timePerPixel) * d->sampleRate);
+
+            if (sample >= 0)
+            {
+                if (sample >= reader->lengthInSamples)
+                    break;
+
+                float lmin, lmax, rmin, rmax;
+
+                reader->readMaxLevels (sample,
+                                       jmax (1, nextSample - sample),
+                                       lmin, lmax, rmin, rmax);
+
+                cacheData[0] = (char) jlimit (-128, 127, roundFloatToInt (lmin * 127.0f));
+                cacheData[1] = (char) jlimit (-128, 127, roundFloatToInt (lmax * 127.0f));
+
+                if (numChannelsCached > 1)
+                {
+                    cacheData[2] = (char) jlimit (-128, 127, roundFloatToInt (rmin * 127.0f));
+                    cacheData[3] = (char) jlimit (-128, 127, roundFloatToInt (rmax * 127.0f));
+                }
+
+                cacheData += 2 * numChannelsCached;
+            }
+
+            startTime += timePerPixel;
+            sample = nextSample;
+        }
+    }
+    else
+    {
+        for (int channelNum = 0; channelNum < numChannelsCached; ++channelNum)
+        {
+            char* const data = getChannelData (channelNum);
+            char* cacheData = ((char*) cachedLevels.getData()) + channelNum * 2;
+
+            const double timeToThumbSampleFactor = d->sampleRate / (double) d->samplesPerThumbSample;
+
+            startTime = cachedStart;
+            int sample = roundDoubleToInt (startTime * timeToThumbSampleFactor);
+            const int numFinished = (int) (d->numFinishedSamples / d->samplesPerThumbSample);
+
+            for (int i = numSamples; --i >= 0;)
+            {
+                const int nextSample = roundDoubleToInt ((startTime + timePerPixel) * timeToThumbSampleFactor);
+
+                if (sample >= 0 && data != 0)
+                {
+                    char mx = -128;
+                    char mn = 127;
+
+                    while (sample <= nextSample)
+                    {
+                        if (sample >= numFinished)
+                            break;
+
+                        const int n = sample << 1;
+                        const char sampMin = data [n];
+                        const char sampMax = data [n + 1];
+
+                        if (sampMin < mn)
+                            mn = sampMin;
+
+                        if (sampMax > mx)
+                            mx = sampMax;
+
+                        ++sample;
+                    }
+
+                    if (mn <= mx)
+                    {
+                        cacheData[0] = mn;
+                        cacheData[1] = mx;
+                    }
+                    else
+                    {
+                        cacheData[0] = 1;
+                        cacheData[1] = 0;
+                    }
+                }
+                else
+                {
+                    cacheData[0] = 1;
+                    cacheData[1] = 0;
+                }
+
+                cacheData += numChannelsCached * 2;
+                startTime += timePerPixel;
+                sample = nextSample;
+            }
+        }
+    }
+}
+
+void AudioThumbnail::drawChannel (Graphics& g,
+                                  int x, int y, int w, int h,
+                                  double startTime,
+                                  double endTime,
+                                  int channelNum,
+                                  const float verticalZoomFactor)
+{
+    refillCache (w, startTime, (endTime - startTime) / w);
+
+    if (numSamplesCached >= w
+        && channelNum >= 0
+        && channelNum < numChannelsCached)
+    {
+        const float topY = (float) y;
+        const float bottomY = topY + h;
+        const float midY = topY + h * 0.5f;
+        const float vscale = verticalZoomFactor * h / 256.0f;
+
+        const Rectangle clip (g.getClipBounds());
+        const int skipLeft = jlimit (0, w, clip.getX() - x);
+        w -= skipLeft;
+        x += skipLeft;
+
+        const char* cacheData = ((const char*) cachedLevels.getData())
+                                   + (channelNum << 1)
+                                   + skipLeft * (numChannelsCached << 1);
+
+        while (--w >= 0)
+        {
+            const char mn = cacheData[0];
+            const char mx = cacheData[1];
+            cacheData += numChannelsCached << 1;
+
+            if (mn <= mx) // if the wrong way round, signifies that the sample's not yet known
+                g.drawLine ((float) x, jmax (midY - mx * vscale - 0.3f, topY),
+                            (float) x, jmin (midY - mn * vscale + 0.3f, bottomY));
+
+            ++x;
+
+            if (x >= clip.getRight())
+                break;
+        }
+    }
+}
+
+
+END_JUCE_NAMESPACE

src/audio/audio_file_formats/juce_AudioThumbnail.h

@@ -0,0 +1,189 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#ifndef __JUCE_AUDIOTHUMBNAIL_JUCEHEADER__
+#define __JUCE_AUDIOTHUMBNAIL_JUCEHEADER__
+
+#include "../../threads/juce_TimeSliceThread.h"
+#include "../../io/streams/juce_InputSource.h"
+#include "../../io/streams/juce_OutputStream.h"
+#include "../../events/juce_ChangeBroadcaster.h"
+#include "../../events/juce_Timer.h"
+#include "../../gui/graphics/contexts/juce_Graphics.h"
+#include "juce_AudioFormatReader.h"
+#include "juce_AudioFormatManager.h"
+
+class AudioThumbnailCache;
+
+
+//==============================================================================
+/**
+    Makes it easy to quickly draw scaled views of the waveform shape of an
+    audio file.
+
+    To use this class, just create an AudioThumbNail class for the file you want
+    to draw, call setSource to tell it which file or resource to use, then call
+    drawChannel() to draw it.
+
+    The class will asynchronously scan the wavefile to create its scaled-down view,
+    so you should make your UI repaint itself as this data comes in. To do this, the
+    AudioThumbnail is a ChangeBroadcaster, and will broadcast a message when its
+    listeners should repaint themselves.
+
+    The thumbnail stores an internal low-res version of the wave data, and this can
+    be loaded and saved to avoid having to scan the file again.
+
+    @see AudioThumbnailCache
+*/
+class JUCE_API  AudioThumbnail    : public ChangeBroadcaster,
+                                    public TimeSliceClient,
+                                    private Timer
+{
+public:
+    //==============================================================================
+    /** Creates an audio thumbnail.
+
+        @param sourceSamplesPerThumbnailSample  when creating a stored, low-res version
+                        of the audio data, this is the scale at which it should be done
+        @param formatManagerToUse   the audio format manager that is used to open the file
+        @param cacheToUse   an instance of an AudioThumbnailCache - this provides a background
+                            thread and storage that is used to by the thumbnail, and the cache
+                            object can be shared between multiple thumbnails
+    */
+    AudioThumbnail (const int sourceSamplesPerThumbnailSample,
+                    AudioFormatManager& formatManagerToUse,
+                    AudioThumbnailCache& cacheToUse);
+
+    /** Destructor. */
+    ~AudioThumbnail();
+
+    //==============================================================================
+    /** Specifies the file or stream that contains the audio file.
+
+        For a file, just call
+        @code
+        setSource (new FileInputSource (file))
+        @endcode
+
+        You can pass a zero in here to clear the thumbnail.
+
+        The source that is passed in will be deleted by this object when it is no
+        longer needed
+    */
+    void setSource (InputSource* const newSource);
+
+    /** Reloads the low res thumbnail data from an input stream.
+
+        The thumb will automatically attempt to reload itself from its
+        AudioThumbnailCache.
+    */
+    void loadFrom (InputStream& input);
+
+    /** Saves the low res thumbnail data to an output stream.
+
+        The thumb will automatically attempt to save itself to its
+        AudioThumbnailCache after it finishes scanning the wave file.
+    */
+    void saveTo (OutputStream& output) const;
+
+    //==============================================================================
+    /** Returns the number of channels in the file.
+    */
+    int getNumChannels() const throw();
+
+    /** Returns the length of the audio file.
+    */
+    double getTotalLength() const throw();
+
+    /** Renders the waveform shape for a channel.
+
+        The waveform will be drawn within  the specified rectangle, where startTime
+        and endTime specify the times within the audio file that should be positioned
+        at the left and right edges of the rectangle.
+
+        The waveform will be scaled vertically so that a full-volume sample will fill
+        the rectangle vertically, but you can also specify an extra vertical scale factor
+        with the verticalZoomFactor parameter.
+    */
+    void drawChannel (Graphics& g,
+                      int x, int y, int w, int h,
+                      double startTime,
+                      double endTime,
+                      int channelNum,
+                      const float verticalZoomFactor);
+
+    /** Returns true if the low res preview is fully generated.
+    */
+    bool isFullyLoaded() const throw();
+
+    //==============================================================================
+    /** @internal */
+    bool useTimeSlice();
+    /** @internal */
+    void timerCallback();
+
+    //==============================================================================
+    juce_UseDebuggingNewOperator
+
+private:
+    AudioFormatManager& formatManagerToUse;
+    AudioThumbnailCache& cache;
+    InputSource* source;
+
+    CriticalSection readerLock;
+    AudioFormatReader* reader;
+
+    MemoryBlock data, cachedLevels;
+    int orginalSamplesPerThumbnailSample;
+
+    int numChannelsCached, numSamplesCached;
+    double cachedStart, cachedTimePerPixel;
+    bool cacheNeedsRefilling;
+
+    void clear();
+
+    AudioFormatReader* createReader() const;
+
+    void generateSection (AudioFormatReader& reader,
+                          int64 startSample,
+                          int numSamples);
+
+    char* getChannelData (int channel) const;
+
+    void refillCache (const int numSamples,
+                      double startTime,
+                      const double timePerPixel);
+
+    friend class AudioThumbnailCache;
+
+    // true if it needs more callbacks from the readNextBlockFromAudioFile() method
+    bool initialiseFromAudioFile (AudioFormatReader& reader);
+
+    // returns true if more needs to be read
+    bool readNextBlockFromAudioFile (AudioFormatReader& reader);
+};
+
+
+#endif   // __JUCE_AUDIOTHUMBNAIL_JUCEHEADER__

src/audio/audio_file_formats/juce_AudioThumbnailCache.cpp

@@ -0,0 +1,139 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#include "../../core/juce_StandardHeader.h"
+
+BEGIN_JUCE_NAMESPACE
+
+#include "juce_AudioThumbnailCache.h"
+#include "../../io/streams/juce_MemoryInputStream.h"
+#include "../../io/streams/juce_MemoryOutputStream.h"
+
+
+//==============================================================================
+struct ThumbnailCacheEntry
+{
+    int64 hash;
+    uint32 lastUsed;
+    MemoryBlock data;
+
+    juce_UseDebuggingNewOperator
+};
+
+//==============================================================================
+AudioThumbnailCache::AudioThumbnailCache (const int maxNumThumbsToStore_)
+    : TimeSliceThread (T("thumb cache")),
+      maxNumThumbsToStore (maxNumThumbsToStore_)
+{
+    startThread (2);
+}
+
+AudioThumbnailCache::~AudioThumbnailCache()
+{
+}
+
+bool AudioThumbnailCache::loadThumb (AudioThumbnail& thumb, const int64 hashCode)
+{
+    for (int i = thumbs.size(); --i >= 0;)
+    {
+        if (thumbs[i]->hash == hashCode)
+        {
+            MemoryInputStream in ((const char*) thumbs[i]->data.getData(),
+                                  thumbs[i]->data.getSize(),
+                                  false);
+
+            thumb.loadFrom (in);
+
+            thumbs[i]->lastUsed = Time::getMillisecondCounter();
+            return true;
+        }
+    }
+
+    return false;
+}
+
+void AudioThumbnailCache::storeThumb (const AudioThumbnail& thumb,
+                                      const int64 hashCode)
+{
+    MemoryOutputStream out;
+    thumb.saveTo (out);
+
+    ThumbnailCacheEntry* te = 0;
+
+    for (int i = thumbs.size(); --i >= 0;)
+    {
+        if (thumbs[i]->hash == hashCode)
+        {
+            te = thumbs[i];
+            break;
+        }
+    }
+
+    if (te == 0)
+    {
+        te = new ThumbnailCacheEntry();
+        te->hash = hashCode;
+
+        if (thumbs.size() < maxNumThumbsToStore)
+        {
+            thumbs.add (te);
+        }
+        else
+        {
+            int oldest = 0;
+            unsigned int oldestTime = Time::getMillisecondCounter() + 1;
+
+            int i;
+            for (i = thumbs.size(); --i >= 0;)
+                if (thumbs[i]->lastUsed < oldestTime)
+                    oldest = i;
+
+            thumbs.set (i, te);
+        }
+    }
+
+    te->lastUsed = Time::getMillisecondCounter();
+    te->data.setSize (0);
+    te->data.append (out.getData(), out.getDataSize());
+}
+
+void AudioThumbnailCache::clear()
+{
+    thumbs.clear();
+}
+
+//==============================================================================
+void AudioThumbnailCache::addThumbnail (AudioThumbnail* const thumb)
+{
+    addTimeSliceClient (thumb);
+}
+
+void AudioThumbnailCache::removeThumbnail (AudioThumbnail* const thumb)
+{
+    removeTimeSliceClient (thumb);
+}
+
+
+END_JUCE_NAMESPACE

src/audio/audio_file_formats/juce_AudioThumbnailCache.h

@@ -0,0 +1,92 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#ifndef __JUCE_AUDIOTHUMBNAILCACHE_JUCEHEADER__
+#define __JUCE_AUDIOTHUMBNAILCACHE_JUCEHEADER__
+
+#include "juce_AudioThumbnail.h"
+struct ThumbnailCacheEntry;
+
+
+//==============================================================================
+/**
+    An instance of this class is used to manage multiple AudioThumbnail objects.
+
+    The cache runs a single background thread that is shared by all the thumbnails
+    that need it, and it maintains a set of low-res previews in memory, to avoid
+    having to re-scan audio files too often.
+
+    @see AudioThumbnail
+*/
+class JUCE_API  AudioThumbnailCache   : public TimeSliceThread
+{
+public:
+    //==============================================================================
+    /** Creates a cache object.
+
+        The maxNumThumbsToStore parameter lets you specify how many previews should
+        be kept in memory at once.
+    */
+    AudioThumbnailCache (const int maxNumThumbsToStore);
+
+    /** Destructor. */
+    ~AudioThumbnailCache();
+
+    //==============================================================================
+    /** Clears out any stored thumbnails.
+    */
+    void clear();
+
+    /** Reloads the specified thumb if this cache contains the appropriate stored
+        data.
+
+        This is called automatically by the AudioThumbnail class, so you shouldn't
+        normally need to call it directly.
+    */
+    bool loadThumb (AudioThumbnail& thumb, const int64 hashCode);
+
+    /** Stores the cachable data from the specified thumb in this cache.
+
+        This is called automatically by the AudioThumbnail class, so you shouldn't
+        normally need to call it directly.
+    */
+    void storeThumb (const AudioThumbnail& thumb, const int64 hashCode);
+
+
+    //==============================================================================
+    juce_UseDebuggingNewOperator
+
+private:
+    //==============================================================================
+    OwnedArray <ThumbnailCacheEntry> thumbs;
+    int maxNumThumbsToStore;
+
+    friend class AudioThumbnail;
+    void addThumbnail (AudioThumbnail* const thumb);
+    void removeThumbnail (AudioThumbnail* const thumb);
+};
+
+
+#endif   // __JUCE_AUDIOTHUMBNAILCACHE_JUCEHEADER__

src/audio/audio_file_formats/juce_FlacAudioFormat.cpp

@@ -0,0 +1,554 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#include "../../../juce_Config.h"
+#ifdef _MSC_VER
+ #include <windows.h>
+#endif
+
+#if JUCE_USE_FLAC
+
+#include "../../core/juce_StandardHeader.h"
+
+#ifdef _MSC_VER
+  #pragma warning (disable : 4505)
+  #pragma warning (push)
+#endif
+
+namespace FlacNamespace
+{
+#if JUCE_INCLUDE_FLAC_CODE
+ #define FLAC__NO_DLL 1
+
+ #if ! defined (SIZE_MAX)
+  #define SIZE_MAX 0xffffffff
+ #endif
+
+ #define __STDC_LIMIT_MACROS 1
+ #include "flac/all.h"
+ #include "flac/libFLAC/bitmath.c"
+ #include "flac/libFLAC/bitreader.c"
+ #include "flac/libFLAC/bitwriter.c"
+ #include "flac/libFLAC/cpu.c"
+ #include "flac/libFLAC/crc.c"
+ #include "flac/libFLAC/fixed.c"
+ #include "flac/libFLAC/float.c"
+ #include "flac/libFLAC/format.c"
+ #include "flac/libFLAC/lpc_flac.c"
+ #include "flac/libFLAC/md5.c"
+ #include "flac/libFLAC/memory.c"
+ #include "flac/libFLAC/stream_decoder.c"
+ #include "flac/libFLAC/stream_encoder.c"
+ #include "flac/libFLAC/stream_encoder_framing.c"
+ #include "flac/libFLAC/window_flac.c"
+#else
+ #include <FLAC/all.h>
+#endif
+}
+
+#ifdef _MSC_VER
+  #pragma warning (pop)
+#endif
+
+BEGIN_JUCE_NAMESPACE
+
+#include "juce_FlacAudioFormat.h"
+#include "../../text/juce_LocalisedStrings.h"
+
+using namespace FlacNamespace;
+
+//==============================================================================
+#define flacFormatName                          TRANS("FLAC file")
+static const tchar* const flacExtensions[] =    { T(".flac"), 0 };
+
+
+//==============================================================================
+class FlacReader  : public AudioFormatReader
+{
+    FLAC__StreamDecoder* decoder;
+    AudioSampleBuffer reservoir;
+    int reservoirStart, samplesInReservoir;
+    bool ok, scanningForLength;
+
+public:
+    //==============================================================================
+    FlacReader (InputStream* const in)
+        : AudioFormatReader (in, flacFormatName),
+          reservoir (2, 0),
+          reservoirStart (0),
+          samplesInReservoir (0),
+          scanningForLength (false)
+    {
+        using namespace FlacNamespace;
+        lengthInSamples = 0;
+
+        decoder = FLAC__stream_decoder_new();
+
+        ok = FLAC__stream_decoder_init_stream (decoder,
+                                               readCallback_, seekCallback_, tellCallback_, lengthCallback_,
+                                               eofCallback_, writeCallback_, metadataCallback_, errorCallback_,
+                                               (void*) 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()
+    {
+        FLAC__stream_decoder_delete (decoder);
+    }
+
+    void useMetadata (const FLAC__StreamMetadata_StreamInfo& info)
+    {
+        sampleRate = info.sample_rate;
+        bitsPerSample = info.bits_per_sample;
+        lengthInSamples = (unsigned int) info.total_samples;
+        numChannels = info.channels;
+
+        reservoir.setSize (numChannels, 2 * info.max_blocksize, false, false, true);
+    }
+
+    // returns the number of samples read
+    bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                      int64 startSampleInFile, int numSamples)
+    {
+        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] != 0)
+                        memcpy (destSamples[i] + startOffsetInDestBuffer,
+                                reservoir.getSampleData (i, (int) (startSampleInFile - reservoirStart)),
+                                sizeof (int) * num);
+
+                startOffsetInDestBuffer += num;
+                startSampleInFile += num;
+                numSamples -= num;
+            }
+            else
+            {
+                if (startSampleInFile < reservoirStart
+                     || startSampleInFile > reservoirStart + jmax (samplesInReservoir, 511))
+                {
+                    if (startSampleInFile >= (int) lengthInSamples)
+                    {
+                        samplesInReservoir = 0;
+                        break;
+                    }
+
+                    // 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);
+                    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] != 0)
+                    zeromem (destSamples[i] + startOffsetInDestBuffer,
+                             sizeof (int) * numSamples);
+        }
+
+        return true;
+    }
+
+    void useSamples (const FLAC__int32* const buffer[], int numSamples)
+    {
+        if (scanningForLength)
+        {
+            lengthInSamples += numSamples;
+        }
+        else
+        {
+            if (numSamples > reservoir.getNumSamples())
+                reservoir.setSize (numChannels, numSamples, false, false, true);
+
+            const int bitsToShift = 32 - bitsPerSample;
+
+            for (int i = 0; i < (int) numChannels; ++i)
+            {
+                const FLAC__int32* src = buffer[i];
+
+                int n = i;
+                while (src == 0 && n > 0)
+                    src = buffer [--n];
+
+                if (src != 0)
+                {
+                    int* dest = (int*) reservoir.getSampleData(i);
+
+                    for (int j = 0; j < numSamples; ++j)
+                        dest[j] = src[j] << bitsToShift;
+                }
+            }
+
+            samplesInReservoir = numSamples;
+        }
+    }
+
+    //==============================================================================
+    static FLAC__StreamDecoderReadStatus readCallback_ (const FLAC__StreamDecoder*, FLAC__byte buffer[], size_t* bytes, void* client_data)
+    {
+        *bytes = (unsigned int) ((const FlacReader*) client_data)->input->read (buffer, (int) *bytes);
+        return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
+    }
+
+    static FLAC__StreamDecoderSeekStatus seekCallback_ (const FLAC__StreamDecoder*, FLAC__uint64 absolute_byte_offset, void* client_data)
+    {
+        ((const FlacReader*) client_data)->input->setPosition ((int) absolute_byte_offset);
+        return FLAC__STREAM_DECODER_SEEK_STATUS_OK;
+    }
+
+    static FLAC__StreamDecoderTellStatus tellCallback_ (const FLAC__StreamDecoder*, FLAC__uint64* absolute_byte_offset, void* client_data)
+    {
+        *absolute_byte_offset = ((const FlacReader*) client_data)->input->getPosition();
+        return FLAC__STREAM_DECODER_TELL_STATUS_OK;
+    }
+
+    static FLAC__StreamDecoderLengthStatus lengthCallback_ (const FLAC__StreamDecoder*, FLAC__uint64* stream_length, void* client_data)
+    {
+        *stream_length = ((const FlacReader*) client_data)->input->getTotalLength();
+        return FLAC__STREAM_DECODER_LENGTH_STATUS_OK;
+    }
+
+    static FLAC__bool eofCallback_ (const FLAC__StreamDecoder*, void* client_data)
+    {
+        return ((const FlacReader*) client_data)->input->isExhausted();
+    }
+
+    static FLAC__StreamDecoderWriteStatus writeCallback_ (const FLAC__StreamDecoder*,
+                                                          const FLAC__Frame* frame,
+                                                          const FLAC__int32* const buffer[],
+                                                          void* client_data)
+    {
+        ((FlacReader*) client_data)->useSamples (buffer, frame->header.blocksize);
+        return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
+    }
+
+    static void metadataCallback_ (const FLAC__StreamDecoder*,
+                                   const FLAC__StreamMetadata* metadata,
+                                   void* client_data)
+    {
+        ((FlacReader*) client_data)->useMetadata (metadata->data.stream_info);
+    }
+
+    static void errorCallback_ (const FLAC__StreamDecoder*, FLAC__StreamDecoderErrorStatus, void*)
+    {
+    }
+
+    juce_UseDebuggingNewOperator
+};
+
+
+//==============================================================================
+class FlacWriter  : public AudioFormatWriter
+{
+    FLAC__StreamEncoder* encoder;
+    MemoryBlock temp;
+
+public:
+    bool ok;
+
+    //==============================================================================
+    FlacWriter (OutputStream* const out,
+                const double sampleRate,
+                const int numChannels,
+                const int bitsPerSample_)
+        : AudioFormatWriter (out, flacFormatName,
+                             sampleRate,
+                             numChannels,
+                             bitsPerSample_)
+    {
+        using namespace FlacNamespace;
+        encoder = FLAC__stream_encoder_new();
+
+        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 (24, bitsPerSample));
+        FLAC__stream_encoder_set_sample_rate (encoder, (unsigned int) sampleRate);
+        FLAC__stream_encoder_set_blocksize (encoder, 2048);
+        FLAC__stream_encoder_set_do_escape_coding (encoder, true);
+
+        ok = FLAC__stream_encoder_init_stream (encoder,
+                                               encodeWriteCallback, encodeSeekCallback,
+                                               encodeTellCallback, encodeMetadataCallback,
+                                               (void*) this) == FLAC__STREAM_ENCODER_INIT_STATUS_OK;
+    }
+
+    ~FlacWriter()
+    {
+        if (ok)
+        {
+            FLAC__stream_encoder_finish (encoder);
+            output->flush();
+        }
+        else
+        {
+            output = 0; // to stop the base class deleting this, as it needs to be returned
+                        // to the caller of createWriter()
+        }
+
+        FLAC__stream_encoder_delete (encoder);
+    }
+
+    //==============================================================================
+    bool write (const int** samplesToWrite, int numSamples)
+    {
+        if (! ok)
+            return false;
+
+        int* buf[3];
+        const int bitsToShift = 32 - bitsPerSample;
+
+        if (bitsToShift > 0)
+        {
+            const int numChannels = (samplesToWrite[1] == 0) ? 1 : 2;
+            temp.setSize (sizeof (int) * numSamples * numChannels);
+
+            buf[0] = (int*) temp.getData();
+            buf[1] = buf[0] + numSamples;
+            buf[2] = 0;
+
+            for (int i = numChannels; --i >= 0;)
+            {
+                if (samplesToWrite[i] != 0)
+                {
+                    for (int j = 0; j < numSamples; ++j)
+                        buf [i][j] = (samplesToWrite [i][j] >> bitsToShift);
+                }
+            }
+
+            samplesToWrite = (const int**) buf;
+        }
+
+        return FLAC__stream_encoder_process (encoder,
+                                             (const FLAC__int32**) samplesToWrite,
+                                             numSamples) != 0;
+    }
+
+    bool writeData (const void* const data, const int size) const
+    {
+        return output->write (data, size);
+    }
+
+    static void packUint32 (FLAC__uint32 val, FLAC__byte* b, const int bytes)
+    {
+        b += bytes;
+
+        for (int i = 0; i < bytes; ++i)
+        {
+            *(--b) = (FLAC__byte) (val & 0xff);
+            val >>= 8;
+        }
+    }
+
+    void writeMetaData (const 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 ok = output->setPosition (4);
+        (void) ok;
+
+        // 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 (ok);
+
+        output->writeIntBigEndian (FLAC__STREAM_METADATA_STREAMINFO_LENGTH);
+        output->write (buffer, FLAC__STREAM_METADATA_STREAMINFO_LENGTH);
+    }
+
+    //==============================================================================
+    static FLAC__StreamEncoderWriteStatus encodeWriteCallback (const FLAC__StreamEncoder*,
+                                                               const FLAC__byte buffer[],
+                                                               size_t bytes,
+                                                               unsigned int /*samples*/,
+                                                               unsigned int /*current_frame*/,
+                                                               void* client_data)
+    {
+        using namespace FlacNamespace;
+        return ((FlacWriter*) client_data)->writeData (buffer, (int) bytes)
+                ? FLAC__STREAM_ENCODER_WRITE_STATUS_OK
+                : FLAC__STREAM_ENCODER_WRITE_STATUS_FATAL_ERROR;
+    }
+
+    static FLAC__StreamEncoderSeekStatus encodeSeekCallback (const FLAC__StreamEncoder*, FLAC__uint64, void*)
+    {
+        return FLAC__STREAM_ENCODER_SEEK_STATUS_UNSUPPORTED;
+    }
+
+    static FLAC__StreamEncoderTellStatus encodeTellCallback (const FLAC__StreamEncoder*, FLAC__uint64*, void*)
+    {
+        return FLAC__STREAM_ENCODER_TELL_STATUS_UNSUPPORTED;
+    }
+
+    static void encodeMetadataCallback (const FLAC__StreamEncoder*,
+                                        const FLAC__StreamMetadata* metadata,
+                                        void* client_data)
+    {
+        ((FlacWriter*) client_data)->writeMetaData (metadata);
+    }
+
+    juce_UseDebuggingNewOperator
+};
+
+
+//==============================================================================
+FlacAudioFormat::FlacAudioFormat()
+    : AudioFormat (flacFormatName, (const tchar**) flacExtensions)
+{
+}
+
+FlacAudioFormat::~FlacAudioFormat()
+{
+}
+
+const Array <int> FlacAudioFormat::getPossibleSampleRates()
+{
+    const int rates[] = { 22050, 32000, 44100, 48000, 88200, 96000, 0 };
+    return Array <int> (rates);
+}
+
+const Array <int> FlacAudioFormat::getPossibleBitDepths()
+{
+    const int depths[] = { 16, 24, 0 };
+    return Array <int> (depths);
+}
+
+bool FlacAudioFormat::canDoStereo()
+{
+    return true;
+}
+
+bool FlacAudioFormat::canDoMono()
+{
+    return true;
+}
+
+bool FlacAudioFormat::isCompressed()
+{
+    return true;
+}
+
+AudioFormatReader* FlacAudioFormat::createReaderFor (InputStream* in,
+                                                     const bool deleteStreamIfOpeningFails)
+{
+    FlacReader* r = new FlacReader (in);
+
+    if (r->sampleRate == 0)
+    {
+        if (! deleteStreamIfOpeningFails)
+            r->input = 0;
+
+        deleteAndZero (r);
+    }
+
+    return r;
+}
+
+AudioFormatWriter* FlacAudioFormat::createWriterFor (OutputStream* out,
+                                                     double sampleRate,
+                                                     unsigned int numberOfChannels,
+                                                     int bitsPerSample,
+                                                     const StringPairArray& /*metadataValues*/,
+                                                     int /*qualityOptionIndex*/)
+{
+    if (getPossibleBitDepths().contains (bitsPerSample))
+    {
+        FlacWriter* w = new FlacWriter (out,
+                                        sampleRate,
+                                        numberOfChannels,
+                                        bitsPerSample);
+
+        if (! w->ok)
+            deleteAndZero (w);
+
+        return w;
+    }
+
+    return 0;
+}
+
+END_JUCE_NAMESPACE
+
+#endif

src/audio/audio_file_formats/juce_FlacAudioFormat.h

@@ -0,0 +1,75 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#ifndef __JUCE_FLACAUDIOFORMAT_JUCEHEADER__
+#define __JUCE_FLACAUDIOFORMAT_JUCEHEADER__
+
+#include "juce_AudioFormat.h" // (must keep this outside the conditional define)
+
+#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 in juce_Config.h,
+    and make sure your include search path and library search path are set up to find
+    the FLAC header files and static libraries.
+
+    @see AudioFormat
+*/
+class JUCE_API  FlacAudioFormat    : public AudioFormat
+{
+public:
+    //==============================================================================
+    FlacAudioFormat();
+    ~FlacAudioFormat();
+
+    //==============================================================================
+    const Array <int> getPossibleSampleRates();
+    const Array <int> getPossibleBitDepths();
+    bool canDoStereo();
+    bool canDoMono();
+    bool isCompressed();
+
+    //==============================================================================
+    AudioFormatReader* createReaderFor (InputStream* sourceStream,
+                                        const bool deleteStreamIfOpeningFails);
+
+    AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
+                                        double sampleRateToUse,
+                                        unsigned int numberOfChannels,
+                                        int bitsPerSample,
+                                        const StringPairArray& metadataValues,
+                                        int qualityOptionIndex);
+
+    //==============================================================================
+    juce_UseDebuggingNewOperator
+};
+
+
+#endif
+#endif   // __JUCE_FLACAUDIOFORMAT_JUCEHEADER__

src/audio/audio_file_formats/juce_OggVorbisAudioFormat.cpp

@@ -0,0 +1,504 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#include "../../../juce_Config.h"
+
+#if JUCE_USE_OGGVORBIS
+
+#include "../../core/juce_StandardHeader.h"
+
+#if JUCE_MAC
+  #define __MACOSX__ 1
+#endif
+
+
+namespace OggVorbisNamespace
+{
+#if JUCE_INCLUDE_OGGVORBIS_CODE
+ #include "oggvorbis/vorbisenc.h"
+ #include "oggvorbis/codec.h"
+ #include "oggvorbis/vorbisfile.h"
+
+ #include "oggvorbis/bitwise.c"
+ #include "oggvorbis/framing.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/analysis.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/bitrate.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/block.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/codebook.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/envelope.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/floor0.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/floor1.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/info.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/lpc.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/lsp.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/mapping0.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/mdct.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/psy.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/registry.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/res0.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/sharedbook.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/smallft.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/synthesis.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/vorbisenc.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/vorbisfile.c"
+ #include "oggvorbis/libvorbis-1.1.2/lib/window.c"
+#else
+ #include <vorbis/vorbisenc.h>
+ #include <vorbis/codec.h>
+ #include <vorbis/vorbisfile.h>
+#endif
+}
+
+BEGIN_JUCE_NAMESPACE
+
+#include "juce_OggVorbisAudioFormat.h"
+#include "../../application/juce_Application.h"
+#include "../../core/juce_Random.h"
+#include "../../io/files/juce_FileInputStream.h"
+#include "../../text/juce_LocalisedStrings.h"
+
+using namespace OggVorbisNamespace;
+
+//==============================================================================
+#define oggFormatName                          TRANS("Ogg-Vorbis file")
+static const tchar* const oggExtensions[] =    { T(".ogg"), 0 };
+
+
+//==============================================================================
+class OggReader : public AudioFormatReader
+{
+    OggVorbis_File ovFile;
+    ov_callbacks callbacks;
+    AudioSampleBuffer reservoir;
+    int reservoirStart, samplesInReservoir;
+
+public:
+    //==============================================================================
+    OggReader (InputStream* const inp)
+        : AudioFormatReader (inp, oggFormatName),
+          reservoir (2, 4096),
+          reservoirStart (0),
+          samplesInReservoir (0)
+    {
+        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);
+            lengthInSamples = (uint32) ov_pcm_total (&ovFile, -1);
+            numChannels = info->channels;
+            bitsPerSample = 16;
+            sampleRate = info->rate;
+
+            reservoir.setSize (numChannels,
+                               (int) jmin (lengthInSamples, (int64) reservoir.getNumSamples()));
+        }
+    }
+
+    ~OggReader()
+    {
+        ov_clear (&ovFile);
+    }
+
+    //==============================================================================
+    bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                      int64 startSampleInFile, int numSamples)
+    {
+        while (numSamples > 0)
+        {
+            const int numAvailable = 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] != 0)
+                        memcpy (destSamples[i] + startOffsetInDestBuffer,
+                                reservoir.getSampleData (i, (int) (startSampleInFile - reservoirStart)),
+                                sizeof (float) * 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) ov_pcm_tell (&ovFile))
+                    ov_pcm_seek (&ovFile, reservoirStart);
+
+                int offset = 0;
+                int numToRead = samplesInReservoir;
+
+                while (numToRead > 0)
+                {
+                    float** dataIn = 0;
+
+                    const int samps = ov_read_float (&ovFile, &dataIn, numToRead, &bitStream);
+                    if (samps == 0)
+                        break;
+
+                    jassert (samps <= numToRead);
+
+                    for (int i = jmin (numChannels, reservoir.getNumChannels()); --i >= 0;)
+                    {
+                        memcpy (reservoir.getSampleData (i, offset),
+                                dataIn[i],
+                                sizeof (float) * samps);
+                    }
+
+                    numToRead -= samps;
+                    offset += samps;
+                }
+
+                if (numToRead > 0)
+                    reservoir.clear (offset, numToRead);
+            }
+        }
+
+        if (numSamples > 0)
+        {
+            for (int i = numDestChannels; --i >= 0;)
+                if (destSamples[i] != 0)
+                    zeromem (destSamples[i] + startOffsetInDestBuffer,
+                             sizeof (int) * numSamples);
+        }
+
+        return true;
+    }
+
+    //==============================================================================
+    static size_t oggReadCallback (void* ptr, size_t size, size_t nmemb, void* datasource)
+    {
+        return (size_t) (((InputStream*) datasource)->read (ptr, (int) (size * nmemb)) / size);
+    }
+
+    static int oggSeekCallback (void* datasource, ogg_int64_t offset, int whence)
+    {
+        InputStream* const in = (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) ((InputStream*) datasource)->getPosition();
+    }
+
+    juce_UseDebuggingNewOperator
+};
+
+//==============================================================================
+class OggWriter  : public AudioFormatWriter
+{
+    ogg_stream_state os;
+    ogg_page og;
+    ogg_packet op;
+    vorbis_info vi;
+    vorbis_comment vc;
+    vorbis_dsp_state vd;
+    vorbis_block vb;
+
+public:
+    bool ok;
+
+    //==============================================================================
+    OggWriter (OutputStream* const out,
+               const double sampleRate,
+               const int numChannels,
+               const int bitsPerSample,
+               const int qualityIndex)
+        : AudioFormatWriter (out, oggFormatName,
+                             sampleRate,
+                             numChannels,
+                             bitsPerSample)
+    {
+        ok = false;
+
+        vorbis_info_init (&vi);
+
+        if (vorbis_encode_init_vbr (&vi,
+                                    numChannels,
+                                    (int) sampleRate,
+                                    jlimit (0.0f, 1.0f, qualityIndex * 0.5f)) == 0)
+        {
+            vorbis_comment_init (&vc);
+
+            if (JUCEApplication::getInstance() != 0)
+                vorbis_comment_add_tag (&vc, "ENCODER",
+                                        (char*) (const char*) JUCEApplication::getInstance()->getApplicationName());
+
+            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, og.header_len);
+                output->write (og.body, og.body_len);
+            }
+
+            ok = true;
+        }
+    }
+
+    ~OggWriter()
+    {
+        if (ok)
+        {
+            // write a zero-length packet to show ogg that we're finished..
+            write (0, 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 = 0; // 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)
+    {
+        if (! ok)
+            return false;
+
+        if (numSamples > 0)
+        {
+            const double gain = 1.0 / 0x80000000u;
+            float** const vorbisBuffer = vorbis_analysis_buffer (&vd, numSamples);
+
+            for (int i = numChannels; --i >= 0;)
+            {
+                float* const dst = vorbisBuffer[i];
+                const int* const src = samplesToWrite [i];
+
+                if (src != 0 && dst != 0)
+                {
+                    for (int j = 0; j < numSamples; ++j)
+                        dst[j] = (float) (src[j] * gain);
+                }
+            }
+        }
+
+        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, og.header_len);
+                    output->write (og.body, og.body_len);
+
+                    if (ogg_page_eos (&og))
+                        break;
+                }
+            }
+        }
+
+        return true;
+    }
+
+    juce_UseDebuggingNewOperator
+};
+
+
+//==============================================================================
+OggVorbisAudioFormat::OggVorbisAudioFormat()
+    : AudioFormat (oggFormatName, (const tchar**) oggExtensions)
+{
+}
+
+OggVorbisAudioFormat::~OggVorbisAudioFormat()
+{
+}
+
+const Array <int> OggVorbisAudioFormat::getPossibleSampleRates()
+{
+    const int rates[] = { 22050, 32000, 44100, 48000, 0 };
+    return Array <int> (rates);
+}
+
+const Array <int> OggVorbisAudioFormat::getPossibleBitDepths()
+{
+    Array <int> depths;
+    depths.add (32);
+    return depths;
+}
+
+bool OggVorbisAudioFormat::canDoStereo()
+{
+    return true;
+}
+
+bool OggVorbisAudioFormat::canDoMono()
+{
+    return true;
+}
+
+AudioFormatReader* OggVorbisAudioFormat::createReaderFor (InputStream* in,
+                                                          const bool deleteStreamIfOpeningFails)
+{
+    OggReader* r = new OggReader (in);
+
+    if (r->sampleRate == 0)
+    {
+        if (! deleteStreamIfOpeningFails)
+            r->input = 0;
+
+        deleteAndZero (r);
+    }
+
+    return r;
+}
+
+AudioFormatWriter* OggVorbisAudioFormat::createWriterFor (OutputStream* out,
+                                                          double sampleRate,
+                                                          unsigned int numChannels,
+                                                          int bitsPerSample,
+                                                          const StringPairArray& /*metadataValues*/,
+                                                          int qualityOptionIndex)
+{
+    OggWriter* w = new OggWriter (out,
+                                  sampleRate,
+                                  numChannels,
+                                  bitsPerSample,
+                                  qualityOptionIndex);
+
+    if (! w->ok)
+        deleteAndZero (w);
+
+    return w;
+}
+
+bool OggVorbisAudioFormat::isCompressed()
+{
+    return true;
+}
+
+const StringArray OggVorbisAudioFormat::getQualityOptions()
+{
+    StringArray s;
+    s.add ("Low Quality");
+    s.add ("Medium Quality");
+    s.add ("High Quality");
+    return s;
+}
+
+int OggVorbisAudioFormat::estimateOggFileQuality (const File& source)
+{
+    FileInputStream* const in = source.createInputStream();
+
+    if (in != 0)
+    {
+        AudioFormatReader* const r = createReaderFor (in, true);
+
+        if (r != 0)
+        {
+            const int64 numSamps = r->lengthInSamples;
+            delete r;
+
+            const int64 fileNumSamps = source.getSize() / 4;
+            const double ratio = numSamps / (double) fileNumSamps;
+
+            if (ratio > 12.0)
+                return 0;
+            else if (ratio > 6.0)
+                return 1;
+            else
+                return 2;
+        }
+    }
+
+    return 1;
+}
+
+END_JUCE_NAMESPACE
+
+#endif

src/audio/audio_file_formats/juce_OggVorbisAudioFormat.h

@@ -0,0 +1,88 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#ifndef __JUCE_OGGVORBISAUDIOFORMAT_JUCEHEADER__
+#define __JUCE_OGGVORBISAUDIOFORMAT_JUCEHEADER__
+
+#include "juce_AudioFormat.h" // (must keep this outside the conditional define)
+
+#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 in juce_Config.h,
+    and make sure your include search path and library search path are set up to find
+    the Vorbis and Ogg header files and static libraries.
+
+    @see AudioFormat,
+*/
+class JUCE_API  OggVorbisAudioFormat : public AudioFormat
+{
+public:
+    //==============================================================================
+    OggVorbisAudioFormat();
+    ~OggVorbisAudioFormat();
+
+    //==============================================================================
+    const Array <int> getPossibleSampleRates();
+    const Array <int> getPossibleBitDepths();
+    bool canDoStereo();
+    bool canDoMono();
+    bool isCompressed();
+    const StringArray getQualityOptions();
+
+    //==============================================================================
+    /** 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);
+
+
+    //==============================================================================
+    AudioFormatReader* createReaderFor (InputStream* sourceStream,
+                                        const bool deleteStreamIfOpeningFails);
+
+    AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
+                                        double sampleRateToUse,
+                                        unsigned int numberOfChannels,
+                                        int bitsPerSample,
+                                        const StringPairArray& metadataValues,
+                                        int qualityOptionIndex);
+
+    //==============================================================================
+    juce_UseDebuggingNewOperator
+};
+
+
+#endif
+#endif   // __JUCE_OGGVORBISAUDIOFORMAT_JUCEHEADER__

src/audio/audio_file_formats/juce_QuickTimeAudioFormat.cpp

@@ -0,0 +1,407 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#include "../../../juce_Config.h"
+
+#if JUCE_QUICKTIME
+
+#if ! defined (_WIN32)
+ #include <QuickTime/Movies.h>
+ #include <QuickTime/QTML.h>
+ #include <QuickTime/QuickTimeComponents.h>
+ #include <QuickTime/MediaHandlers.h>
+ #include <QuickTime/ImageCodec.h>
+#else
+ #ifdef _MSC_VER
+  #pragma warning (push)
+  #pragma warning (disable : 4100)
+ #endif
+
+ /* If you've got an include error here, you probably need to install the QuickTime SDK and
+    add its header directory to your include path.
+
+    Alternatively, if you don't need any QuickTime services, just turn off the JUC_QUICKTIME
+    flag in juce_Config.h
+ */
+ #include <Movies.h>
+ #include <QTML.h>
+ #include <QuickTimeComponents.h>
+ #include <MediaHandlers.h>
+ #include <ImageCodec.h>
+
+ #ifdef _MSC_VER
+   #pragma warning (pop)
+ #endif
+#endif
+
+#include "../../core/juce_StandardHeader.h"
+
+#if ! (JUCE_MAC && JUCE_64BIT)
+
+BEGIN_JUCE_NAMESPACE
+
+#include "juce_QuickTimeAudioFormat.h"
+#include "../../text/juce_LocalisedStrings.h"
+#include "../../threads/juce_Thread.h"
+#include "../../io/network/juce_URL.h"
+
+bool juce_OpenQuickTimeMovieFromStream (InputStream* input, Movie& movie, Handle& dataHandle);
+
+#define quickTimeFormatName                          TRANS("QuickTime file")
+static const tchar* const quickTimeExtensions[] =    { T(".mov"), T(".mp3"), T(".mp4"), 0 };
+
+//==============================================================================
+class QTAudioReader     : public AudioFormatReader
+{
+public:
+    QTAudioReader (InputStream* const input_, const int trackNum_)
+        : AudioFormatReader (input_, quickTimeFormatName),
+          ok (false),
+          movie (0),
+          trackNum (trackNum_),
+          extractor (0),
+          lastSampleRead (0),
+          lastThreadId (0),
+          dataHandle (0)
+    {
+        bufferList = (AudioBufferList*) juce_calloc (256);
+
+#ifdef WIN32
+        if (InitializeQTML (0) != noErr)
+            return;
+#endif
+        if (EnterMovies() != noErr)
+            return;
+
+        bool opened = juce_OpenQuickTimeMovieFromStream (input_, movie, dataHandle);
+
+        if (! opened)
+            return;
+
+        {
+            const int numTracks = GetMovieTrackCount (movie);
+            int trackCount = 0;
+
+            for (int i = 1; i <= numTracks; ++i)
+            {
+                track = GetMovieIndTrack (movie, i);
+                media = GetTrackMedia (track);
+
+                OSType mediaType;
+                GetMediaHandlerDescription (media, &mediaType, 0, 0);
+
+                if (mediaType == SoundMediaType
+                     && trackCount++ == trackNum_)
+                {
+                    ok = true;
+                    break;
+                }
+            }
+        }
+
+        if (! ok)
+            return;
+
+        ok = false;
+
+        lengthInSamples = GetMediaDecodeDuration (media);
+        usesFloatingPointData = false;
+
+        samplesPerFrame = (int) (GetMediaDecodeDuration (media) / GetMediaSampleCount (media));
+
+        trackUnitsPerFrame = GetMovieTimeScale (movie) * samplesPerFrame
+                                / GetMediaTimeScale (media);
+
+        OSStatus err = MovieAudioExtractionBegin (movie, 0, &extractor);
+
+        unsigned long output_layout_size;
+        err = MovieAudioExtractionGetPropertyInfo (extractor,
+                                                   kQTPropertyClass_MovieAudioExtraction_Audio,
+                                                   kQTMovieAudioExtractionAudioPropertyID_AudioChannelLayout,
+                                                   0, &output_layout_size, 0);
+        if (err != noErr)
+            return;
+
+        AudioChannelLayout* const qt_audio_channel_layout
+                = (AudioChannelLayout*) juce_calloc (output_layout_size);
+
+        err = MovieAudioExtractionGetProperty (extractor,
+                                               kQTPropertyClass_MovieAudioExtraction_Audio,
+                                               kQTMovieAudioExtractionAudioPropertyID_AudioChannelLayout,
+                                               output_layout_size, qt_audio_channel_layout, 0);
+
+        qt_audio_channel_layout->mChannelLayoutTag = kAudioChannelLayoutTag_Stereo;
+
+        err = MovieAudioExtractionSetProperty (extractor,
+                                               kQTPropertyClass_MovieAudioExtraction_Audio,
+                                               kQTMovieAudioExtractionAudioPropertyID_AudioChannelLayout,
+                                               sizeof (qt_audio_channel_layout),
+                                               qt_audio_channel_layout);
+
+        juce_free (qt_audio_channel_layout);
+
+        err = MovieAudioExtractionGetProperty (extractor,
+                                               kQTPropertyClass_MovieAudioExtraction_Audio,
+                                               kQTMovieAudioExtractionAudioPropertyID_AudioStreamBasicDescription,
+                                               sizeof (inputStreamDesc),
+                                               &inputStreamDesc, 0);
+        if (err != noErr)
+            return;
+
+        inputStreamDesc.mFormatFlags = kAudioFormatFlagIsSignedInteger
+                                        | kAudioFormatFlagIsPacked
+                                        | kAudioFormatFlagsNativeEndian;
+        inputStreamDesc.mBitsPerChannel = sizeof (SInt16) * 8;
+        inputStreamDesc.mChannelsPerFrame = jmin (2, inputStreamDesc.mChannelsPerFrame);
+        inputStreamDesc.mBytesPerFrame = sizeof (SInt16) * inputStreamDesc.mChannelsPerFrame;
+        inputStreamDesc.mBytesPerPacket = inputStreamDesc.mBytesPerFrame;
+
+        err = MovieAudioExtractionSetProperty (extractor,
+                                               kQTPropertyClass_MovieAudioExtraction_Audio,
+                                               kQTMovieAudioExtractionAudioPropertyID_AudioStreamBasicDescription,
+                                               sizeof (inputStreamDesc),
+                                               &inputStreamDesc);
+        if (err != noErr)
+            return;
+
+        Boolean allChannelsDiscrete = false;
+        err = MovieAudioExtractionSetProperty (extractor,
+                                               kQTPropertyClass_MovieAudioExtraction_Movie,
+                                               kQTMovieAudioExtractionMoviePropertyID_AllChannelsDiscrete,
+                                               sizeof (allChannelsDiscrete),
+                                               &allChannelsDiscrete);
+
+        if (err != noErr)
+            return;
+
+        bufferList->mNumberBuffers = 1;
+        bufferList->mBuffers[0].mNumberChannels = inputStreamDesc.mChannelsPerFrame;
+        bufferList->mBuffers[0].mDataByteSize = (UInt32) (samplesPerFrame * inputStreamDesc.mBytesPerFrame) + 16;
+        bufferList->mBuffers[0].mData = malloc (bufferList->mBuffers[0].mDataByteSize);
+
+        sampleRate = inputStreamDesc.mSampleRate;
+        bitsPerSample = 16;
+        numChannels = inputStreamDesc.mChannelsPerFrame;
+
+        detachThread();
+        ok = true;
+    }
+
+    ~QTAudioReader()
+    {
+        if (dataHandle != 0)
+            DisposeHandle (dataHandle);
+
+        if (extractor != 0)
+        {
+            MovieAudioExtractionEnd (extractor);
+            extractor = 0;
+        }
+
+        checkThreadIsAttached();
+        DisposeMovie (movie);
+
+        juce_free (bufferList->mBuffers[0].mData);
+        juce_free (bufferList);
+
+#if JUCE_MAC
+        ExitMoviesOnThread ();
+#endif
+    }
+
+    bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                      int64 startSampleInFile, int numSamples)
+    {
+        checkThreadIsAttached();
+
+        while (numSamples > 0)
+        {
+            if (! loadFrame ((int) startSampleInFile))
+                return false;
+
+            const int numToDo = jmin (numSamples, samplesPerFrame);
+
+            for (int j = numDestChannels; --j >= 0;)
+            {
+                if (destSamples[j] != 0)
+                {
+                    const short* const src = ((const short*) bufferList->mBuffers[0].mData) + j;
+
+                    for (int i = 0; i < numToDo; ++i)
+                        destSamples[j][startOffsetInDestBuffer + i] = src [i << 1] << 16;
+                }
+            }
+
+            startOffsetInDestBuffer += numToDo;
+            startSampleInFile += numToDo;
+            numSamples -= numToDo;
+        }
+
+        detachThread();
+        return true;
+    }
+
+    bool loadFrame (const int sampleNum)
+    {
+        if (lastSampleRead != sampleNum)
+        {
+            TimeRecord time;
+            time.scale = (TimeScale) inputStreamDesc.mSampleRate;
+            time.base = 0;
+            time.value.hi = 0;
+            time.value.lo = (UInt32) sampleNum;
+
+            OSStatus err = MovieAudioExtractionSetProperty (extractor,
+                                                            kQTPropertyClass_MovieAudioExtraction_Movie,
+                                                            kQTMovieAudioExtractionMoviePropertyID_CurrentTime,
+                                                            sizeof (time), &time);
+
+            if (err != noErr)
+                return false;
+        }
+
+        bufferList->mBuffers[0].mDataByteSize = inputStreamDesc.mBytesPerFrame * samplesPerFrame;
+
+        UInt32 outFlags = 0;
+        UInt32 actualNumSamples = samplesPerFrame;
+        OSStatus err = MovieAudioExtractionFillBuffer (extractor, &actualNumSamples,
+                                                       bufferList, &outFlags);
+
+        lastSampleRead = sampleNum + samplesPerFrame;
+
+        return err == noErr;
+    }
+
+    juce_UseDebuggingNewOperator
+
+    bool ok;
+
+private:
+    Movie movie;
+    Media media;
+    Track track;
+    const int trackNum;
+    double trackUnitsPerFrame;
+    int samplesPerFrame;
+    int lastSampleRead;
+    Thread::ThreadID lastThreadId;
+    MovieAudioExtractionRef extractor;
+    AudioStreamBasicDescription inputStreamDesc;
+    AudioBufferList* bufferList;
+    Handle dataHandle;
+
+    /*OSErr readMovieStream (long offset, long size, void* dataPtr)
+    {
+        input->setPosition (offset);
+        input->read (dataPtr, size);
+        return noErr;
+    }
+
+    static OSErr readMovieStreamProc (long offset, long size, void* dataPtr, void* userRef)
+    {
+        return ((QTAudioReader*) userRef)->readMovieStream (offset, size, dataPtr);
+    }*/
+
+    //==============================================================================
+    void checkThreadIsAttached()
+    {
+#if JUCE_MAC
+        if (Thread::getCurrentThreadId() != lastThreadId)
+            EnterMoviesOnThread (0);
+        AttachMovieToCurrentThread (movie);
+#endif
+    }
+
+    void detachThread()
+    {
+#if JUCE_MAC
+        DetachMovieFromCurrentThread (movie);
+#endif
+    }
+};
+
+
+//==============================================================================
+QuickTimeAudioFormat::QuickTimeAudioFormat()
+    : AudioFormat (quickTimeFormatName, (const tchar**) quickTimeExtensions)
+{
+}
+
+QuickTimeAudioFormat::~QuickTimeAudioFormat()
+{
+}
+
+const Array <int> QuickTimeAudioFormat::getPossibleSampleRates()
+{
+    return Array<int>();
+}
+
+const Array <int> QuickTimeAudioFormat::getPossibleBitDepths()
+{
+    return Array<int>();
+}
+
+bool QuickTimeAudioFormat::canDoStereo()
+{
+    return true;
+}
+
+bool QuickTimeAudioFormat::canDoMono()
+{
+    return true;
+}
+
+//==============================================================================
+AudioFormatReader* QuickTimeAudioFormat::createReaderFor (InputStream* sourceStream,
+                                                          const bool deleteStreamIfOpeningFails)
+{
+    QTAudioReader* r = new QTAudioReader (sourceStream, 0);
+
+    if (! r->ok)
+    {
+        if (! deleteStreamIfOpeningFails)
+            r->input = 0;
+
+        deleteAndZero (r);
+    }
+
+    return r;
+}
+
+AudioFormatWriter* QuickTimeAudioFormat::createWriterFor (OutputStream* /*streamToWriteTo*/,
+                                                          double /*sampleRateToUse*/,
+                                                          unsigned int /*numberOfChannels*/,
+                                                          int /*bitsPerSample*/,
+                                                          const StringPairArray& /*metadataValues*/,
+                                                          int /*qualityOptionIndex*/)
+{
+    jassertfalse // not yet implemented!
+    return 0;
+}
+
+END_JUCE_NAMESPACE
+
+#endif
+#endif

src/audio/audio_file_formats/juce_QuickTimeAudioFormat.h

@@ -0,0 +1,76 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#ifndef __JUCE_QUICKTIMEAUDIOFORMAT_JUCEHEADER__
+#define __JUCE_QUICKTIMEAUDIOFORMAT_JUCEHEADER__
+
+#include "juce_AudioFormat.h"
+#if JUCE_QUICKTIME
+
+
+//==============================================================================
+/**
+    Uses QuickTime to read the audio track a movie or media file.
+
+    As well as QuickTime movies, this should also manage to open other audio
+    files that quicktime can understand, like mp3, m4a, etc.
+
+    @see AudioFormat
+*/
+class JUCE_API  QuickTimeAudioFormat  : public AudioFormat
+{
+public:
+    //==============================================================================
+    /** Creates a format object. */
+    QuickTimeAudioFormat();
+
+    /** Destructor. */
+    ~QuickTimeAudioFormat();
+
+    //==============================================================================
+    const Array <int> getPossibleSampleRates();
+    const Array <int> getPossibleBitDepths();
+    bool canDoStereo();
+    bool canDoMono();
+
+    //==============================================================================
+    AudioFormatReader* createReaderFor (InputStream* sourceStream,
+                                        const bool deleteStreamIfOpeningFails);
+
+    AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
+                                        double sampleRateToUse,
+                                        unsigned int numberOfChannels,
+                                        int bitsPerSample,
+                                        const StringPairArray& metadataValues,
+                                        int qualityOptionIndex);
+
+
+    //==============================================================================
+    juce_UseDebuggingNewOperator
+};
+
+
+#endif
+#endif   // __JUCE_QUICKTIMEAUDIOFORMAT_JUCEHEADER__

src/audio/audio_file_formats/juce_WavAudioFormat.cpp

@@ -0,0 +1,863 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#include "../../core/juce_StandardHeader.h"
+
+BEGIN_JUCE_NAMESPACE
+
+#include "juce_WavAudioFormat.h"
+#include "../../io/streams/juce_BufferedInputStream.h"
+#include "../../text/juce_LocalisedStrings.h"
+#include "../../io/files/juce_FileInputStream.h"
+#include "../../io/files/juce_FileOutputStream.h"
+
+
+//==============================================================================
+#define wavFormatName                          TRANS("WAV file")
+static const tchar* const wavExtensions[] =    { T(".wav"), T(".bwf"), 0 };
+
+
+//==============================================================================
+const tchar* const WavAudioFormat::bwavDescription      = T("bwav description");
+const tchar* const WavAudioFormat::bwavOriginator       = T("bwav originator");
+const tchar* const WavAudioFormat::bwavOriginatorRef    = T("bwav originator ref");
+const tchar* const WavAudioFormat::bwavOriginationDate  = T("bwav origination date");
+const tchar* const WavAudioFormat::bwavOriginationTime  = T("bwav origination time");
+const tchar* const WavAudioFormat::bwavTimeReference    = T("bwav time reference");
+const tchar* const WavAudioFormat::bwavCodingHistory    = T("bwav coding history");
+
+const StringPairArray WavAudioFormat::createBWAVMetadata (const String& description,
+                                                          const String& originator,
+                                                          const String& originatorRef,
+                                                          const Time& date,
+                                                          const int64 timeReferenceSamples,
+                                                          const String& codingHistory)
+{
+    StringPairArray m;
+
+    m.set (bwavDescription, description);
+    m.set (bwavOriginator, originator);
+    m.set (bwavOriginatorRef, originatorRef);
+    m.set (bwavOriginationDate, date.formatted (T("%Y-%m-%d")));
+    m.set (bwavOriginationTime, date.formatted (T("%H:%M:%S")));
+    m.set (bwavTimeReference, String (timeReferenceSamples));
+    m.set (bwavCodingHistory, codingHistory);
+
+    return m;
+}
+
+
+//==============================================================================
+#if JUCE_MSVC
+  #pragma pack (push, 1)
+  #define PACKED
+#elif defined (JUCE_GCC)
+  #define PACKED __attribute__((packed))
+#else
+  #define PACKED
+#endif
+
+struct BWAVChunk
+{
+    uint8 description [256];
+    uint8 originator [32];
+    uint8 originatorRef [32];
+    uint8 originationDate [10];
+    uint8 originationTime [8];
+    uint32 timeRefLow;
+    uint32 timeRefHigh;
+    uint16 version;
+    uint8 umid[64];
+    uint8 reserved[190];
+    uint8 codingHistory[1];
+
+    void copyTo (StringPairArray& values) const
+    {
+        values.set (WavAudioFormat::bwavDescription, String::fromUTF8 (description, 256));
+        values.set (WavAudioFormat::bwavOriginator, String::fromUTF8 (originator, 32));
+        values.set (WavAudioFormat::bwavOriginatorRef, String::fromUTF8 (originatorRef, 32));
+        values.set (WavAudioFormat::bwavOriginationDate, String::fromUTF8 (originationDate, 10));
+        values.set (WavAudioFormat::bwavOriginationTime, String::fromUTF8 (originationTime, 8));
+
+        const uint32 timeLow = swapIfBigEndian (timeRefLow);
+        const uint32 timeHigh = swapIfBigEndian (timeRefHigh);
+        const int64 time = (((int64)timeHigh) << 32) + timeLow;
+
+        values.set (WavAudioFormat::bwavTimeReference, String (time));
+        values.set (WavAudioFormat::bwavCodingHistory, String::fromUTF8 (codingHistory));
+    }
+
+    static MemoryBlock createFrom (const StringPairArray& values)
+    {
+        const int sizeNeeded = sizeof (BWAVChunk) + values [WavAudioFormat::bwavCodingHistory].copyToUTF8 (0) - 1;
+        MemoryBlock data ((sizeNeeded + 3) & ~3);
+        data.fillWith (0);
+
+        BWAVChunk* b = (BWAVChunk*) data.getData();
+
+        // Allow these calls to overwrite an extra byte at the end, which is fine as long
+        // as they get called in the right order..
+        values [WavAudioFormat::bwavDescription].copyToUTF8 (b->description, 257);
+        values [WavAudioFormat::bwavOriginator].copyToUTF8 (b->originator, 33);
+        values [WavAudioFormat::bwavOriginatorRef].copyToUTF8 (b->originatorRef, 33);
+        values [WavAudioFormat::bwavOriginationDate].copyToUTF8 (b->originationDate, 11);
+        values [WavAudioFormat::bwavOriginationTime].copyToUTF8 (b->originationTime, 9);
+
+        const int64 time = values [WavAudioFormat::bwavTimeReference].getLargeIntValue();
+        b->timeRefLow = swapIfBigEndian ((uint32) (time & 0xffffffff));
+        b->timeRefHigh = swapIfBigEndian ((uint32) (time >> 32));
+
+        values [WavAudioFormat::bwavCodingHistory].copyToUTF8 (b->codingHistory);
+
+        if (b->description[0] != 0
+            || b->originator[0] != 0
+            || b->originationDate[0] != 0
+            || b->originationTime[0] != 0
+            || b->codingHistory[0] != 0
+            || time != 0)
+        {
+            return data;
+        }
+
+        return MemoryBlock();
+    }
+
+} PACKED;
+
+
+//==============================================================================
+struct SMPLChunk
+{
+    struct SampleLoop
+    {
+        uint32 identifier;
+        uint32 type;
+        uint32 start;
+        uint32 end;
+        uint32 fraction;
+        uint32 playCount;
+    } PACKED;
+
+    uint32 manufacturer;
+    uint32 product;
+    uint32 samplePeriod;
+    uint32 midiUnityNote;
+    uint32 midiPitchFraction;
+    uint32 smpteFormat;
+    uint32 smpteOffset;
+    uint32 numSampleLoops;
+    uint32 samplerData;
+    SampleLoop loops[1];
+
+    void copyTo (StringPairArray& values, const int totalSize) const
+    {
+        values.set (T("Manufacturer"),      String (swapIfBigEndian (manufacturer)));
+        values.set (T("Product"),           String (swapIfBigEndian (product)));
+        values.set (T("SamplePeriod"),      String (swapIfBigEndian (samplePeriod)));
+        values.set (T("MidiUnityNote"),     String (swapIfBigEndian (midiUnityNote)));
+        values.set (T("MidiPitchFraction"), String (swapIfBigEndian (midiPitchFraction)));
+        values.set (T("SmpteFormat"),       String (swapIfBigEndian (smpteFormat)));
+        values.set (T("SmpteOffset"),       String (swapIfBigEndian (smpteOffset)));
+        values.set (T("NumSampleLoops"),    String (swapIfBigEndian (numSampleLoops)));
+        values.set (T("SamplerData"),       String (swapIfBigEndian (samplerData)));
+
+        for (uint32 i = 0; i < numSampleLoops; ++i)
+        {
+            if ((uint8*) (loops + (i + 1)) > ((uint8*) this) + totalSize)
+                break;
+
+            values.set (String::formatted (T("Loop%dIdentifier"), i),   String (swapIfBigEndian (loops[i].identifier)));
+            values.set (String::formatted (T("Loop%dType"), i),         String (swapIfBigEndian (loops[i].type)));
+            values.set (String::formatted (T("Loop%dStart"), i),        String (swapIfBigEndian (loops[i].start)));
+            values.set (String::formatted (T("Loop%dEnd"), i),          String (swapIfBigEndian (loops[i].end)));
+            values.set (String::formatted (T("Loop%dFraction"), i),     String (swapIfBigEndian (loops[i].fraction)));
+            values.set (String::formatted (T("Loop%dPlayCount"), i),    String (swapIfBigEndian (loops[i].playCount)));
+        }
+    }
+} PACKED;
+
+#if JUCE_MSVC
+  #pragma pack (pop)
+#endif
+
+#undef PACKED
+
+#undef chunkName
+#define chunkName(a) ((int) littleEndianInt(a))
+
+//==============================================================================
+class WavAudioFormatReader  : public AudioFormatReader
+{
+    int bytesPerFrame;
+    int64 dataChunkStart, dataLength;
+
+    WavAudioFormatReader (const WavAudioFormatReader&);
+    const WavAudioFormatReader& operator= (const WavAudioFormatReader&);
+
+public:
+    int64 bwavChunkStart, bwavSize;
+
+    //==============================================================================
+    WavAudioFormatReader (InputStream* const in)
+        : AudioFormatReader (in, wavFormatName),
+          dataLength (0),
+          bwavChunkStart (0),
+          bwavSize (0)
+    {
+        if (input->readInt() == chunkName ("RIFF"))
+        {
+            const uint32 len = (uint32) input->readInt();
+            const int64 end = input->getPosition() + len;
+            bool hasGotType = false;
+            bool hasGotData = false;
+
+            if (input->readInt() == chunkName ("WAVE"))
+            {
+                while (input->getPosition() < end
+                        && ! input->isExhausted())
+                {
+                    const int chunkType = input->readInt();
+                    uint32 length = (uint32) input->readInt();
+                    const int64 chunkEnd = input->getPosition() + length + (length & 1);
+
+                    if (chunkType == chunkName ("fmt "))
+                    {
+                        // read the format chunk
+                        const short format = input->readShort();
+                        const short numChans = input->readShort();
+                        sampleRate = input->readInt();
+                        const int bytesPerSec = input->readInt();
+
+                        numChannels = numChans;
+                        bytesPerFrame = bytesPerSec / (int)sampleRate;
+                        bitsPerSample = 8 * bytesPerFrame / numChans;
+
+                        if (format == 3)
+                            usesFloatingPointData = true;
+                        else if (format != 1)
+                            bytesPerFrame = 0;
+
+                        hasGotType = true;
+                    }
+                    else if (chunkType == chunkName ("data"))
+                    {
+                        // get the data chunk's position
+                        dataLength = length;
+                        dataChunkStart = input->getPosition();
+                        lengthInSamples = (bytesPerFrame > 0) ? (dataLength / bytesPerFrame) : 0;
+
+                        hasGotData = true;
+                    }
+                    else if (chunkType == chunkName ("bext"))
+                    {
+                        bwavChunkStart = input->getPosition();
+                        bwavSize = length;
+
+                        // Broadcast-wav extension chunk..
+                        BWAVChunk* const bwav = (BWAVChunk*) juce_calloc (jmax (length + 1, (int) sizeof (BWAVChunk)));
+                        input->read (bwav, length);
+                        bwav->copyTo (metadataValues);
+                        juce_free (bwav);
+                    }
+                    else if (chunkType == chunkName ("smpl"))
+                    {
+                        SMPLChunk* const smpl = (SMPLChunk*) juce_calloc (jmax (length + 1, (int) sizeof (SMPLChunk)));
+                        input->read (smpl, length);
+                        smpl->copyTo (metadataValues, length);
+                        juce_free (smpl);
+                    }
+                    else if (chunkEnd <= input->getPosition())
+                    {
+                        break;
+                    }
+
+                    input->setPosition (chunkEnd);
+                }
+            }
+        }
+    }
+
+    ~WavAudioFormatReader()
+    {
+    }
+
+    //==============================================================================
+    bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
+                      int64 startSampleInFile, int numSamples)
+    {
+        numSamples = (int) jmin ((int64) numSamples, lengthInSamples - startSampleInFile);
+
+        if (numSamples <= 0)
+            return true;
+
+        input->setPosition (dataChunkStart + startSampleInFile * bytesPerFrame);
+
+        const int tempBufSize = 480 * 3 * 4; // (keep this a multiple of 3)
+        char tempBuffer [tempBufSize];
+
+        while (numSamples > 0)
+        {
+            int* left = destSamples[0];
+            if (left != 0)
+                left += startOffsetInDestBuffer;
+
+            int* right = numDestChannels > 1 ? destSamples[1] : 0;
+            if (right != 0)
+                right += startOffsetInDestBuffer;
+
+            const int numThisTime = jmin (tempBufSize / bytesPerFrame, numSamples);
+            const int bytesRead = input->read (tempBuffer, numThisTime * bytesPerFrame);
+
+            if (bytesRead < numThisTime * bytesPerFrame)
+                zeromem (tempBuffer + bytesRead, numThisTime * bytesPerFrame - bytesRead);
+
+            if (bitsPerSample == 16)
+            {
+                const short* src = (const short*) tempBuffer;
+
+                if (numChannels > 1)
+                {
+                    if (left == 0)
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            ++src;
+                            *right++ = (int) swapIfBigEndian ((unsigned short) *src++) << 16;
+                        }
+                    }
+                    else if (right == 0)
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *left++ = (int) swapIfBigEndian ((unsigned short) *src++) << 16;
+                            ++src;
+                        }
+                    }
+                    else
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *left++ = (int) swapIfBigEndian ((unsigned short) *src++) << 16;
+                            *right++ = (int) swapIfBigEndian ((unsigned short) *src++) << 16;
+                        }
+                    }
+                }
+                else
+                {
+                    for (int i = numThisTime; --i >= 0;)
+                    {
+                        *left++ = (int) swapIfBigEndian ((unsigned short) *src++) << 16;
+                    }
+                }
+            }
+            else if (bitsPerSample == 24)
+            {
+                const char* src = (const char*) tempBuffer;
+
+                if (numChannels > 1)
+                {
+                    if (left == 0)
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            src += 3;
+                            *right++ = littleEndian24Bit (src) << 8;
+                            src += 3;
+                        }
+                    }
+                    else if (right == 0)
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *left++ = littleEndian24Bit (src) << 8;
+                            src += 6;
+                        }
+                    }
+                    else
+                    {
+                        for (int i = 0; i < numThisTime; ++i)
+                        {
+                            *left++ = littleEndian24Bit (src) << 8;
+                            src += 3;
+                            *right++ = littleEndian24Bit (src) << 8;
+                            src += 3;
+                        }
+                    }
+                }
+                else
+                {
+                    for (int i = 0; i < numThisTime; ++i)
+                    {
+                        *left++ = littleEndian24Bit (src) << 8;
+                        src += 3;
+                    }
+                }
+            }
+            else if (bitsPerSample == 32)
+            {
+                const unsigned int* src = (const unsigned int*) tempBuffer;
+                unsigned int* l = (unsigned int*) left;
+                unsigned int* r = (unsigned int*) right;
+
+                if (numChannels > 1)
+                {
+                    if (l == 0)
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            ++src;
+                            *r++ = swapIfBigEndian (*src++);
+                        }
+                    }
+                    else if (r == 0)
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *l++ = swapIfBigEndian (*src++);
+                            ++src;
+                        }
+                    }
+                    else
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *l++ = swapIfBigEndian (*src++);
+                            *r++ = swapIfBigEndian (*src++);
+                        }
+                    }
+                }
+                else
+                {
+                    for (int i = numThisTime; --i >= 0;)
+                    {
+                        *l++ = swapIfBigEndian (*src++);
+                    }
+                }
+
+                left = (int*)l;
+                right = (int*)r;
+            }
+            else if (bitsPerSample == 8)
+            {
+                const unsigned char* src = (const unsigned char*) tempBuffer;
+
+                if (numChannels > 1)
+                {
+                    if (left == 0)
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            ++src;
+                            *right++ = ((int) *src++ - 128) << 24;
+                        }
+                    }
+                    else if (right == 0)
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *left++ = ((int) *src++ - 128) << 24;
+                            ++src;
+                        }
+                    }
+                    else
+                    {
+                        for (int i = numThisTime; --i >= 0;)
+                        {
+                            *left++ = ((int) *src++ - 128) << 24;
+                            *right++ = ((int) *src++ - 128) << 24;
+                        }
+                    }
+                }
+                else
+                {
+                    for (int i = numThisTime; --i >= 0;)
+                    {
+                        *left++ = ((int)*src++ - 128) << 24;
+                    }
+                }
+            }
+
+            startOffsetInDestBuffer += numThisTime;
+            numSamples -= numThisTime;
+        }
+
+        if (numSamples > 0)
+        {
+            for (int i = numDestChannels; --i >= 0;)
+                if (destSamples[i] != 0)
+                    zeromem (destSamples[i] + startOffsetInDestBuffer,
+                             sizeof (int) * numSamples);
+        }
+
+        return true;
+    }
+
+    juce_UseDebuggingNewOperator
+};
+
+//==============================================================================
+class WavAudioFormatWriter  : public AudioFormatWriter
+{
+    MemoryBlock tempBlock, bwavChunk;
+    uint32 lengthInSamples, bytesWritten;
+    int64 headerPosition;
+    bool writeFailed;
+
+    WavAudioFormatWriter (const WavAudioFormatWriter&);
+    const WavAudioFormatWriter& operator= (const WavAudioFormatWriter&);
+
+    void writeHeader()
+    {
+        const bool seekedOk = output->setPosition (headerPosition);
+        (void) seekedOk;
+
+        // 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 (seekedOk);
+
+        const int bytesPerFrame = numChannels * bitsPerSample / 8;
+        output->writeInt (chunkName ("RIFF"));
+        output->writeInt (lengthInSamples * bytesPerFrame
+                            + ((bwavChunk.getSize() > 0) ? (44 + bwavChunk.getSize()) : 36));
+
+        output->writeInt (chunkName ("WAVE"));
+        output->writeInt (chunkName ("fmt "));
+        output->writeInt (16);
+        output->writeShort ((bitsPerSample < 32) ? (short) 1 /*WAVE_FORMAT_PCM*/
+                                                 : (short) 3 /*WAVE_FORMAT_IEEE_FLOAT*/);
+        output->writeShort ((short) numChannels);
+        output->writeInt ((int) sampleRate);
+        output->writeInt (bytesPerFrame * (int) sampleRate);
+        output->writeShort ((short) bytesPerFrame);
+        output->writeShort ((short) bitsPerSample);
+
+        if (bwavChunk.getSize() > 0)
+        {
+            output->writeInt (chunkName ("bext"));
+            output->writeInt (bwavChunk.getSize());
+            output->write (bwavChunk.getData(), bwavChunk.getSize());
+        }
+
+        output->writeInt (chunkName ("data"));
+        output->writeInt (lengthInSamples * bytesPerFrame);
+
+        usesFloatingPointData = (bitsPerSample == 32);
+    }
+
+public:
+    //==============================================================================
+    WavAudioFormatWriter (OutputStream* const out,
+                          const double sampleRate,
+                          const unsigned int numChannels_,
+                          const int bits,
+                          const StringPairArray& metadataValues)
+        : AudioFormatWriter (out,
+                             wavFormatName,
+                             sampleRate,
+                             numChannels_,
+                             bits),
+          lengthInSamples (0),
+          bytesWritten (0),
+          writeFailed (false)
+    {
+        if (metadataValues.size() > 0)
+            bwavChunk = BWAVChunk::createFrom (metadataValues);
+
+        headerPosition = out->getPosition();
+        writeHeader();
+    }
+
+    ~WavAudioFormatWriter()
+    {
+        writeHeader();
+    }
+
+    //==============================================================================
+    bool write (const int** data, int numSamples)
+    {
+        if (writeFailed)
+            return false;
+
+        const int bytes = numChannels * numSamples * bitsPerSample / 8;
+        tempBlock.ensureSize (bytes, false);
+        char* buffer = (char*) tempBlock.getData();
+
+        const int* left = data[0];
+        const int* right = data[1];
+        if (right == 0)
+            right = left;
+
+        if (bitsPerSample == 16)
+        {
+            short* b = (short*) buffer;
+
+            if (numChannels > 1)
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    *b++ = (short) swapIfBigEndian ((unsigned short) (*left++ >> 16));
+                    *b++ = (short) swapIfBigEndian ((unsigned short) (*right++ >> 16));
+                }
+            }
+            else
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    *b++ = (short) swapIfBigEndian ((unsigned short) (*left++ >> 16));
+                }
+            }
+        }
+        else if (bitsPerSample == 24)
+        {
+            char* b = (char*) buffer;
+
+            if (numChannels > 1)
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    littleEndian24BitToChars ((*left++) >> 8, b);
+                    b += 3;
+                    littleEndian24BitToChars ((*right++) >> 8, b);
+                    b += 3;
+                }
+            }
+            else
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    littleEndian24BitToChars ((*left++) >> 8, b);
+                    b += 3;
+                }
+            }
+        }
+        else if (bitsPerSample == 32)
+        {
+            unsigned int* b = (unsigned int*) buffer;
+
+            if (numChannels > 1)
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    *b++ = swapIfBigEndian ((unsigned int) *left++);
+                    *b++ = swapIfBigEndian ((unsigned int) *right++);
+                }
+            }
+            else
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    *b++ = swapIfBigEndian ((unsigned int) *left++);
+                }
+            }
+        }
+        else if (bitsPerSample == 8)
+        {
+            unsigned char* b = (unsigned char*) buffer;
+
+            if (numChannels > 1)
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    *b++ = (unsigned char) (128 + (*left++ >> 24));
+                    *b++ = (unsigned char) (128 + (*right++ >> 24));
+                }
+            }
+            else
+            {
+                for (int i = numSamples; --i >= 0;)
+                {
+                    *b++ = (unsigned char) (128 + (*left++ >> 24));
+                }
+            }
+        }
+
+        if (bytesWritten + bytes >= (uint32) 0xfff00000
+             || ! output->write (buffer, bytes))
+        {
+            // failed to write to disk, so let's try writing the header.
+            // If it's just run out of disk space, then if it does manage
+            // to write the header, we'll still have a useable file..
+            writeHeader();
+            writeFailed = true;
+            return false;
+        }
+        else
+        {
+            bytesWritten += bytes;
+            lengthInSamples += numSamples;
+
+            return true;
+        }
+    }
+
+    juce_UseDebuggingNewOperator
+};
+
+//==============================================================================
+WavAudioFormat::WavAudioFormat()
+    : AudioFormat (wavFormatName, (const tchar**) wavExtensions)
+{
+}
+
+WavAudioFormat::~WavAudioFormat()
+{
+}
+
+const Array <int> WavAudioFormat::getPossibleSampleRates()
+{
+    const int rates[] = { 22050, 32000, 44100, 48000, 88200, 96000, 176400, 192000, 0 };
+    return Array <int> (rates);
+}
+
+const Array <int> WavAudioFormat::getPossibleBitDepths()
+{
+    const int depths[] = { 8, 16, 24, 32, 0 };
+    return Array <int> (depths);
+}
+
+bool WavAudioFormat::canDoStereo()
+{
+    return true;
+}
+
+bool WavAudioFormat::canDoMono()
+{
+    return true;
+}
+
+AudioFormatReader* WavAudioFormat::createReaderFor (InputStream* sourceStream,
+                                                    const bool deleteStreamIfOpeningFails)
+{
+    WavAudioFormatReader* r = new WavAudioFormatReader (sourceStream);
+
+    if (r->sampleRate == 0)
+    {
+        if (! deleteStreamIfOpeningFails)
+            r->input = 0;
+
+        deleteAndZero (r);
+    }
+
+    return r;
+}
+
+AudioFormatWriter* WavAudioFormat::createWriterFor (OutputStream* out,
+                                                    double sampleRate,
+                                                    unsigned int numChannels,
+                                                    int bitsPerSample,
+                                                    const StringPairArray& metadataValues,
+                                                    int /*qualityOptionIndex*/)
+{
+    if (getPossibleBitDepths().contains (bitsPerSample))
+    {
+        return new WavAudioFormatWriter (out,
+                                         sampleRate,
+                                         numChannels,
+                                         bitsPerSample,
+                                         metadataValues);
+    }
+
+    return 0;
+}
+
+static bool juce_slowCopyOfWavFileWithNewMetadata (const File& file, const StringPairArray& metadata)
+{
+    bool ok = false;
+    WavAudioFormat wav;
+
+    const File dest (file.getNonexistentSibling());
+
+    OutputStream* outStream = dest.createOutputStream();
+
+    if (outStream != 0)
+    {
+        AudioFormatReader* reader = wav.createReaderFor (file.createInputStream(), true);
+
+        if (reader != 0)
+        {
+            AudioFormatWriter* writer = wav.createWriterFor (outStream, reader->sampleRate,
+                                                             reader->numChannels, reader->bitsPerSample,
+                                                             metadata, 0);
+
+            if (writer != 0)
+            {
+                ok = writer->writeFromAudioReader (*reader, 0, -1);
+
+                outStream = 0;
+                delete writer;
+            }
+
+            delete reader;
+        }
+
+        delete outStream;
+    }
+
+    if (ok)
+        ok = dest.moveFileTo (file);
+
+    if (! ok)
+        dest.deleteFile();
+
+    return ok;
+}
+
+bool WavAudioFormat::replaceMetadataInFile (const File& wavFile, const StringPairArray& newMetadata)
+{
+    WavAudioFormatReader* reader = (WavAudioFormatReader*) createReaderFor (wavFile.createInputStream(), true);
+
+    if (reader != 0)
+    {
+        const int64 bwavPos = reader->bwavChunkStart;
+        const int64 bwavSize = reader->bwavSize;
+        delete reader;
+
+        if (bwavSize > 0)
+        {
+            MemoryBlock chunk = BWAVChunk::createFrom (newMetadata);
+
+            if (chunk.getSize() <= bwavSize)
+            {
+                // the new one will fit in the space available, so write it directly..
+                const int64 oldSize = wavFile.getSize();
+
+                FileOutputStream* out = wavFile.createOutputStream();
+                out->setPosition (bwavPos);
+                out->write (chunk.getData(), chunk.getSize());
+                out->setPosition (oldSize);
+                delete out;
+
+                jassert (wavFile.getSize() == oldSize);
+
+                return true;
+            }
+        }
+    }
+
+    return juce_slowCopyOfWavFileWithNewMetadata (wavFile, newMetadata);
+}
+
+
+END_JUCE_NAMESPACE

src/audio/audio_file_formats/juce_WavAudioFormat.h

@@ -0,0 +1,151 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+#ifndef __JUCE_WAVAUDIOFORMAT_JUCEHEADER__
+#define __JUCE_WAVAUDIOFORMAT_JUCEHEADER__
+
+#include "juce_AudioFormat.h"
+
+
+//==============================================================================
+/**
+    Reads and Writes WAV format audio files.
+
+    @see AudioFormat
+*/
+class JUCE_API  WavAudioFormat  : public AudioFormat
+{
+public:
+    //==============================================================================
+    /** Creates a format object. */
+    WavAudioFormat();
+
+    /** Destructor. */
+    ~WavAudioFormat();
+
+    //==============================================================================
+    /** Metadata property name used by wav readers and writers for adding
+        a BWAV chunk to the file.
+
+        @see AudioFormatReader::metadataValues, createWriterFor
+    */
+    static const tchar* const bwavDescription;
+
+    /** Metadata property name used by wav readers and writers for adding
+        a BWAV chunk to the file.
+
+        @see AudioFormatReader::metadataValues, createWriterFor
+    */
+    static const tchar* const bwavOriginator;
+
+    /** Metadata property name used by wav readers and writers for adding
+        a BWAV chunk to the file.
+
+        @see AudioFormatReader::metadataValues, createWriterFor
+    */
+    static const tchar* const bwavOriginatorRef;
+
+    /** Metadata property name used by wav readers and writers for adding
+        a BWAV chunk to the file.
+
+        Date format is: yyyy-mm-dd
+
+        @see AudioFormatReader::metadataValues, createWriterFor
+    */
+    static const tchar* const bwavOriginationDate;
+
+    /** Metadata property name used by wav readers and writers for adding
+        a BWAV chunk to the file.
+
+        Time format is: hh-mm-ss
+
+        @see AudioFormatReader::metadataValues, createWriterFor
+    */
+    static const tchar* const bwavOriginationTime;
+
+    /** Metadata property name used by wav readers and writers for adding
+        a BWAV chunk to the file.
+
+        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 tchar* const bwavTimeReference;
+
+    /** Metadata property name used by wav readers and writers for adding
+        a BWAV chunk to the file.
+
+        This is a
+
+        @see AudioFormatReader::metadataValues, createWriterFor
+    */
+    static const tchar* const bwavCodingHistory;
+
+    /** 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 const StringPairArray createBWAVMetadata (const String& description,
+                                                     const String& originator,
+                                                     const String& originatorRef,
+                                                     const Time& dateAndTime,
+                                                     const int64 timeReferenceSamples,
+                                                     const String& codingHistory);
+
+    //==============================================================================
+    const Array <int> getPossibleSampleRates();
+    const Array <int> getPossibleBitDepths();
+    bool canDoStereo();
+    bool canDoMono();
+
+    //==============================================================================
+    AudioFormatReader* createReaderFor (InputStream* sourceStream,
+                                        const bool deleteStreamIfOpeningFails);
+
+    AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
+                                        double sampleRateToUse,
+                                        unsigned int numberOfChannels,
+                                        int bitsPerSample,
+                                        const StringPairArray& metadataValues,
+                                        int qualityOptionIndex);
+
+    //==============================================================================
+    /** 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);
+
+    //==============================================================================
+    juce_UseDebuggingNewOperator
+};
+
+
+#endif   // __JUCE_WAVAUDIOFORMAT_JUCEHEADER__

src/audio/audio_file_formats/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.

src/audio/audio_file_formats/oggvorbis/bitwise.c

@@ -0,0 +1,789 @@
+/********************************************************************
+ *                                                                  *
+ * 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 $
+
+ ********************************************************************/
+
+/* We're 'LSb' endian; if we write a word but read individual bits,
+   then we'll read the lsb first */
+
+#include "juce_OggVorbisHeader.h"
+#if JUCE_USE_OGGVORBIS
+
+#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
+
+#endif

src/audio/audio_file_formats/oggvorbis/codec.h

@@ -0,0 +1,239 @@
+/********************************************************************
+ *                                                                  *
+ * 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 XIPHOPHORUS Company http://www.xiph.org/                  *
+
+ ********************************************************************
+
+ function: libvorbis codec headers
+ last mod: $Id: codec.h,v 1.1 2007/06/07 17:48:18 jules_rms Exp $
+
+ ********************************************************************/
+
+#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 independant 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, char *comment);
+extern void     vorbis_comment_add_tag(vorbis_comment *vc,
+				       const char *tag, char *contents);
+extern char    *vorbis_comment_query(vorbis_comment *vc, char *tag, int count);
+extern int      vorbis_comment_query_count(vorbis_comment *vc, 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);
+
+/* 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_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

src/audio/audio_file_formats/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

src/audio/audio_file_formats/oggvorbis/juce_OggVorbisHeader.h

@@ -0,0 +1,34 @@
+/*
+  ==============================================================================
+
+   This file is part of the JUCE library - "Jules' Utility Class Extensions"
+   Copyright 2004-9 by Raw Material Software Ltd.
+
+  ------------------------------------------------------------------------------
+
+   JUCE can be redistributed and/or modified under the terms of the GNU General
+   Public License (Version 2), as published by the Free Software Foundation.
+   A copy of the license is included in the JUCE distribution, or can be found
+   online at www.gnu.org/licenses.
+
+   JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
+   WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
+   A PARTICULAR PURPOSE.  See the GNU General Public License for more details.
+
+  ------------------------------------------------------------------------------
+
+   To release a closed-source product which uses JUCE, commercial licenses are
+   available: visit www.rawmaterialsoftware.com/juce for more information.
+
+  ==============================================================================
+*/
+
+// This file is included at the start of each Ogg-Vorbis .c file, just to do a few housekeeping
+// tasks..
+
+
+#include "../../../../juce_Config.h"
+
+#ifdef _MSC_VER
+  #pragma warning (disable: 4267 4127 4244 4996 4100 4701 4702 4013 4133 4206 4305 4189 4706)
+#endif

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/AUTHORS

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

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/COPYING

@@ -0,0 +1,28 @@
+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.

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/README

@@ -0,0 +1,132 @@
+********************************************************************
+*                                                                  *
+* 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 1994-2004             *
+* 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 
+http://svn.xiph.org/trunk/vorbis/. You will also need the 
+newest versions of autoconf, automake, and libtool 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.
+
+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.
+

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/analysis.c

@@ -0,0 +1,113 @@
+/********************************************************************
+ *                                                                  *
+ * 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 XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: single-block PCM analysis mode dispatch
+ last mod: $Id: analysis.c,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+#include "../../juce_OggVorbisHeader.h"
+#if JUCE_USE_OGGVORBIS
+
+#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"
+
+int analysis_noisy=1;
+
+/* 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);
+}
+
+/* there was no great place to put this.... */
+void _analysis_output_always(const char *base,int i,float *v,int n,int bark,int dB,ogg_int64_t off){
+  int j;
+  FILE *of;
+  char buffer[80];
+
+  /*  if(i==5870){*/
+    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

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/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 1994-2002             *
+ * by the XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: libvorbis backend and mapping structures; needed for
+           static mode headers
+ last mod: $Id: backends.h,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+/* 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 *,
+			float **,int *,int);
+  int  (*forward)      (oggpack_buffer *,struct vorbis_block *,
+			vorbis_look_residue *,
+			float **,float **,int *,int,long **);
+  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    groupbook;        /* huffbook for partitioning */
+  int    secondstages[64]; /* expanded out to pointers in lookup */
+  int    booklist[256];    /* list of second stage books */
+
+  float  classmetric1[64];
+  float  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

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/bitrate.c

@@ -0,0 +1,259 @@
+/********************************************************************
+ *                                                                  *
+ * 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 XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: bitrate tracking and management
+ last mod: $Id: bitrate.c,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+#include "../../juce_OggVorbisHeader.h"
+#if JUCE_USE_OGGVORBIS
+
+
+#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= rint(1.*bi->avg_rate*halfsamples/ratesamples);
+    bm->min_bitsper= rint(1.*bi->min_rate*halfsamples/ratesamples);
+    bm->max_bitsper= 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=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=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=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=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);
+}
+
+#endif

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/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 1994-2002             *
+ * by the XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: bitrate tracking and management
+ last mod: $Id: bitrate.h,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+#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

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/block.c

@@ -0,0 +1,983 @@
+/********************************************************************
+ *                                                                  *
+ * 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 1994-2003             *
+ * by the XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: PCM data vector blocking, windowing and dis/reassembly
+ last mod: $Id: block.c,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ Handle windowing, overlap-add, etc of the PCM vectors.  This is made
+ more amusing by Vorbis' current two allowed block sizes.
+
+ ********************************************************************/
+
+#include "../../juce_OggVorbisHeader.h"
+#if JUCE_USE_OGGVORBIS
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include "../../ogg.h"
+#include "../../codec.h"
+#include "codec_internal.h"
+
+#include "window.h"
+#include "mdct.h"
+#include "lpc.h"
+#include "registry.h"
+#include "misc.h"
+
+/* pcm accumulator examples (not exhaustive):
+
+ <-------------- lW ---------------->
+                   <--------------- W ---------------->
+:            .....|.....       _______________         |
+:        .'''     |     '''_---      |       |\        |
+:.....'''         |_____--- '''......|       | \_______|
+:.................|__________________|_______|__|______|
+                  |<------ Sl ------>|      > Sr <     |endW
+                  |beginSl           |endSl  |  |endSr
+                  |beginW            |endlW  |beginSr
+
+
+                      |< lW >|
+                   <--------------- W ---------------->
+                  |   |  ..  ______________            |
+                  |   | '  `/        |     ---_        |
+                  |___.'___/`.       |         ---_____|
+                  |_______|__|_______|_________________|
+                  |      >|Sl|<      |<------ Sr ----->|endW
+                  |       |  |endSl  |beginSr          |endSr
+                  |beginW |  |endlW
+                  mult[0] |beginSl                     mult[n]
+
+ <-------------- lW ----------------->
+                          |<--W-->|
+:            ..............  ___  |   |
+:        .'''             |`/   \ |   |
+:.....'''                 |/`....\|...|
+:.........................|___|___|___|
+                          |Sl |Sr |endW
+                          |   |   |endSr
+                          |   |beginSr
+                          |   |endSl
+			  |beginSl
+			  |beginW
+*/
+
+/* block abstraction setup *********************************************/
+
+#ifndef WORD_ALIGN
+#define WORD_ALIGN 8
+#endif
+
+int vorbis_block_init(vorbis_dsp_state *v, vorbis_block *vb){
+  int i;
+  memset(vb,0,sizeof(*vb));
+  vb->vd=v;
+  vb->localalloc=0;
+  vb->localstore=NULL;
+  if(v->analysisp){
+    vorbis_block_internal *vbi=(vorbis_block_internal*)
+      (vb->internal=(vorbis_block_internal*)_ogg_calloc(1,sizeof(vorbis_block_internal)));
+    vbi->ampmax=-9999;
+
+    for(i=0;i<PACKETBLOBS;i++){
+      if(i==PACKETBLOBS/2){
+	vbi->packetblob[i]=&vb->opb;
+      }else{
+	vbi->packetblob[i]=
+	  (oggpack_buffer*) _ogg_calloc(1,sizeof(oggpack_buffer));
+      }
+      oggpack_writeinit(vbi->packetblob[i]);
+    }
+  }
+
+  return(0);
+}
+
+void *_vorbis_block_alloc(vorbis_block *vb,long bytes){
+  bytes=(bytes+(WORD_ALIGN-1)) & ~(WORD_ALIGN-1);
+  if(bytes+vb->localtop>vb->localalloc){
+    /* can't just _ogg_realloc... there are outstanding pointers */
+    if(vb->localstore){
+      struct alloc_chain *link=(struct alloc_chain*)_ogg_malloc(sizeof(*link));
+      vb->totaluse+=vb->localtop;
+      link->next=vb->reap;
+      link->ptr=vb->localstore;
+      vb->reap=link;
+    }
+    /* highly conservative */
+    vb->localalloc=bytes;
+    vb->localstore=_ogg_malloc(vb->localalloc);
+    vb->localtop=0;
+  }
+  {
+    void *ret=(void *)(((char *)vb->localstore)+vb->localtop);
+    vb->localtop+=bytes;
+    return ret;
+  }
+}
+
+/* reap the chain, pull the ripcord */
+void _vorbis_block_ripcord(vorbis_block *vb){
+  /* reap the chain */
+  struct alloc_chain *reap=vb->reap;
+  while(reap){
+    struct alloc_chain *next=reap->next;
+    _ogg_free(reap->ptr);
+    memset(reap,0,sizeof(*reap));
+    _ogg_free(reap);
+    reap=next;
+  }
+  /* consolidate storage */
+  if(vb->totaluse){
+    vb->localstore=_ogg_realloc(vb->localstore,vb->totaluse+vb->localalloc);
+    vb->localalloc+=vb->totaluse;
+    vb->totaluse=0;
+  }
+
+  /* pull the ripcord */
+  vb->localtop=0;
+  vb->reap=NULL;
+}
+
+int vorbis_block_clear(vorbis_block *vb){
+  int i;
+  vorbis_block_internal *vbi=(vorbis_block_internal*)vb->internal;
+
+  _vorbis_block_ripcord(vb);
+  if(vb->localstore)_ogg_free(vb->localstore);
+
+  if(vbi){
+    for(i=0;i<PACKETBLOBS;i++){
+      oggpack_writeclear(vbi->packetblob[i]);
+      if(i!=PACKETBLOBS/2)_ogg_free(vbi->packetblob[i]);
+    }
+    _ogg_free(vbi);
+  }
+  memset(vb,0,sizeof(*vb));
+  return(0);
+}
+
+/* Analysis side code, but directly related to blocking.  Thus it's
+   here and not in analysis.c (which is for analysis transforms only).
+   The init is here because some of it is shared */
+
+static int _vds_shared_init(vorbis_dsp_state *v,vorbis_info *vi,int encp){
+  int i;
+  codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
+  private_state *b=NULL;
+  int hs;
+
+  if(ci==NULL) return 1;
+  hs=ci->halfrate_flag;
+
+  memset(v,0,sizeof(*v));
+  b=(private_state*) (v->backend_state=(private_state*)_ogg_calloc(1,sizeof(*b)));
+
+  v->vi=vi;
+  b->modebits=ilog2(ci->modes);
+
+  b->transform[0]=(vorbis_look_transform**)_ogg_calloc(VI_TRANSFORMB,sizeof(*b->transform[0]));
+  b->transform[1]=(vorbis_look_transform**)_ogg_calloc(VI_TRANSFORMB,sizeof(*b->transform[1]));
+
+  /* MDCT is tranform 0 */
+
+  b->transform[0][0]=_ogg_calloc(1,sizeof(mdct_lookup));
+  b->transform[1][0]=_ogg_calloc(1,sizeof(mdct_lookup));
+  mdct_init((mdct_lookup*)b->transform[0][0],ci->blocksizes[0]>>hs);
+  mdct_init((mdct_lookup*)b->transform[1][0],ci->blocksizes[1]>>hs);
+
+  /* Vorbis I uses only window type 0 */
+  b->window[0]=ilog2(ci->blocksizes[0])-6;
+  b->window[1]=ilog2(ci->blocksizes[1])-6;
+
+  if(encp){ /* encode/decode differ here */
+
+    /* analysis always needs an fft */
+    drft_init(&b->fft_look[0],ci->blocksizes[0]);
+    drft_init(&b->fft_look[1],ci->blocksizes[1]);
+
+    /* finish the codebooks */
+    if(!ci->fullbooks){
+      ci->fullbooks=(codebook*) _ogg_calloc(ci->books,sizeof(*ci->fullbooks));
+      for(i=0;i<ci->books;i++)
+	vorbis_book_init_encode(ci->fullbooks+i,ci->book_param[i]);
+    }
+
+    b->psy=(vorbis_look_psy*)_ogg_calloc(ci->psys,sizeof(*b->psy));
+    for(i=0;i<ci->psys;i++){
+      _vp_psy_init(b->psy+i,
+		   ci->psy_param[i],
+		   &ci->psy_g_param,
+		   ci->blocksizes[ci->psy_param[i]->blockflag]/2,
+		   vi->rate);
+    }
+
+    v->analysisp=1;
+  }else{
+    /* finish the codebooks */
+    if(!ci->fullbooks){
+      ci->fullbooks=(codebook*) _ogg_calloc(ci->books,sizeof(*ci->fullbooks));
+      for(i=0;i<ci->books;i++){
+	vorbis_book_init_decode(ci->fullbooks+i,ci->book_param[i]);
+	/* decode codebooks are now standalone after init */
+	vorbis_staticbook_destroy(ci->book_param[i]);
+	ci->book_param[i]=NULL;
+      }
+    }
+  }
+
+  /* initialize the storage vectors. blocksize[1] is small for encode,
+     but the correct size for decode */
+  v->pcm_storage=ci->blocksizes[1];
+  v->pcm=(float**)_ogg_malloc(vi->channels*sizeof(*v->pcm));
+  v->pcmret=(float**)_ogg_malloc(vi->channels*sizeof(*v->pcmret));
+  {
+    int i;
+    for(i=0;i<vi->channels;i++)
+      v->pcm[i]=(float*)_ogg_calloc(v->pcm_storage,sizeof(*v->pcm[i]));
+  }
+
+  /* all 1 (large block) or 0 (small block) */
+  /* explicitly set for the sake of clarity */
+  v->lW=0; /* previous window size */
+  v->W=0;  /* current window size */
+
+  /* all vector indexes */
+  v->centerW=ci->blocksizes[1]/2;
+
+  v->pcm_current=v->centerW;
+
+  /* initialize all the backend lookups */
+  b->flr=(vorbis_look_floor**)_ogg_calloc(ci->floors,sizeof(*b->flr));
+  b->residue=(vorbis_look_residue**)_ogg_calloc(ci->residues,sizeof(*b->residue));
+
+  for(i=0;i<ci->floors;i++)
+    b->flr[i]=_floor_P[ci->floor_type[i]]->
+      look(v,ci->floor_param[i]);
+
+  for(i=0;i<ci->residues;i++)
+    b->residue[i]=_residue_P[ci->residue_type[i]]->
+      look(v,ci->residue_param[i]);
+
+  return 0;
+}
+
+/* arbitrary settings and spec-mandated numbers get filled in here */
+int vorbis_analysis_init(vorbis_dsp_state *v,vorbis_info *vi){
+  private_state *b=NULL;
+
+  if(_vds_shared_init(v,vi,1))return 1;
+  b=(private_state*)v->backend_state;
+  b->psy_g_look=_vp_global_look(vi);
+
+  /* Initialize the envelope state storage */
+  b->ve=(envelope_lookup*)_ogg_calloc(1,sizeof(*b->ve));
+  _ve_envelope_init(b->ve,vi);
+
+  vorbis_bitrate_init(vi,&b->bms);
+
+  /* compressed audio packets start after the headers
+     with sequence number 3 */
+  v->sequence=3;
+
+  return(0);
+}
+
+void vorbis_dsp_clear(vorbis_dsp_state *v){
+  int i;
+  if(v){
+    vorbis_info *vi=v->vi;
+    codec_setup_info *ci=(codec_setup_info*)(vi?vi->codec_setup:NULL);
+    private_state *b=(private_state*)v->backend_state;
+
+    if(b){
+
+      if(b->ve){
+	_ve_envelope_clear(b->ve);
+	_ogg_free(b->ve);
+      }
+
+      if(b->transform[0]){
+	mdct_clear((mdct_lookup*) b->transform[0][0]);
+	_ogg_free(b->transform[0][0]);
+	_ogg_free(b->transform[0]);
+      }
+      if(b->transform[1]){
+	mdct_clear((mdct_lookup*) b->transform[1][0]);
+	_ogg_free(b->transform[1][0]);
+	_ogg_free(b->transform[1]);
+      }
+
+      if(b->flr){
+	for(i=0;i<ci->floors;i++)
+	  _floor_P[ci->floor_type[i]]->
+	    free_look(b->flr[i]);
+	_ogg_free(b->flr);
+      }
+      if(b->residue){
+	for(i=0;i<ci->residues;i++)
+	  _residue_P[ci->residue_type[i]]->
+	    free_look(b->residue[i]);
+	_ogg_free(b->residue);
+      }
+      if(b->psy){
+	for(i=0;i<ci->psys;i++)
+	  _vp_psy_clear(b->psy+i);
+	_ogg_free(b->psy);
+      }
+
+      if(b->psy_g_look)_vp_global_free(b->psy_g_look);
+      vorbis_bitrate_clear(&b->bms);
+
+      drft_clear(&b->fft_look[0]);
+      drft_clear(&b->fft_look[1]);
+
+    }
+
+    if(v->pcm){
+      for(i=0;i<vi->channels;i++)
+	if(v->pcm[i])_ogg_free(v->pcm[i]);
+      _ogg_free(v->pcm);
+      if(v->pcmret)_ogg_free(v->pcmret);
+    }
+
+    if(b){
+      /* free header, header1, header2 */
+      if(b->header)_ogg_free(b->header);
+      if(b->header1)_ogg_free(b->header1);
+      if(b->header2)_ogg_free(b->header2);
+      _ogg_free(b);
+    }
+
+    memset(v,0,sizeof(*v));
+  }
+}
+
+float **vorbis_analysis_buffer(vorbis_dsp_state *v, int vals){
+  int i;
+  vorbis_info *vi=v->vi;
+  private_state *b=(private_state*)v->backend_state;
+
+  /* free header, header1, header2 */
+  if(b->header)_ogg_free(b->header);b->header=NULL;
+  if(b->header1)_ogg_free(b->header1);b->header1=NULL;
+  if(b->header2)_ogg_free(b->header2);b->header2=NULL;
+
+  /* Do we have enough storage space for the requested buffer? If not,
+     expand the PCM (and envelope) storage */
+
+  if(v->pcm_current+vals>=v->pcm_storage){
+    v->pcm_storage=v->pcm_current+vals*2;
+
+    for(i=0;i<vi->channels;i++){
+      v->pcm[i]=(float*)_ogg_realloc(v->pcm[i],v->pcm_storage*sizeof(*v->pcm[i]));
+    }
+  }
+
+  for(i=0;i<vi->channels;i++)
+    v->pcmret[i]=v->pcm[i]+v->pcm_current;
+
+  return(v->pcmret);
+}
+
+static void _preextrapolate_helper(vorbis_dsp_state *v){
+  int i;
+  int order=32;
+  float *lpc=(float*)alloca(order*sizeof(*lpc));
+  float *work=(float*)alloca(v->pcm_current*sizeof(*work));
+  long j;
+  v->preextrapolate=1;
+
+  if(v->pcm_current-v->centerW>order*2){ /* safety */
+    for(i=0;i<v->vi->channels;i++){
+      /* need to run the extrapolation in reverse! */
+      for(j=0;j<v->pcm_current;j++)
+	work[j]=v->pcm[i][v->pcm_current-j-1];
+
+      /* prime as above */
+      vorbis_lpc_from_data(work,lpc,v->pcm_current-v->centerW,order);
+
+      /* run the predictor filter */
+      vorbis_lpc_predict(lpc,work+v->pcm_current-v->centerW-order,
+			 order,
+			 work+v->pcm_current-v->centerW,
+			 v->centerW);
+
+      for(j=0;j<v->pcm_current;j++)
+	v->pcm[i][v->pcm_current-j-1]=work[j];
+
+    }
+  }
+}
+
+
+/* call with val<=0 to set eof */
+
+int vorbis_analysis_wrote(vorbis_dsp_state *v, int vals){
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
+
+  if(vals<=0){
+    int order=32;
+    int i;
+    float *lpc=(float*) alloca(order*sizeof(*lpc));
+
+    /* if it wasn't done earlier (very short sample) */
+    if(!v->preextrapolate)
+      _preextrapolate_helper(v);
+
+    /* We're encoding the end of the stream.  Just make sure we have
+       [at least] a few full blocks of zeroes at the end. */
+    /* actually, we don't want zeroes; that could drop a large
+       amplitude off a cliff, creating spread spectrum noise that will
+       suck to encode.  Extrapolate for the sake of cleanliness. */
+
+    vorbis_analysis_buffer(v,ci->blocksizes[1]*3);
+    v->eofflag=v->pcm_current;
+    v->pcm_current+=ci->blocksizes[1]*3;
+
+    for(i=0;i<vi->channels;i++){
+      if(v->eofflag>order*2){
+	/* extrapolate with LPC to fill in */
+	long n;
+
+	/* make a predictor filter */
+	n=v->eofflag;
+	if(n>ci->blocksizes[1])n=ci->blocksizes[1];
+	vorbis_lpc_from_data(v->pcm[i]+v->eofflag-n,lpc,n,order);
+
+	/* run the predictor filter */
+	vorbis_lpc_predict(lpc,v->pcm[i]+v->eofflag-order,order,
+			   v->pcm[i]+v->eofflag,v->pcm_current-v->eofflag);
+      }else{
+	/* not enough data to extrapolate (unlikely to happen due to
+           guarding the overlap, but bulletproof in case that
+           assumtion goes away). zeroes will do. */
+	memset(v->pcm[i]+v->eofflag,0,
+	       (v->pcm_current-v->eofflag)*sizeof(*v->pcm[i]));
+
+      }
+    }
+  }else{
+
+    if(v->pcm_current+vals>v->pcm_storage)
+      return(OV_EINVAL);
+
+    v->pcm_current+=vals;
+
+    /* we may want to reverse extrapolate the beginning of a stream
+       too... in case we're beginning on a cliff! */
+    /* clumsy, but simple.  It only runs once, so simple is good. */
+    if(!v->preextrapolate && v->pcm_current-v->centerW>ci->blocksizes[1])
+      _preextrapolate_helper(v);
+
+  }
+  return(0);
+}
+
+/* do the deltas, envelope shaping, pre-echo and determine the size of
+   the next block on which to continue analysis */
+int vorbis_analysis_blockout(vorbis_dsp_state *v,vorbis_block *vb){
+  int i;
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
+  private_state *b=(private_state*)v->backend_state;
+  vorbis_look_psy_global *g=b->psy_g_look;
+  long beginW=v->centerW-ci->blocksizes[v->W]/2,centerNext;
+  vorbis_block_internal *vbi=(vorbis_block_internal *)vb->internal;
+
+  /* check to see if we're started... */
+  if(!v->preextrapolate)return(0);
+
+  /* check to see if we're done... */
+  if(v->eofflag==-1)return(0);
+
+  /* By our invariant, we have lW, W and centerW set.  Search for
+     the next boundary so we can determine nW (the next window size)
+     which lets us compute the shape of the current block's window */
+
+  /* we do an envelope search even on a single blocksize; we may still
+     be throwing more bits at impulses, and envelope search handles
+     marking impulses too. */
+  {
+    long bp=_ve_envelope_search(v);
+    if(bp==-1){
+
+      if(v->eofflag==0)return(0); /* not enough data currently to search for a
+				     full long block */
+      v->nW=0;
+    }else{
+
+      if(ci->blocksizes[0]==ci->blocksizes[1])
+	v->nW=0;
+      else
+	v->nW=bp;
+    }
+  }
+
+  centerNext=v->centerW+ci->blocksizes[v->W]/4+ci->blocksizes[v->nW]/4;
+
+  {
+    /* center of next block + next block maximum right side. */
+
+    long blockbound=centerNext+ci->blocksizes[v->nW]/2;
+    if(v->pcm_current<blockbound)return(0); /* not enough data yet;
+                                               although this check is
+                                               less strict that the
+                                               _ve_envelope_search,
+                                               the search is not run
+                                               if we only use one
+                                               block size */
+
+
+  }
+
+  /* fill in the block.  Note that for a short window, lW and nW are *short*
+     regardless of actual settings in the stream */
+
+  _vorbis_block_ripcord(vb);
+  vb->lW=v->lW;
+  vb->W=v->W;
+  vb->nW=v->nW;
+
+  if(v->W){
+    if(!v->lW || !v->nW){
+      vbi->blocktype=BLOCKTYPE_TRANSITION;
+      /*fprintf(stderr,"-");*/
+    }else{
+      vbi->blocktype=BLOCKTYPE_LONG;
+      /*fprintf(stderr,"_");*/
+    }
+  }else{
+    if(_ve_envelope_mark(v)){
+      vbi->blocktype=BLOCKTYPE_IMPULSE;
+      /*fprintf(stderr,"|");*/
+
+    }else{
+      vbi->blocktype=BLOCKTYPE_PADDING;
+      /*fprintf(stderr,".");*/
+
+    }
+  }
+
+  vb->vd=v;
+  vb->sequence=v->sequence++;
+  vb->granulepos=v->granulepos;
+  vb->pcmend=ci->blocksizes[v->W];
+
+  /* copy the vectors; this uses the local storage in vb */
+
+  /* this tracks 'strongest peak' for later psychoacoustics */
+  /* moved to the global psy state; clean this mess up */
+  if(vbi->ampmax>g->ampmax)g->ampmax=vbi->ampmax;
+  g->ampmax=_vp_ampmax_decay(g->ampmax,v);
+  vbi->ampmax=g->ampmax;
+
+  vb->pcm=(float**)_vorbis_block_alloc(vb,sizeof(*vb->pcm)*vi->channels);
+  vbi->pcmdelay=(float**)_vorbis_block_alloc(vb,sizeof(*vbi->pcmdelay)*vi->channels);
+  for(i=0;i<vi->channels;i++){
+    vbi->pcmdelay[i]=
+      (float*) _vorbis_block_alloc(vb,(vb->pcmend+beginW)*sizeof(*vbi->pcmdelay[i]));
+    memcpy(vbi->pcmdelay[i],v->pcm[i],(vb->pcmend+beginW)*sizeof(*vbi->pcmdelay[i]));
+    vb->pcm[i]=vbi->pcmdelay[i]+beginW;
+
+    /* before we added the delay
+       vb->pcm[i]=_vorbis_block_alloc(vb,vb->pcmend*sizeof(*vb->pcm[i]));
+       memcpy(vb->pcm[i],v->pcm[i]+beginW,ci->blocksizes[v->W]*sizeof(*vb->pcm[i]));
+    */
+
+  }
+
+  /* handle eof detection: eof==0 means that we've not yet received EOF
+                           eof>0  marks the last 'real' sample in pcm[]
+                           eof<0  'no more to do'; doesn't get here */
+
+  if(v->eofflag){
+    if(v->centerW>=v->eofflag){
+      v->eofflag=-1;
+      vb->eofflag=1;
+      return(1);
+    }
+  }
+
+  /* advance storage vectors and clean up */
+  {
+    int new_centerNext=ci->blocksizes[1]/2;
+    int movementW=centerNext-new_centerNext;
+
+    if(movementW>0){
+
+      _ve_envelope_shift(b->ve,movementW);
+      v->pcm_current-=movementW;
+
+      for(i=0;i<vi->channels;i++)
+	memmove(v->pcm[i],v->pcm[i]+movementW,
+		v->pcm_current*sizeof(*v->pcm[i]));
+
+
+      v->lW=v->W;
+      v->W=v->nW;
+      v->centerW=new_centerNext;
+
+      if(v->eofflag){
+	v->eofflag-=movementW;
+	if(v->eofflag<=0)v->eofflag=-1;
+	/* do not add padding to end of stream! */
+	if(v->centerW>=v->eofflag){
+	  v->granulepos+=movementW-(v->centerW-v->eofflag);
+	}else{
+	  v->granulepos+=movementW;
+	}
+      }else{
+	v->granulepos+=movementW;
+      }
+    }
+  }
+
+  /* done */
+  return(1);
+}
+
+int vorbis_synthesis_restart(vorbis_dsp_state *v){
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci;
+  int hs;
+
+  if(!v->backend_state)return -1;
+  if(!vi)return -1;
+  ci=(codec_setup_info*) vi->codec_setup;
+  if(!ci)return -1;
+  hs=ci->halfrate_flag;
+
+  v->centerW=ci->blocksizes[1]>>(hs+1);
+  v->pcm_current=v->centerW>>hs;
+
+  v->pcm_returned=-1;
+  v->granulepos=-1;
+  v->sequence=-1;
+  v->eofflag=0;
+  ((private_state *)(v->backend_state))->sample_count=-1;
+
+  return(0);
+}
+
+int vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi){
+  if(_vds_shared_init(v,vi,0)) return 1;
+  vorbis_synthesis_restart(v);
+
+  return 0;
+}
+
+/* Unlike in analysis, the window is only partially applied for each
+   block.  The time domain envelope is not yet handled at the point of
+   calling (as it relies on the previous block). */
+
+int vorbis_synthesis_blockin(vorbis_dsp_state *v,vorbis_block *vb){
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
+  private_state *b=(private_state*)v->backend_state;
+  int hs=ci->halfrate_flag;
+  int i,j;
+
+  if(!vb)return(OV_EINVAL);
+  if(v->pcm_current>v->pcm_returned  && v->pcm_returned!=-1)return(OV_EINVAL);
+
+  v->lW=v->W;
+  v->W=vb->W;
+  v->nW=-1;
+
+  if((v->sequence==-1)||
+     (v->sequence+1 != vb->sequence)){
+    v->granulepos=-1; /* out of sequence; lose count */
+    b->sample_count=-1;
+  }
+
+  v->sequence=vb->sequence;
+
+  if(vb->pcm){  /* no pcm to process if vorbis_synthesis_trackonly
+		   was called on block */
+    int n=ci->blocksizes[v->W]>>(hs+1);
+    int n0=ci->blocksizes[0]>>(hs+1);
+    int n1=ci->blocksizes[1]>>(hs+1);
+
+    int thisCenter;
+    int prevCenter;
+
+    v->glue_bits+=vb->glue_bits;
+    v->time_bits+=vb->time_bits;
+    v->floor_bits+=vb->floor_bits;
+    v->res_bits+=vb->res_bits;
+
+    if(v->centerW){
+      thisCenter=n1;
+      prevCenter=0;
+    }else{
+      thisCenter=0;
+      prevCenter=n1;
+    }
+
+    /* v->pcm is now used like a two-stage double buffer.  We don't want
+       to have to constantly shift *or* adjust memory usage.  Don't
+       accept a new block until the old is shifted out */
+
+    for(j=0;j<vi->channels;j++){
+      /* the overlap/add section */
+      if(v->lW){
+	if(v->W){
+	  /* large/large */
+	  float *w=_vorbis_window_get(b->window[1]-hs);
+	  float *pcm=v->pcm[j]+prevCenter;
+	  float *p=vb->pcm[j];
+	  for(i=0;i<n1;i++)
+	    pcm[i]=pcm[i]*w[n1-i-1] + p[i]*w[i];
+	}else{
+	  /* large/small */
+	  float *w=_vorbis_window_get(b->window[0]-hs);
+	  float *pcm=v->pcm[j]+prevCenter+n1/2-n0/2;
+	  float *p=vb->pcm[j];
+	  for(i=0;i<n0;i++)
+	    pcm[i]=pcm[i]*w[n0-i-1] +p[i]*w[i];
+	}
+      }else{
+	if(v->W){
+	  /* small/large */
+	  float *w=_vorbis_window_get(b->window[0]-hs);
+	  float *pcm=v->pcm[j]+prevCenter;
+	  float *p=vb->pcm[j]+n1/2-n0/2;
+	  for(i=0;i<n0;i++)
+	    pcm[i]=pcm[i]*w[n0-i-1] +p[i]*w[i];
+	  for(;i<n1/2+n0/2;i++)
+	    pcm[i]=p[i];
+	}else{
+	  /* small/small */
+	  float *w=_vorbis_window_get(b->window[0]-hs);
+	  float *pcm=v->pcm[j]+prevCenter;
+	  float *p=vb->pcm[j];
+	  for(i=0;i<n0;i++)
+	    pcm[i]=pcm[i]*w[n0-i-1] +p[i]*w[i];
+	}
+      }
+
+      /* the copy section */
+      {
+	float *pcm=v->pcm[j]+thisCenter;
+	float *p=vb->pcm[j]+n;
+	for(i=0;i<n;i++)
+	  pcm[i]=p[i];
+      }
+    }
+
+    if(v->centerW)
+      v->centerW=0;
+    else
+      v->centerW=n1;
+
+    /* deal with initial packet state; we do this using the explicit
+       pcm_returned==-1 flag otherwise we're sensitive to first block
+       being short or long */
+
+    if(v->pcm_returned==-1){
+      v->pcm_returned=thisCenter;
+      v->pcm_current=thisCenter;
+    }else{
+      v->pcm_returned=prevCenter;
+      v->pcm_current=prevCenter+
+	((ci->blocksizes[v->lW]/4+
+	ci->blocksizes[v->W]/4)>>hs);
+    }
+
+  }
+
+  /* track the frame number... This is for convenience, but also
+     making sure our last packet doesn't end with added padding.  If
+     the last packet is partial, the number of samples we'll have to
+     return will be past the vb->granulepos.
+
+     This is not foolproof!  It will be confused if we begin
+     decoding at the last page after a seek or hole.  In that case,
+     we don't have a starting point to judge where the last frame
+     is.  For this reason, vorbisfile will always try to make sure
+     it reads the last two marked pages in proper sequence */
+
+  if(b->sample_count==-1){
+    b->sample_count=0;
+  }else{
+    b->sample_count+=ci->blocksizes[v->lW]/4+ci->blocksizes[v->W]/4;
+  }
+
+  if(v->granulepos==-1){
+    if(vb->granulepos!=-1){ /* only set if we have a position to set to */
+
+      v->granulepos=vb->granulepos;
+
+      /* is this a short page? */
+      if(b->sample_count>v->granulepos){
+	/* corner case; if this is both the first and last audio page,
+	   then spec says the end is cut, not beginning */
+	if(vb->eofflag){
+	  /* trim the end */
+	  /* no preceeding granulepos; assume we started at zero (we'd
+	     have to in a short single-page stream) */
+	  /* granulepos could be -1 due to a seek, but that would result
+	     in a long count, not short count */
+
+	  v->pcm_current-=(b->sample_count-v->granulepos)>>hs;
+	}else{
+	  /* trim the beginning */
+	  v->pcm_returned+=(b->sample_count-v->granulepos)>>hs;
+	  if(v->pcm_returned>v->pcm_current)
+	    v->pcm_returned=v->pcm_current;
+	}
+
+      }
+
+    }
+  }else{
+    v->granulepos+=ci->blocksizes[v->lW]/4+ci->blocksizes[v->W]/4;
+    if(vb->granulepos!=-1 && v->granulepos!=vb->granulepos){
+
+      if(v->granulepos>vb->granulepos){
+	long extra=v->granulepos-vb->granulepos;
+
+	if(extra)
+	  if(vb->eofflag){
+	    /* partial last frame.  Strip the extra samples off */
+	    v->pcm_current-=extra>>hs;
+	  } /* else {Shouldn't happen *unless* the bitstream is out of
+	       spec.  Either way, believe the bitstream } */
+      } /* else {Shouldn't happen *unless* the bitstream is out of
+	   spec.  Either way, believe the bitstream } */
+      v->granulepos=vb->granulepos;
+    }
+  }
+
+  /* Update, cleanup */
+
+  if(vb->eofflag)v->eofflag=1;
+  return(0);
+
+}
+
+/* pcm==NULL indicates we just want the pending samples, no more */
+int vorbis_synthesis_pcmout(vorbis_dsp_state *v,float ***pcm){
+  vorbis_info *vi=v->vi;
+
+  if(v->pcm_returned>-1 && v->pcm_returned<v->pcm_current){
+    if(pcm){
+      int i;
+      for(i=0;i<vi->channels;i++)
+	v->pcmret[i]=v->pcm[i]+v->pcm_returned;
+      *pcm=v->pcmret;
+    }
+    return(v->pcm_current-v->pcm_returned);
+  }
+  return(0);
+}
+
+int vorbis_synthesis_read(vorbis_dsp_state *v,int n){
+  if(n && v->pcm_returned+n>v->pcm_current)return(OV_EINVAL);
+  v->pcm_returned+=n;
+  return(0);
+}
+
+/* intended for use with a specific vorbisfile feature; we want access
+   to the [usually synthetic/postextrapolated] buffer and lapping at
+   the end of a decode cycle, specifically, a half-short-block worth.
+   This funtion works like pcmout above, except it will also expose
+   this implicit buffer data not normally decoded. */
+int vorbis_synthesis_lapout(vorbis_dsp_state *v,float ***pcm){
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci=(codec_setup_info *)vi->codec_setup;
+  int hs=ci->halfrate_flag;
+
+  int n=ci->blocksizes[v->W]>>(hs+1);
+  int n0=ci->blocksizes[0]>>(hs+1);
+  int n1=ci->blocksizes[1]>>(hs+1);
+  int i,j;
+
+  if(v->pcm_returned<0)return 0;
+
+  /* our returned data ends at pcm_returned; because the synthesis pcm
+     buffer is a two-fragment ring, that means our data block may be
+     fragmented by buffering, wrapping or a short block not filling
+     out a buffer.  To simplify things, we unfragment if it's at all
+     possibly needed. Otherwise, we'd need to call lapout more than
+     once as well as hold additional dsp state.  Opt for
+     simplicity. */
+
+  /* centerW was advanced by blockin; it would be the center of the
+     *next* block */
+  if(v->centerW==n1){
+    /* the data buffer wraps; swap the halves */
+    /* slow, sure, small */
+    for(j=0;j<vi->channels;j++){
+      float *p=v->pcm[j];
+      for(i=0;i<n1;i++){
+	float temp=p[i];
+	p[i]=p[i+n1];
+	p[i+n1]=temp;
+      }
+    }
+
+    v->pcm_current-=n1;
+    v->pcm_returned-=n1;
+    v->centerW=0;
+  }
+
+  /* solidify buffer into contiguous space */
+  if((v->lW^v->W)==1){
+    /* long/short or short/long */
+    for(j=0;j<vi->channels;j++){
+      float *s=v->pcm[j];
+      float *d=v->pcm[j]+(n1-n0)/2;
+      for(i=(n1+n0)/2-1;i>=0;--i)
+	d[i]=s[i];
+    }
+    v->pcm_returned+=(n1-n0)/2;
+    v->pcm_current+=(n1-n0)/2;
+  }else{
+    if(v->lW==0){
+      /* short/short */
+      for(j=0;j<vi->channels;j++){
+	float *s=v->pcm[j];
+	float *d=v->pcm[j]+n1-n0;
+	for(i=n0-1;i>=0;--i)
+	  d[i]=s[i];
+      }
+      v->pcm_returned+=n1-n0;
+      v->pcm_current+=n1-n0;
+    }
+  }
+
+  if(pcm){
+    int i;
+    for(i=0;i<vi->channels;i++)
+      v->pcmret[i]=v->pcm[i]+v->pcm_returned;
+    *pcm=v->pcmret;
+  }
+
+  return(n1+n-v->pcm_returned);
+
+}
+
+float *vorbis_window(vorbis_dsp_state *v,int W){
+  vorbis_info *vi=v->vi;
+  codec_setup_info *ci=(codec_setup_info*) vi->codec_setup;
+  int hs=ci->halfrate_flag;
+  private_state *b=(private_state*)v->backend_state;
+
+  if(b->window[W]-1<0)return NULL;
+  return _vorbis_window_get(b->window[W]-hs);
+}
+
+#endif

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/codebook.c

@@ -0,0 +1,614 @@
+/********************************************************************
+ *                                                                  *
+ * 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 XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: basic codebook pack/unpack/code/decode operations
+ last mod: $Id: codebook.c,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+#include "../../juce_OggVorbisHeader.h"
+#if JUCE_USE_OGGVORBIS
+
+
+#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 *************/
+int vorbis_staticbook_unpack(oggpack_buffer *opb,static_codebook *s){
+  long i,j;
+  memset(s,0,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;
+
+  /* codeword ordering.... length ordered or unordered? */
+  switch((int)oggpack_read(opb,1)){
+  case 0:
+    /* unordered */
+    s->lengthlist=(long*)_ogg_malloc(sizeof(*s->lengthlist)*s->entries);
+
+    /* allocated but unused entries? */
+    if(oggpack_read(opb,1)){
+      /* 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;
+      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;
+	for(j=0;j<num && i<s->entries;j++,i++)
+	  s->lengthlist[i]=length;
+	length++;
+      }
+    }
+    break;
+  default:
+    /* EOF */
+    return(-1);
+  }
+
+  /* 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);
+
+    {
+      int quantvals=0;
+      switch(s->maptype){
+      case 1:
+	quantvals=_book_maptype1_quantvals(s);
+	break;
+      case 2:
+	quantvals=s->entries*s->dim;
+	break;
+      }
+
+      /* quantized values */
+      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(0);
+
+ _errout:
+ _eofout:
+  vorbis_staticbook_clear(s);
+  return(-1);
+}
+
+/* returns the number of bits ************************************************/
+int vorbis_book_encode(codebook *book, int a, oggpack_buffer *b){
+  oggpack_write(b,book->codelist[a],book->c->lengthlist[a]);
+  return(book->c->lengthlist[a]);
+}
+
+/* One the encode side, our vector writers are each designed for a
+specific purpose, and the encoder is not flexible without modification:
+
+The LSP vector coder uses a single stage nearest-match with no
+interleave, so no step and no error return.  This is specced by floor0
+and doesn't change.
+
+Residue0 encoding interleaves, uses multiple stages, and each stage
+peels of a specific amount of resolution from a lattice (thus we want
+to match by threshold, not nearest match).  Residue doesn't *have* to
+be encoded that way, but to change it, one will need to add more
+infrastructure on the encode side (decode side is specced and simpler) */
+
+/* floor0 LSP (single stage, non interleaved, nearest match) */
+/* returns entry number and *modifies a* to the quantization value *****/
+int vorbis_book_errorv(codebook *book,float *a){
+  int dim=book->dim,k;
+  int best=_best(book,a,1);
+  for(k=0;k<dim;k++)
+    a[k]=(book->valuelist+best*dim)[k];
+  return(best);
+}
+
+/* returns the number of bits and *modifies a* to the quantization value *****/
+int vorbis_book_encodev(codebook *book,int best,float *a,oggpack_buffer *b){
+  int k,dim=book->dim;
+  for(k=0;k<dim;k++)
+    a[k]=(book->valuelist+best*dim)[k];
+  return(vorbis_book_encode(book,best,b));
+}
+
+/* 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. */
+
+
+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=ogg_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){
+  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(packed_entry);
+}
+
+/* returns 0 on OK or -1 on eof *************************************/
+long vorbis_book_decodevs_add(codebook *book,float *a,oggpack_buffer *b,int n){
+  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){
+  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){
+  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++];
+  }
+  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;
+
+  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);
+}
+
+#ifdef _V_SELFTEST
+/* Simple enough; pack a few candidate codebooks, unpack them.  Code a
+   number of vectors through (keeping track of the quantized values),
+   and decode using the unpacked book.  quantized version of in should
+   exactly equal out */
+
+#include <stdio.h>
+
+#include "vorbis/book/lsp20_0.vqh"
+#include "vorbis/book/res0a_13.vqh"
+#define TESTSIZE 40
+
+float test1[TESTSIZE]={
+  0.105939f,
+  0.215373f,
+  0.429117f,
+  0.587974f,
+
+  0.181173f,
+  0.296583f,
+  0.515707f,
+  0.715261f,
+
+  0.162327f,
+  0.263834f,
+  0.342876f,
+  0.406025f,
+
+  0.103571f,
+  0.223561f,
+  0.368513f,
+  0.540313f,
+
+  0.136672f,
+  0.395882f,
+  0.587183f,
+  0.652476f,
+
+  0.114338f,
+  0.417300f,
+  0.525486f,
+  0.698679f,
+
+  0.147492f,
+  0.324481f,
+  0.643089f,
+  0.757582f,
+
+  0.139556f,
+  0.215795f,
+  0.324559f,
+  0.399387f,
+
+  0.120236f,
+  0.267420f,
+  0.446940f,
+  0.608760f,
+
+  0.115587f,
+  0.287234f,
+  0.571081f,
+  0.708603f,
+};
+
+float test3[TESTSIZE]={
+  0,1,-2,3,4,-5,6,7,8,9,
+  8,-2,7,-1,4,6,8,3,1,-9,
+  10,11,12,13,14,15,26,17,18,19,
+  30,-25,-30,-1,-5,-32,4,3,-2,0};
+
+static_codebook *testlist[]={&_vq_book_lsp20_0,
+			     &_vq_book_res0a_13,NULL};
+float   *testvec[]={test1,test3};
+
+int main(){
+  oggpack_buffer write;
+  oggpack_buffer read;
+  long ptr=0,i;
+  oggpack_writeinit(&write);
+
+  fprintf(stderr,"Testing codebook abstraction...:\n");
+
+  while(testlist[ptr]){
+    codebook c;
+    static_codebook s;
+    float *qv=alloca(sizeof(*qv)*TESTSIZE);
+    float *iv=alloca(sizeof(*iv)*TESTSIZE);
+    memcpy(qv,testvec[ptr],sizeof(*qv)*TESTSIZE);
+    memset(iv,0,sizeof(*iv)*TESTSIZE);
+
+    fprintf(stderr,"\tpacking/coding %ld... ",ptr);
+
+    /* pack the codebook, write the testvector */
+    oggpack_reset(&write);
+    vorbis_book_init_encode(&c,testlist[ptr]); /* get it into memory
+                                                  we can write */
+    vorbis_staticbook_pack(testlist[ptr],&write);
+    fprintf(stderr,"Codebook size %ld bytes... ",oggpack_bytes(&write));
+    for(i=0;i<TESTSIZE;i+=c.dim){
+      int best=_best(&c,qv+i,1);
+      vorbis_book_encodev(&c,best,qv+i,&write);
+    }
+    vorbis_book_clear(&c);
+
+    fprintf(stderr,"OK.\n");
+    fprintf(stderr,"\tunpacking/decoding %ld... ",ptr);
+
+    /* transfer the write data to a read buffer and unpack/read */
+    oggpack_readinit(&read,oggpack_get_buffer(&write),oggpack_bytes(&write));
+    if(vorbis_staticbook_unpack(&read,&s)){
+      fprintf(stderr,"Error unpacking codebook.\n");
+      exit(1);
+    }
+    if(vorbis_book_init_decode(&c,&s)){
+      fprintf(stderr,"Error initializing codebook.\n");
+      exit(1);
+    }
+
+    for(i=0;i<TESTSIZE;i+=c.dim)
+      if(vorbis_book_decodev_set(&c,iv+i,&read,c.dim)==-1){
+	fprintf(stderr,"Error reading codebook test data (EOP).\n");
+	exit(1);
+      }
+    for(i=0;i<TESTSIZE;i++)
+      if(fabs(qv[i]-iv[i])>.000001){
+	fprintf(stderr,"read (%g) != written (%g) at position (%ld)\n",
+		iv[i],qv[i],i);
+	exit(1);
+      }
+
+    fprintf(stderr,"OK\n");
+    ptr++;
+  }
+
+  /* The above is the trivial stuff; now try unquantizing a log scale codebook */
+
+  exit(0);
+}
+
+#endif
+
+#endif

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/codebook.h

@@ -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 1994-2002             *
+ * by the XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: basic shared codebook operations
+ last mod: $Id: codebook.h,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+#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
+			*/
+
+  /* encode helpers ********************************************************/
+  struct encode_aux_nearestmatch *nearest_tree;
+  struct encode_aux_threshmatch  *thresh_tree;
+  struct encode_aux_pigeonhole  *pigeon_tree;
+
+  int allocedp;
+} static_codebook;
+
+/* this structures an arbitrary trained book to quickly find the
+   nearest cell match */
+typedef struct encode_aux_nearestmatch{
+  /* pre-calculated partitioning tree */
+  long   *ptr0;
+  long   *ptr1;
+
+  long   *p;         /* decision points (each is an entry) */
+  long   *q;         /* decision points (each is an entry) */
+  long   aux;        /* number of tree entries */
+  long   alloc;
+} encode_aux_nearestmatch;
+
+/* assumes a maptype of 1; encode side only, so that's OK */
+typedef struct encode_aux_threshmatch{
+  float *quantthresh;
+  long   *quantmap;
+  int     quantvals;
+  int     threshvals;
+} encode_aux_threshmatch;
+
+typedef struct encode_aux_pigeonhole{
+  float min;
+  float del;
+
+  int  mapentries;
+  int  quantvals;
+  long *pigeonmap;
+
+  long fittotal;
+  long *fitlist;
+  long *fitmap;
+  long *fitlength;
+} encode_aux_pigeonhole;
+
+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;
+
+} codebook;
+
+extern void vorbis_staticbook_clear(static_codebook *b);
+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 int vorbis_staticbook_unpack(oggpack_buffer *b,static_codebook *c);
+
+extern int vorbis_book_encode(codebook *book, int a, oggpack_buffer *b);
+extern int vorbis_book_errorv(codebook *book, float *a);
+extern int vorbis_book_encodev(codebook *book, int best,float *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

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/codec_internal.h

@@ -0,0 +1,156 @@
+/********************************************************************
+ *                                                                  *
+ * 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 XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: libvorbis codec headers
+ last mod: $Id: codec_internal.h,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+#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);
+
+#endif

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/envelope.c

@@ -0,0 +1,383 @@
+/********************************************************************
+ *                                                                  *
+ * 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 XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: PCM data envelope analysis
+ last mod: $Id: envelope.c,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+#include "../../juce_OggVorbisHeader.h"
+#if JUCE_USE_OGGVORBIS
+
+
+#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 pos){
+  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,j);
+    }
+
+    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=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;
+}
+
+
+#endif

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/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 1994-2002             *
+ * by the XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: PCM data envelope analysis and manipulation
+ last mod: $Id: envelope.h,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+#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

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/floor0.c

@@ -0,0 +1,227 @@
+/********************************************************************
+ *                                                                  *
+ * 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 XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: floor backend 0 implementation
+ last mod: $Id: floor0.c,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+#include "../../juce_OggVorbisHeader.h"
+#if JUCE_USE_OGGVORBIS
+
+
+#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;
+  }
+  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 */
+vorbis_func_floor floor0_exportbundle={
+  NULL,&floor0_unpack,&floor0_look,&floor0_free_info,
+  &floor0_free_look,&floor0_inverse1,&floor0_inverse2
+};
+
+
+#endif

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/highlevel.h

@@ -0,0 +1,56 @@
+/********************************************************************
+ *                                                                  *
+ * 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 XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: highlevel encoder setup struct seperated out for vorbisenc clarity
+ last mod: $Id: highlevel.h,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+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 {
+  void *setup;
+  int   set_in_stone;
+
+  double base_setting;
+  double long_setting;
+  double short_setting;
+  double impulse_noisetune;
+
+  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;
+
+  double stereo_point_setting;
+  double lowpass_kHz;
+
+  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;

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/info.c

@@ -0,0 +1,602 @@
+/********************************************************************
+ *                                                                  *
+ * 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 1994-2003             *
+ * by the XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+ function: maintain the info structure, info <-> header packets
+ last mod: $Id: info.c,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+#include "../../juce_OggVorbisHeader.h"
+#if JUCE_USE_OGGVORBIS
+
+/* 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"
+
+
+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,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, 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, 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, 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;
+    for(i=0;i<vc->comments;i++)
+      if(vc->user_comments[i])_ogg_free(vc->user_comments[i]);
+    if(vc->user_comments)_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 guarantted 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 */
+      _mapping_P[ci->map_type[i]]->free_info(ci->map_param[i]);
+
+    for(i=0;i<ci->floors;i++) /* unpack does the range checking */
+      _floor_P[ci->floor_type[i]]->free_info(ci->floor_param[i]);
+
+    for(i=0;i<ci->residues;i++) /* unpack does the range checking */
+      _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]<8)goto err_out;
+  if(ci->blocksizes[1]<ci->blocksizes[0])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;
+  vc->vendor=(char*)_ogg_calloc(vendorlen+1,1);
+  _v_readstring(opb,vc->vendor,vendorlen);
+  vc->comments=oggpack_read(opb,32);
+  if(vc->comments<0)goto err_out;
+  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;
+	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;
+  /*ci->book_param=_ogg_calloc(ci->books,sizeof(*ci->book_param));*/
+  for(i=0;i<ci->books;i++){
+    ci->book_param[i]=(static_codebook*)_ogg_calloc(1,sizeof(*ci->book_param[i]));
+    if(vorbis_staticbook_unpack(opb,ci->book_param[i]))goto err_out;
+  }
+
+  /* time backend settings; hooks are unused */
+  {
+    int times=oggpack_read(opb,6)+1;
+    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;
+  /*ci->floor_type=_ogg_malloc(ci->floors*sizeof(*ci->floor_type));*/
+  /*ci->floor_param=_ogg_calloc(ci->floors,sizeof(void *));*/
+  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;
+  /*ci->residue_type=_ogg_malloc(ci->residues*sizeof(*ci->residue_type));*/
+  /*ci->residue_param=_ogg_calloc(ci->residues,sizeof(void *));*/
+  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;
+  /*ci->map_type=_ogg_malloc(ci->maps*sizeof(*ci->map_type));*/
+  /*ci->map_param=_ogg_calloc(ci->maps,sizeof(void *));*/
+  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;
+  /*vi->mode_param=_ogg_calloc(vi->modes,sizeof(void *));*/
+  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(oggpack_read(opb,1)!=1)goto err_out; /* top level EOP check */
+
+  return(0);
+ err_out:
+  vorbis_info_clear(vi);
+  return(OV_EBADHEADER);
+}
+
+/* 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){
+  char temp[]="Xiph.Org libVorbis I 20050304";
+  int bytes = strlen(temp);
+
+  /* preamble */
+  oggpack_write(opb,0x03,8);
+  _v_writestring(opb,"vorbis", 6);
+
+  /* vendor */
+  oggpack_write(opb,bytes,32);
+  _v_writestring(opb,temp, 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:
+  oggpack_writeclear(&opb);
+  memset(op,0,sizeof(*op));
+  memset(op_comm,0,sizeof(*op_comm));
+  memset(op_code,0,sizeof(*op_code));
+
+  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>=0)
+    return((double)granulepos/v->vi->rate);
+  return(-1);
+}
+
+#endif

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/lookup.c

@@ -0,0 +1,99 @@
+/********************************************************************
+ *                                                                  *
+ * 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 XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+  function: lookup based functions
+  last mod: $Id: lookup.c,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+#include "../../juce_OggVorbisHeader.h"
+#if JUCE_USE_OGGVORBIS
+
+#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
+
+#endif

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/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 1994-2002             *
+ * by the XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+  function: lookup based functions
+  last mod: $Id: lookup.h,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+#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

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/lookup_data.h

@@ -0,0 +1,189 @@
+/********************************************************************
+ *                                                                  *
+ * 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 XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+  function: lookup data; generated by lookups.pl; edit there
+  last mod: $Id: lookup_data.h,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+#ifndef _V_LOOKUP_DATA_H_
+
+#ifdef FLOAT_LOOKUP
+#define COS_LOOKUP_SZ 128
+static 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 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 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
+static 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 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,
+};
+
+#ifdef INT_LOOKUP
+
+#define INVSQ_LOOKUP_I_SHIFT 10
+#define INVSQ_LOOKUP_I_MASK 1023
+static 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 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

src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/lookups.pl

@@ -0,0 +1,142 @@
+#!/usr/bin/perl
+print <<'EOD';
+/********************************************************************
+ *                                                                  *
+ * 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 XIPHOPHORUS Company http://www.xiph.org/                  *
+ *                                                                  *
+ ********************************************************************
+
+  function: lookup data; generated by lookups.pl; edit there
+  last mod: $Id: lookups.pl,v 1.1 2007/06/07 17:49:17 jules_rms Exp $
+
+ ********************************************************************/
+
+#ifndef _V_LOOKUP_DATA_H_
+
+#ifdef FLOAT_LOOKUP
+EOD
+
+$cos_sz=128;
+$invsq_sz=32;
+$invsq2exp_min=-32;
+$invsq2exp_max=32;
+
+$fromdB_sz=35;
+$fromdB_shift=5;
+$fromdB2_shift=3;
+
+$invsq_i_shift=10;
+$cos_i_shift=9;
+$delta_shift=6;
+
+print "#define COS_LOOKUP_SZ $cos_sz\n";
+print "static float COS_LOOKUP[COS_LOOKUP_SZ+1]={\n";
+
+for($i=0;$i<=$cos_sz;){
+    print "\t";
+    for($j=0;$j<4 && $i<=$cos_sz;$j++){
+	printf "%+.13f,", cos(3.14159265358979323846*($i++)/$cos_sz) ;
+    }
+    print "\n";
+}
+print "};\n\n";
+
+print "#define INVSQ_LOOKUP_SZ $invsq_sz\n";
+print "static float INVSQ_LOOKUP[INVSQ_LOOKUP_SZ+1]={\n";
+
+for($i=0;$i<=$invsq_sz;){
+    print "\t";
+    for($j=0;$j<4 && $i<=$invsq_sz;$j++){
+	my$indexmap=$i++/$invsq_sz*.5+.5;
+	printf "%.12f,", 1./sqrt($indexmap);
+    }
+    print "\n";
+}
+print "};\n\n";
+
+print "#define INVSQ2EXP_LOOKUP_MIN $invsq2exp_min\n";
+print "#define INVSQ2EXP_LOOKUP_MAX $invsq2exp_max\n";
+print "static float INVSQ2EXP_LOOKUP[INVSQ2EXP_LOOKUP_MAX-\\\n".
+      "                              INVSQ2EXP_LOOKUP_MIN+1]={\n";
+
+for($i=$invsq2exp_min;$i<=$invsq2exp_max;){
+    print "\t";
+    for($j=0;$j<4 && $i<=$invsq2exp_max;$j++){
+	printf "%15.10g,", 2**($i++*-.5);
+    }
+    print "\n";
+}
+print "};\n\n#endif\n\n";
+
+
+# 0 to -140 dB
+$fromdB2_sz=1<<$fromdB_shift;
+$fromdB_gran=1<<($fromdB_shift-$fromdB2_shift);
+print "#define FROMdB_LOOKUP_SZ $fromdB_sz\n";
+print "#define FROMdB2_LOOKUP_SZ $fromdB2_sz\n";
+print "#define FROMdB_SHIFT $fromdB_shift\n";
+print "#define FROMdB2_SHIFT $fromdB2_shift\n";
+print "#define FROMdB2_MASK ".((1<<$fromdB_shift)-1)."\n";
+
+print "static float FROMdB_LOOKUP[FROMdB_LOOKUP_SZ]={\n";
+
+for($i=0;$i<$fromdB_sz;){
+    print "\t";
+    for($j=0;$j<4 && $i<$fromdB_sz;$j++){
+	printf "%15.10g,", 10**(.05*(-$fromdB_gran*$i++));
+    }
+    print "\n";
+}
+print "};\n\n";
+
+print "static float FROMdB2_LOOKUP[FROMdB2_LOOKUP_SZ]={\n";
+
+for($i=0;$i<$fromdB2_sz;){
+    print "\t";
+    for($j=0;$j<4 && $i<$fromdB_sz;$j++){
+	printf "%15.10g,", 10**(.05*(-$fromdB_gran/$fromdB2_sz*(.5+$i++)));
+    }
+    print "\n";
+}
+print "};\n\n#ifdef INT_LOOKUP\n\n";
+
+
+$iisz=0x10000>>$invsq_i_shift;
+print "#define INVSQ_LOOKUP_I_SHIFT $invsq_i_shift\n";
+print "#define INVSQ_LOOKUP_I_MASK ".(0x0ffff>>(16-$invsq_i_shift))."\n";
+print "static long INVSQ_LOOKUP_I[$iisz+1]={\n";
+for($i=0;$i<=$iisz;){
+    print "\t";
+    for($j=0;$j<4 && $i<=$iisz;$j++){
+	my$indexmap=$i++/$iisz*.5+.5;
+	printf "%8d,", int(1./sqrt($indexmap)*65536.+.5);
+    }
+    print "\n";
+}
+print "};\n\n";
+
+$cisz=0x10000>>$cos_i_shift;
+print "#define COS_LOOKUP_I_SHIFT $cos_i_shift\n";
+print "#define COS_LOOKUP_I_MASK ".(0x0ffff>>(16-$cos_i_shift))."\n";
+print "#define COS_LOOKUP_I_SZ $cisz\n";
+print "static long COS_LOOKUP_I[COS_LOOKUP_I_SZ+1]={\n";
+
+for($i=0;$i<=$cisz;){
+    print "\t";
+    for($j=0;$j<4 && $i<=$cisz;$j++){
+	printf "%8d,", int(cos(3.14159265358979323846*($i++)/$cos_sz)*16384.+.5) ;
+    }
+    print "\n";
+}
+print "};\n\n";
+
+
+print "#endif\n\n#endif\n";
+
+