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49
src/audio/audio_file_formats/flac/Flac Licence.txt Normal file
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=====================================================================
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.

402
src/audio/audio_file_formats/flac/all.h Normal file
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/* 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

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src/audio/audio_file_formats/flac/alloc.h Normal file
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/* 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

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/* 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

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/* 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

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/* 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

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/* 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

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/* 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

423
src/audio/audio_file_formats/flac/libFLAC/cpu.c Normal file
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@ -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

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/* 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

441
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/* 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

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/* 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

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/* 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

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/* 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

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/* 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

99
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/* 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

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/* 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

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/* 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

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/* 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

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/* 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

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/* 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

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/* 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

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/* 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

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#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

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/* 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

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/* 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

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/* 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

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/* 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

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/* 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

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/* 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

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/* 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

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

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#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

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/* 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

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/* 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

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/* 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

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/* 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

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

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/*
==============================================================================
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__

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/*
==============================================================================
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__

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

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/*
==============================================================================
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__

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

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/*
==============================================================================
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__

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

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/*
==============================================================================
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__

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/*
==============================================================================
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__

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/*
==============================================================================
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__

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

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/*
==============================================================================
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__

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

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/*
==============================================================================
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__

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

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/*
==============================================================================
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__

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

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/*
==============================================================================
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__

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

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/*
==============================================================================
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__

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

76
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/*
==============================================================================
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__

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

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src/audio/audio_file_formats/juce_WavAudioFormat.h Normal file
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/*
==============================================================================
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__

47
src/audio/audio_file_formats/oggvorbis/Ogg Vorbis Licence.txt Normal file
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=====================================================================
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.

789
src/audio/audio_file_formats/oggvorbis/bitwise.c Normal file
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/********************************************************************
* *
* 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

239
src/audio/audio_file_formats/oggvorbis/codec.h Normal file
View file

@ -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

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#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

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

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Monty <monty@xiph.org>
and the rest of the Xiph.org Foundation.

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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.

132
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********************************************************************
* *
* 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.

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/********************************************************************
* *
* 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

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/********************************************************************
* *
* 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

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/********************************************************************
* *
* 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

59
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/********************************************************************
* *
* 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

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/********************************************************************
* *
* 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

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/********************************************************************
* *
* 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

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/********************************************************************
* *
* 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

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/********************************************************************
* *
* 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

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/********************************************************************
* *
* 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

80
src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/envelope.h Normal file
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/********************************************************************
* *
* 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

227
src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/floor0.c Normal file
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/********************************************************************
* *
* 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

1074
src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/floor1.c Normal file
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File diff suppressed because it is too large Load diff

56
src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/highlevel.h Normal file
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/********************************************************************
* *
* 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;

602
src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/info.c Normal file
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/********************************************************************
* *
* 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

99
src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/lookup.c Normal file
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@ -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

32
src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/lookup.h Normal file
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@ -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

189
src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/lookup_data.h Normal file
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@ -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

142
src/audio/audio_file_formats/oggvorbis/libvorbis-1.1.2/lib/lookups.pl Normal file
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@ -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";

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