jens/JaySynth
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity

- port to minGw

Browse Source 
git-svn-id: http://moon:8086/svn/software/trunk/projects/JaySynth@235 b431acfa-c32f-4a4a-93f1-934dc6c82436
This commit is contained in:
jens
2015-02-21 17:09:43 +00:00
parent 843c6e300c
commit cfc9645b0b
1093 changed files with 426078 additions and 143 deletions
Download Patch File Download Diff File Expand all files Collapse all files
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/Flac Licence.txt
+49
View File
@@ -0,0 +1,49 @@
=====================================================================
I've incorporated FLAC directly into the Juce codebase because it makes
things much easier than having to make all your builds link correctly to
the appropriate libraries on every different platform.
I've made minimal changes to the FLAC code - just tweaked a few include paths
to make it build smoothly, added some headers to allow you to turn off FLAC
compilation, and commented-out a couple of unused bits of code.
=====================================================================
The following license is the BSD-style license that comes with the
Flac distribution, and which applies just to the files I've
included in this directory. For more info, and to get the rest of the
distribution, visit the Flac homepage: flac.sourceforge.net
=====================================================================
Copyright (C) 2000,2001,2002,2003,2004,2005,2006 Josh Coalson
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
- Neither the name of the Xiph.org Foundation nor the names of its
contributors may be used to endorse or promote products derived from
this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/all.h
+371
View File
@@ -0,0 +1,371 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__ALL_H
#define FLAC__ALL_H
#include "export.h"
#include "assert.h"
#include "callback.h"
#include "format.h"
#include "metadata.h"
#include "ordinals.h"
#include "stream_decoder.h"
#include "stream_encoder.h"
/** \mainpage
*
* \section intro Introduction
*
* This is the documentation for the FLAC C and C++ APIs. It is
* highly interconnected; this introduction should give you a top
* level idea of the structure and how to find the information you
* need. As a prerequisite you should have at least a basic
* knowledge of the FLAC format, documented
* <A HREF="../format.html">here</A>.
*
* \section c_api FLAC C API
*
* The FLAC C API is the interface to libFLAC, a set of structures
* describing the components of FLAC streams, and functions for
* encoding and decoding streams, as well as manipulating FLAC
* metadata in files. The public include files will be installed
* in your include area (for example /usr/include/FLAC/...).
*
* By writing a little code and linking against libFLAC, it is
* relatively easy to add FLAC support to another program. The
* library is licensed under <A HREF="../license.html">Xiph's BSD license</A>.
* Complete source code of libFLAC as well as the command-line
* encoder and plugins is available and is a useful source of
* examples.
*
* Aside from encoders and decoders, libFLAC provides a powerful
* metadata interface for manipulating metadata in FLAC files. It
* allows the user to add, delete, and modify FLAC metadata blocks
* and it can automatically take advantage of PADDING blocks to avoid
* rewriting the entire FLAC file when changing the size of the
* metadata.
*
* libFLAC usually only requires the standard C library and C math
* library. In particular, threading is not used so there is no
* dependency on a thread library. However, libFLAC does not use
* global variables and should be thread-safe.
*
* libFLAC also supports encoding to and decoding from Ogg FLAC.
* However the metadata editing interfaces currently have limited
* read-only support for Ogg FLAC files.
*
* \section cpp_api FLAC C++ API
*
* The FLAC C++ API is a set of classes that encapsulate the
* structures and functions in libFLAC. They provide slightly more
* functionality with respect to metadata but are otherwise
* equivalent. For the most part, they share the same usage as
* their counterparts in libFLAC, and the FLAC C API documentation
* can be used as a supplement. The public include files
* for the C++ API will be installed in your include area (for
* example /usr/include/FLAC++/...).
*
* libFLAC++ is also licensed under
* <A HREF="../license.html">Xiph's BSD license</A>.
*
* \section getting_started Getting Started
*
* A good starting point for learning the API is to browse through
* the <A HREF="modules.html">modules</A>. Modules are logical
* groupings of related functions or classes, which correspond roughly
* to header files or sections of header files. Each module includes a
* detailed description of the general usage of its functions or
* classes.
*
* From there you can go on to look at the documentation of
* individual functions. You can see different views of the individual
* functions through the links in top bar across this page.
*
* If you prefer a more hands-on approach, you can jump right to some
* <A HREF="../documentation_example_code.html">example code</A>.
*
* \section porting_guide Porting Guide
*
* Starting with FLAC 1.1.3 a \link porting Porting Guide \endlink
* has been introduced which gives detailed instructions on how to
* port your code to newer versions of FLAC.
*
* \section embedded_developers Embedded Developers
*
* libFLAC has grown larger over time as more functionality has been
* included, but much of it may be unnecessary for a particular embedded
* implementation. Unused parts may be pruned by some simple editing of
* src/libFLAC/Makefile.am. In general, the decoders, encoders, and
* metadata interface are all independent from each other.
*
* It is easiest to just describe the dependencies:
*
* - All modules depend on the \link flac_format Format \endlink module.
* - The decoders and encoders depend on the bitbuffer.
* - The decoder is independent of the encoder. The encoder uses the
* decoder because of the verify feature, but this can be removed if
* not needed.
* - Parts of the metadata interface require the stream decoder (but not
* the encoder).
* - Ogg support is selectable through the compile time macro
* \c FLAC__HAS_OGG.
*
* For example, if your application only requires the stream decoder, no
* encoder, and no metadata interface, you can remove the stream encoder
* and the metadata interface, which will greatly reduce the size of the
* library.
*
* Also, there are several places in the libFLAC code with comments marked
* with "OPT:" where a #define can be changed to enable code that might be
* faster on a specific platform. Experimenting with these can yield faster
* binaries.
*/
/** \defgroup porting Porting Guide for New Versions
*
* This module describes differences in the library interfaces from
* version to version. It assists in the porting of code that uses
* the libraries to newer versions of FLAC.
*
* One simple facility for making porting easier that has been added
* in FLAC 1.1.3 is a set of \c #defines in \c export.h of each
* library's includes (e.g. \c include/FLAC/export.h). The
* \c #defines mirror the libraries'
* <A HREF="http://www.gnu.org/software/libtool/manual/libtool.html#Libtool-versioning">libtool version numbers</A>,
* e.g. in libFLAC there are \c FLAC_API_VERSION_CURRENT,
* \c FLAC_API_VERSION_REVISION, and \c FLAC_API_VERSION_AGE.
* These can be used to support multiple versions of an API during the
* transition phase, e.g.
*
* \code
* #if !defined(FLAC_API_VERSION_CURRENT) || FLAC_API_VERSION_CURRENT <= 7
* legacy code
* #else
* new code
* #endif
* \endcode
*
* The the source will work for multiple versions and the legacy code can
* easily be removed when the transition is complete.
*
* Another available symbol is FLAC_API_SUPPORTS_OGG_FLAC (defined in
* include/FLAC/export.h), which can be used to determine whether or not
* the library has been compiled with support for Ogg FLAC. This is
* simpler than trying to call an Ogg init function and catching the
* error.
*/
/** \defgroup porting_1_1_2_to_1_1_3 Porting from FLAC 1.1.2 to 1.1.3
* \ingroup porting
*
* \brief
* This module describes porting from FLAC 1.1.2 to FLAC 1.1.3.
*
* The main change between the APIs in 1.1.2 and 1.1.3 is that they have
* been simplified. First, libOggFLAC has been merged into libFLAC and
* libOggFLAC++ has been merged into libFLAC++. Second, both the three
* decoding layers and three encoding layers have been merged into a
* single stream decoder and stream encoder. That is, the functionality
* of FLAC__SeekableStreamDecoder and FLAC__FileDecoder has been merged
* into FLAC__StreamDecoder, and FLAC__SeekableStreamEncoder and
* FLAC__FileEncoder into FLAC__StreamEncoder. Only the
* FLAC__StreamDecoder and FLAC__StreamEncoder remain. What this means
* is there is now a single API that can be used to encode or decode
* streams to/from native FLAC or Ogg FLAC and the single API can work
* on both seekable and non-seekable streams.
*
* Instead of creating an encoder or decoder of a certain layer, now the
* client will always create a FLAC__StreamEncoder or
* FLAC__StreamDecoder. The old layers are now differentiated by the
* initialization function. For example, for the decoder,
* FLAC__stream_decoder_init() has been replaced by
* FLAC__stream_decoder_init_stream(). This init function takes
* callbacks for the I/O, and the seeking callbacks are optional. This
* allows the client to use the same object for seekable and
* non-seekable streams. For decoding a FLAC file directly, the client
* can use FLAC__stream_decoder_init_file() and pass just a filename
* and fewer callbacks; most of the other callbacks are supplied
* internally. For situations where fopen()ing by filename is not
* possible (e.g. Unicode filenames on Windows) the client can instead
* open the file itself and supply the FILE* to
* FLAC__stream_decoder_init_FILE(). The init functions now returns a
* FLAC__StreamDecoderInitStatus instead of FLAC__StreamDecoderState.
* Since the callbacks and client data are now passed to the init
* function, the FLAC__stream_decoder_set_*_callback() functions and
* FLAC__stream_decoder_set_client_data() are no longer needed. The
* rest of the calls to the decoder are the same as before.
*
* There are counterpart init functions for Ogg FLAC, e.g.
* FLAC__stream_decoder_init_ogg_stream(). All the rest of the calls
* and callbacks are the same as for native FLAC.
*
* As an example, in FLAC 1.1.2 a seekable stream decoder would have
* been set up like so:
*
* \code
* FLAC__SeekableStreamDecoder *decoder = FLAC__seekable_stream_decoder_new();
* if(decoder == NULL) do_something;
* FLAC__seekable_stream_decoder_set_md5_checking(decoder, true);
* [... other settings ...]
* FLAC__seekable_stream_decoder_set_read_callback(decoder, my_read_callback);
* FLAC__seekable_stream_decoder_set_seek_callback(decoder, my_seek_callback);
* FLAC__seekable_stream_decoder_set_tell_callback(decoder, my_tell_callback);
* FLAC__seekable_stream_decoder_set_length_callback(decoder, my_length_callback);
* FLAC__seekable_stream_decoder_set_eof_callback(decoder, my_eof_callback);
* FLAC__seekable_stream_decoder_set_write_callback(decoder, my_write_callback);
* FLAC__seekable_stream_decoder_set_metadata_callback(decoder, my_metadata_callback);
* FLAC__seekable_stream_decoder_set_error_callback(decoder, my_error_callback);
* FLAC__seekable_stream_decoder_set_client_data(decoder, my_client_data);
* if(FLAC__seekable_stream_decoder_init(decoder) != FLAC__SEEKABLE_STREAM_DECODER_OK) do_something;
* \endcode
*
* In FLAC 1.1.3 it is like this:
*
* \code
* FLAC__StreamDecoder *decoder = FLAC__stream_decoder_new();
* if(decoder == NULL) do_something;
* FLAC__stream_decoder_set_md5_checking(decoder, true);
* [... other settings ...]
* if(FLAC__stream_decoder_init_stream(
* decoder,
* my_read_callback,
* my_seek_callback, // or NULL
* my_tell_callback, // or NULL
* my_length_callback, // or NULL
* my_eof_callback, // or NULL
* my_write_callback,
* my_metadata_callback, // or NULL
* my_error_callback,
* my_client_data
* ) != FLAC__STREAM_DECODER_INIT_STATUS_OK) do_something;
* \endcode
*
* or you could do;
*
* \code
* [...]
* FILE *file = fopen("somefile.flac","rb");
* if(file == NULL) do_somthing;
* if(FLAC__stream_decoder_init_FILE(
* decoder,
* file,
* my_write_callback,
* my_metadata_callback, // or NULL
* my_error_callback,
* my_client_data
* ) != FLAC__STREAM_DECODER_INIT_STATUS_OK) do_something;
* \endcode
*
* or just:
*
* \code
* [...]
* if(FLAC__stream_decoder_init_file(
* decoder,
* "somefile.flac",
* my_write_callback,
* my_metadata_callback, // or NULL
* my_error_callback,
* my_client_data
* ) != FLAC__STREAM_DECODER_INIT_STATUS_OK) do_something;
* \endcode
*
* Another small change to the decoder is in how it handles unparseable
* streams. Before, when the decoder found an unparseable stream
* (reserved for when the decoder encounters a stream from a future
* encoder that it can't parse), it changed the state to
* \c FLAC__STREAM_DECODER_UNPARSEABLE_STREAM. Now the decoder instead
* drops sync and calls the error callback with a new error code
* \c FLAC__STREAM_DECODER_ERROR_STATUS_UNPARSEABLE_STREAM. This is
* more robust. If your error callback does not discriminate on the the
* error state, your code does not need to be changed.
*
* The encoder now has a new setting:
* FLAC__stream_encoder_set_apodization(). This is for setting the
* method used to window the data before LPC analysis. You only need to
* add a call to this function if the default is not suitable. There
* are also two new convenience functions that may be useful:
* FLAC__metadata_object_cuesheet_calculate_cddb_id() and
* FLAC__metadata_get_cuesheet().
*
* The \a bytes parameter to FLAC__StreamDecoderReadCallback,
* FLAC__StreamEncoderReadCallback, and FLAC__StreamEncoderWriteCallback
* is now \c size_t instead of \c unsigned.
*/
/** \defgroup porting_1_1_3_to_1_1_4 Porting from FLAC 1.1.3 to 1.1.4
* \ingroup porting
*
* \brief
* This module describes porting from FLAC 1.1.3 to FLAC 1.1.4.
*
* There were no changes to any of the interfaces from 1.1.3 to 1.1.4.
* There was a slight change in the implementation of
* FLAC__stream_encoder_set_metadata(); the function now makes a copy
* of the \a metadata array of pointers so the client no longer needs
* to maintain it after the call. The objects themselves that are
* pointed to by the array are still not copied though and must be
* maintained until the call to FLAC__stream_encoder_finish().
*/
/** \defgroup porting_1_1_4_to_1_2_0 Porting from FLAC 1.1.4 to 1.2.0
* \ingroup porting
*
* \brief
* This module describes porting from FLAC 1.1.4 to FLAC 1.2.0.
*
* There were only very minor changes to the interfaces from 1.1.4 to 1.2.0.
* In libFLAC, \c FLAC__format_sample_rate_is_subset() was added.
* In libFLAC++, \c FLAC::Decoder::Stream::get_decode_position() was added.
*
* Finally, value of the constant \c FLAC__FRAME_HEADER_RESERVED_LEN
* has changed to reflect the conversion of one of the reserved bits
* into active use. It used to be \c 2 and now is \c 1. However the
* FLAC frame header length has not changed, so to skip the proper
* number of bits, use \c FLAC__FRAME_HEADER_RESERVED_LEN +
* \c FLAC__FRAME_HEADER_BLOCKING_STRATEGY_LEN
*/
/** \defgroup flac FLAC C API
*
* The FLAC C API is the interface to libFLAC, a set of structures
* describing the components of FLAC streams, and functions for
* encoding and decoding streams, as well as manipulating FLAC
* metadata in files.
*
* You should start with the format components as all other modules
* are dependent on it.
*/
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/alloc.h
+209
View File
@@ -0,0 +1,209 @@
/* alloc - Convenience routines for safely allocating memory
* Copyright (C) 2007-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__SHARE__ALLOC_H
#define FLAC__SHARE__ALLOC_H
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
/* WATCHOUT: for c++ you may have to #define __STDC_LIMIT_MACROS 1 real early
* before #including this file, otherwise SIZE_MAX might not be defined
*/
#include <limits.h> /* for SIZE_MAX */
#if HAVE_STDINT_H
#include <stdint.h> /* for SIZE_MAX in case limits.h didn't get it */
#endif
#include <stdlib.h> /* for size_t, malloc(), etc */
#include "compat.h"
#ifndef SIZE_MAX
# ifndef SIZE_T_MAX
# ifdef _MSC_VER
# ifdef _WIN64
# define SIZE_T_MAX 0xffffffffffffffffui64
# else
# define SIZE_T_MAX 0xffffffff
# endif
# else
# error
# endif
# endif
# define SIZE_MAX SIZE_T_MAX
#endif
/* avoid malloc()ing 0 bytes, see:
* https://www.securecoding.cert.org/confluence/display/seccode/MEM04-A.+Do+not+make+assumptions+about+the+result+of+allocating+0+bytes?focusedCommentId=5407003
*/
static inline void *safe_malloc_(size_t size)
{
/* malloc(0) is undefined; FLAC src convention is to always allocate */
if(!size)
size++;
return malloc(size);
}
static inline void *safe_calloc_(size_t nmemb, size_t size)
{
if(!nmemb || !size)
return malloc(1); /* malloc(0) is undefined; FLAC src convention is to always allocate */
return calloc(nmemb, size);
}
/*@@@@ there's probably a better way to prevent overflows when allocating untrusted sums but this works for now */
static inline void *safe_malloc_add_2op_(size_t size1, size_t size2)
{
size2 += size1;
if(size2 < size1)
return 0;
return safe_malloc_(size2);
}
static inline void *safe_malloc_add_3op_(size_t size1, size_t size2, size_t size3)
{
size2 += size1;
if(size2 < size1)
return 0;
size3 += size2;
if(size3 < size2)
return 0;
return safe_malloc_(size3);
}
static inline void *safe_malloc_add_4op_(size_t size1, size_t size2, size_t size3, size_t size4)
{
size2 += size1;
if(size2 < size1)
return 0;
size3 += size2;
if(size3 < size2)
return 0;
size4 += size3;
if(size4 < size3)
return 0;
return safe_malloc_(size4);
}
void *safe_malloc_mul_2op_(size_t size1, size_t size2) ;
static inline void *safe_malloc_mul_3op_(size_t size1, size_t size2, size_t size3)
{
if(!size1 || !size2 || !size3)
return malloc(1); /* malloc(0) is undefined; FLAC src convention is to always allocate */
if(size1 > SIZE_MAX / size2)
return 0;
size1 *= size2;
if(size1 > SIZE_MAX / size3)
return 0;
return malloc(size1*size3);
}
/* size1*size2 + size3 */
static inline void *safe_malloc_mul2add_(size_t size1, size_t size2, size_t size3)
{
if(!size1 || !size2)
return safe_malloc_(size3);
if(size1 > SIZE_MAX / size2)
return 0;
return safe_malloc_add_2op_(size1*size2, size3);
}
/* size1 * (size2 + size3) */
static inline void *safe_malloc_muladd2_(size_t size1, size_t size2, size_t size3)
{
if(!size1 || (!size2 && !size3))
return malloc(1); /* malloc(0) is undefined; FLAC src convention is to always allocate */
size2 += size3;
if(size2 < size3)
return 0;
if(size1 > SIZE_MAX / size2)
return 0;
return malloc(size1*size2);
}
static inline void *safe_realloc_add_2op_(void *ptr, size_t size1, size_t size2)
{
size2 += size1;
if(size2 < size1)
return 0;
return realloc(ptr, size2);
}
static inline void *safe_realloc_add_3op_(void *ptr, size_t size1, size_t size2, size_t size3)
{
size2 += size1;
if(size2 < size1)
return 0;
size3 += size2;
if(size3 < size2)
return 0;
return realloc(ptr, size3);
}
static inline void *safe_realloc_add_4op_(void *ptr, size_t size1, size_t size2, size_t size3, size_t size4)
{
size2 += size1;
if(size2 < size1)
return 0;
size3 += size2;
if(size3 < size2)
return 0;
size4 += size3;
if(size4 < size3)
return 0;
return realloc(ptr, size4);
}
static inline void *safe_realloc_mul_2op_(void *ptr, size_t size1, size_t size2)
{
if(!size1 || !size2)
return realloc(ptr, 0); /* preserve POSIX realloc(ptr, 0) semantics */
if(size1 > SIZE_MAX / size2)
return 0;
return realloc(ptr, size1*size2);
}
/* size1 * (size2 + size3) */
static inline void *safe_realloc_muladd2_(void *ptr, size_t size1, size_t size2, size_t size3)
{
if(!size1 || (!size2 && !size3))
return realloc(ptr, 0); /* preserve POSIX realloc(ptr, 0) semantics */
size2 += size3;
if(size2 < size3)
return 0;
return safe_realloc_mul_2op_(ptr, size1, size2);
}
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/assert.h
+46
View File
@@ -0,0 +1,46 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2001-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__ASSERT_H
#define FLAC__ASSERT_H
/* we need this since some compilers (like MSVC) leave assert()s on release code (and we don't want to use their ASSERT) */
#ifdef DEBUG
#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
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/callback.h
+185
View File
@@ -0,0 +1,185 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2004-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__CALLBACK_H
#define FLAC__CALLBACK_H
#include "ordinals.h"
#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
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/compat.h
+200
View File
@@ -0,0 +1,200 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2012-2014 Xiph.org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/* This is the prefered location of all CPP hackery to make $random_compiler
* work like something approaching a C99 (or maybe more accurately GNU99)
* compiler.
*
* It is assumed that this header will be included after "config.h".
*/
#ifndef FLAC__SHARE__COMPAT_H
#define FLAC__SHARE__COMPAT_H
#if defined _WIN32 && !defined __CYGWIN__
/* where MSVC puts unlink() */
# include <io.h>
#else
# include <unistd.h>
#endif
#if defined _MSC_VER || defined __BORLANDC__ || defined __MINGW32__
#include <sys/types.h> /* for off_t */
#define FLAC__off_t __int64 /* use this instead of off_t to fix the 2 GB limit */
#if !defined __MINGW32__
#define fseeko _fseeki64
#define ftello _ftelli64
#else /* MinGW */
#if !defined(HAVE_FSEEKO)
#define fseeko fseeko64
#define ftello ftello64
#endif
#endif
#else
#define FLAC__off_t off_t
#endif
#if HAVE_INTTYPES_H
#define __STDC_FORMAT_MACROS
#include <inttypes.h>
#endif
#if defined(_MSC_VER)
#define strtoll _strtoi64
#define strtoull _strtoui64
#endif
#if defined(_MSC_VER)
#define inline __inline
#endif
#if defined __INTEL_COMPILER || (defined _MSC_VER && defined _WIN64)
/* MSVS generates VERY slow 32-bit code with __restrict */
#define flac_restrict __restrict
#elif defined __GNUC__
#define flac_restrict __restrict__
#else
#define flac_restrict
#endif
#define FLAC__U64L(x) x##ULL
#if defined _MSC_VER || defined __BORLANDC__ || defined __MINGW32__
#define FLAC__STRCASECMP stricmp
#define FLAC__STRNCASECMP strnicmp
#else
#define FLAC__STRCASECMP strcasecmp
#define FLAC__STRNCASECMP strncasecmp
#endif
#if defined _MSC_VER || defined __MINGW32__ || defined __CYGWIN__ || defined __EMX__
#include <io.h> /* for _setmode(), chmod() */
#include <fcntl.h> /* for _O_BINARY */
#else
#include <unistd.h> /* for chown(), unlink() */
#endif
#if defined _MSC_VER || defined __BORLANDC__ || defined __MINGW32__
#if defined __BORLANDC__
#include <utime.h> /* for utime() */
#else
#include <sys/utime.h> /* for utime() */
#endif
#else
#include <sys/types.h> /* some flavors of BSD (like OS X) require this to get time_t */
#include <utime.h> /* for utime() */
#endif
#if defined _MSC_VER
# if _MSC_VER >= 1600
/* Visual Studio 2010 has decent C99 support */
# include <stdint.h>
# define PRIu64 "llu"
# define PRId64 "lld"
# define PRIx64 "llx"
# else
# include <limits.h>
# ifndef UINT32_MAX
# define UINT32_MAX _UI32_MAX
# endif
typedef unsigned __int64 uint64_t;
typedef unsigned __int32 uint32_t;
typedef unsigned __int16 uint16_t;
typedef unsigned __int8 uint8_t;
typedef __int64 int64_t;
typedef __int32 int32_t;
typedef __int16 int16_t;
typedef __int8 int8_t;
# define PRIu64 "I64u"
# define PRId64 "I64d"
# define PRIx64 "I64x"
# endif
#endif /* defined _MSC_VER */
#ifdef _WIN32
/* All char* strings are in UTF-8 format. Added to support Unicode files on Windows */
#include "win_utf8_io.h"
#define flac_printf printf_utf8
#define flac_fprintf fprintf_utf8
#define flac_vfprintf vfprintf_utf8
#define flac_fopen fopen_utf8
#define flac_chmod chmod_utf8
#define flac_utime utime_utf8
#define flac_unlink unlink_utf8
#define flac_rename rename_utf8
#define flac_stat _stat64_utf8
#else
#define flac_printf printf
#define flac_fprintf fprintf
#define flac_vfprintf vfprintf
#define flac_fopen fopen
#define flac_chmod chmod
#define flac_utime utime
#define flac_unlink unlink
#define flac_rename rename
#define flac_stat stat
#endif
#ifdef _WIN32
#define flac_stat_s __stat64 /* stat struct */
#define flac_fstat _fstat64
#else
#define flac_stat_s stat /* stat struct */
#define flac_fstat fstat
#endif
#ifndef M_LN2
#define M_LN2 0.69314718055994530942
#endif
#ifndef M_PI
#define M_PI 3.14159265358979323846
#endif
/* FLAC needs to compile and work correctly on systems with a normal ISO C99
* snprintf as well as Microsoft Visual Studio which has an non-standards
* conformant snprint_s function.
*
* This function wraps the MS version to behave more like the the ISO version.
*/
#ifdef __cplusplus
extern "C" {
#endif
int flac_snprintf(char *str, size_t size, const char *fmt, ...);
int flac_vsnprintf(char *str, size_t size, const char *fmt, va_list va);
#ifdef __cplusplus
};
#endif
#endif /* FLAC__SHARE__COMPAT_H */
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/endswap.h
+78
View File
@@ -0,0 +1,78 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2012-2014 Xiph.org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
/* It is assumed that this header will be included after "config.h". */
#if HAVE_BSWAP32 /* GCC and Clang */
/* GCC prior to 4.8 didn't provide bswap16 on x86_64 */
#if ! HAVE_BSWAP16
static inline unsigned short __builtin_bswap16(unsigned short a)
{
return (a<<8)|(a>>8);
}
#endif
#define ENDSWAP_16(x) (__builtin_bswap16 (x))
#define ENDSWAP_32(x) (__builtin_bswap32 (x))
#elif defined _MSC_VER /* Windows. Apparently in <stdlib.h>. */
#define ENDSWAP_16(x) (_byteswap_ushort (x))
#define ENDSWAP_32(x) (_byteswap_ulong (x))
#elif defined HAVE_BYTESWAP_H /* Linux */
#include <byteswap.h>
#define ENDSWAP_16(x) (bswap_16 (x))
#define ENDSWAP_32(x) (bswap_32 (x))
#else
#define ENDSWAP_16(x) ((((x) >> 8) & 0xFF) | (((x) & 0xFF) << 8))
#define ENDSWAP_32(x) ((((x) >> 24) & 0xFF) | (((x) >> 8) & 0xFF00) | (((x) & 0xFF00) << 8) | (((x) & 0xFF) << 24))
#endif
/* Host to little-endian byte swapping. */
#if CPU_IS_BIG_ENDIAN
#define H2LE_16(x) ENDSWAP_16 (x)
#define H2LE_32(x) ENDSWAP_32 (x)
#else
#define H2LE_16(x) (x)
#define H2LE_32(x) (x)
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/export.h
+97
View File
@@ -0,0 +1,97 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__EXPORT_H
#define FLAC__EXPORT_H
/** \file include/FLAC/export.h
*
* \brief
* This module contains #defines and symbols for exporting function
* calls, and providing version information and compiled-in features.
*
* See the \link flac_export export \endlink module.
*/
/** \defgroup flac_export FLAC/export.h: export symbols
* \ingroup flac
*
* \brief
* This module contains #defines and symbols for exporting function
* calls, and providing version information and compiled-in features.
*
* If you are compiling with MSVC and will link to the static library
* (libFLAC.lib) you should define FLAC__NO_DLL in your project to
* make sure the symbols are exported properly.
*
* \{
*/
#if defined(FLAC__NO_DLL)
#define FLAC_API
#elif defined(_MSC_VER)
#ifdef FLAC_API_EXPORTS
#define FLAC_API __declspec(dllexport)
#else
#define FLAC_API __declspec(dllimport)
#endif
#elif defined(FLAC__USE_VISIBILITY_ATTR)
#define FLAC_API __attribute__ ((visibility ("default")))
#else
#define FLAC_API
#endif
/** These #defines will mirror the libtool-based library version number, see
* http://www.gnu.org/software/libtool/manual/libtool.html#Libtool-versioning
*/
#define FLAC_API_VERSION_CURRENT 11
#define FLAC_API_VERSION_REVISION 0 /**< see above */
#define FLAC_API_VERSION_AGE 3 /**< see above */
#ifdef __cplusplus
extern "C" {
#endif
/** \c 1 if the library has been compiled with support for Ogg FLAC, else \c 0. */
extern FLAC_API int FLAC_API_SUPPORTS_OGG_FLAC;
#ifdef __cplusplus
}
#endif
/* \} */
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/format.h
+1025
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/bitmath.c
+109
View File
@@ -0,0 +1,109 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2001-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include "include/private/bitmath.h"
/* An example of what FLAC__bitmath_silog2() computes:
*
* silog2(-10) = 5
* silog2(- 9) = 5
* silog2(- 8) = 4
* silog2(- 7) = 4
* silog2(- 6) = 4
* silog2(- 5) = 4
* silog2(- 4) = 3
* silog2(- 3) = 3
* silog2(- 2) = 2
* silog2(- 1) = 2
* silog2( 0) = 0
* silog2( 1) = 2
* silog2( 2) = 3
* silog2( 3) = 3
* silog2( 4) = 4
* silog2( 5) = 4
* silog2( 6) = 4
* silog2( 7) = 4
* silog2( 8) = 5
* silog2( 9) = 5
* silog2( 10) = 5
*/
unsigned FLAC__bitmath_silog2(int v)
{
while(1) {
if(v == 0) {
return 0;
}
else if(v > 0) {
unsigned l = 0;
while(v) {
l++;
v >>= 1;
}
return l+1;
}
else if(v == -1) {
return 2;
}
else {
v++;
v = -v;
}
}
}
unsigned FLAC__bitmath_silog2_wide(FLAC__int64 v)
{
while(1) {
if(v == 0) {
return 0;
}
else if(v > 0) {
unsigned l = 0;
while(v) {
l++;
v >>= 1;
}
return l+1;
}
else if(v == -1) {
return 2;
}
else {
v++;
v = -v;
}
}
}
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/bitreader.c
+1058
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/bitwriter.c
+842
View File
@@ -0,0 +1,842 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <stdlib.h>
#include <string.h>
#include "include/private/bitwriter.h"
#include "include/private/crc.h"
#include "../assert.h"
#include "../alloc.h"
#include "../compat.h"
#include "../endswap.h"
/* Things should be fastest when this matches the machine word size */
/* WATCHOUT: if you change this you must also change the following #defines down to SWAP_BE_WORD_TO_HOST below to match */
/* WATCHOUT: there are a few places where the code will not work unless uint32_t is >= 32 bits wide */
#define FLAC__BYTES_PER_WORD 4
#define FLAC__BITS_PER_WORD (8 * FLAC__BYTES_PER_WORD)
#define FLAC__WORD_ALL_ONES ((FLAC__uint32)0xffffffff)
/* SWAP_BE_WORD_TO_HOST swaps bytes in a uint32_t (which is always big-endian) if necessary to match host byte order */
#if WORDS_BIGENDIAN
#define SWAP_BE_WORD_TO_HOST(x) (x)
#else
#define SWAP_BE_WORD_TO_HOST(x) ENDSWAP_32(x)
#endif
/*
* The default capacity here doesn't matter too much. The buffer always grows
* to hold whatever is written to it. Usually the encoder will stop adding at
* a frame or metadata block, then write that out and clear the buffer for the
* next one.
*/
static const unsigned FLAC__BITWRITER_DEFAULT_CAPACITY = 32768u / sizeof(uint32_t); /* size in words */
/* When growing, increment 4K at a time */
static const unsigned FLAC__BITWRITER_DEFAULT_INCREMENT = 4096u / sizeof(uint32_t); /* size in words */
#define FLAC__WORDS_TO_BITS(words) ((words) * FLAC__BITS_PER_WORD)
#define FLAC__TOTAL_BITS(bw) (FLAC__WORDS_TO_BITS((bw)->words) + (bw)->bits)
struct FLAC__BitWriter {
uint32_t *buffer;
uint32_t accum; /* accumulator; bits are right-justified; when full, accum is appended to buffer */
unsigned capacity; /* capacity of buffer in words */
unsigned words; /* # of complete words in buffer */
unsigned bits; /* # of used bits in accum */
};
/* * WATCHOUT: The current implementation only grows the buffer. */
#ifndef __SUNPRO_C
static
#endif
FLAC__bool bitwriter_grow_(FLAC__BitWriter *bw, unsigned bits_to_add)
{
unsigned new_capacity;
uint32_t *new_buffer;
FLAC__ASSERT(0 != bw);
FLAC__ASSERT(0 != bw->buffer);
/* calculate total words needed to store 'bits_to_add' additional bits */
new_capacity = bw->words + ((bw->bits + bits_to_add + FLAC__BITS_PER_WORD - 1) / FLAC__BITS_PER_WORD);
/* it's possible (due to pessimism in the growth estimation that
* leads to this call) that we don't actually need to grow
*/
if(bw->capacity >= new_capacity)
return true;
/* round up capacity increase to the nearest FLAC__BITWRITER_DEFAULT_INCREMENT */
if((new_capacity - bw->capacity) % FLAC__BITWRITER_DEFAULT_INCREMENT)
new_capacity += FLAC__BITWRITER_DEFAULT_INCREMENT - ((new_capacity - bw->capacity) % FLAC__BITWRITER_DEFAULT_INCREMENT);
/* make sure we got everything right */
FLAC__ASSERT(0 == (new_capacity - bw->capacity) % FLAC__BITWRITER_DEFAULT_INCREMENT);
FLAC__ASSERT(new_capacity > bw->capacity);
FLAC__ASSERT(new_capacity >= bw->words + ((bw->bits + bits_to_add + FLAC__BITS_PER_WORD - 1) / FLAC__BITS_PER_WORD));
new_buffer = (uint32_t*) safe_realloc_mul_2op_(bw->buffer, sizeof(uint32_t), /*times*/new_capacity);
if(new_buffer == 0)
return false;
bw->buffer = new_buffer;
bw->capacity = new_capacity;
return true;
}
/***********************************************************************
*
* Class constructor/destructor
*
***********************************************************************/
FLAC__BitWriter *FLAC__bitwriter_new(void)
{
FLAC__BitWriter *bw = (FLAC__BitWriter*) calloc(1, sizeof(FLAC__BitWriter));
/* note that calloc() sets all members to 0 for us */
return bw;
}
void FLAC__bitwriter_delete(FLAC__BitWriter *bw)
{
FLAC__ASSERT(0 != bw);
FLAC__bitwriter_free(bw);
free(bw);
}
/***********************************************************************
*
* Public class methods
*
***********************************************************************/
FLAC__bool FLAC__bitwriter_init(FLAC__BitWriter *bw)
{
FLAC__ASSERT(0 != bw);
bw->words = bw->bits = 0;
bw->capacity = FLAC__BITWRITER_DEFAULT_CAPACITY;
bw->buffer = (uint32_t*) malloc(sizeof(uint32_t) * bw->capacity);
if(bw->buffer == 0)
return false;
return true;
}
void FLAC__bitwriter_free(FLAC__BitWriter *bw)
{
FLAC__ASSERT(0 != bw);
if(0 != bw->buffer)
free(bw->buffer);
bw->buffer = 0;
bw->capacity = 0;
bw->words = bw->bits = 0;
}
void FLAC__bitwriter_clear(FLAC__BitWriter *bw)
{
bw->words = bw->bits = 0;
}
void FLAC__bitwriter_dump(const FLAC__BitWriter *bw, FILE *out)
{
unsigned i, j;
if(bw == 0) {
fprintf(out, "bitwriter is NULL\n");
}
else {
fprintf(out, "bitwriter: capacity=%u words=%u bits=%u total_bits=%u\n", bw->capacity, bw->words, bw->bits, FLAC__TOTAL_BITS(bw));
for(i = 0; i < bw->words; i++) {
fprintf(out, "%08X: ", i);
for(j = 0; j < FLAC__BITS_PER_WORD; j++)
fprintf(out, "%01u", bw->buffer[i] & (1 << (FLAC__BITS_PER_WORD-j-1)) ? 1:0);
fprintf(out, "\n");
}
if(bw->bits > 0) {
fprintf(out, "%08X: ", i);
for(j = 0; j < bw->bits; j++)
fprintf(out, "%01u", bw->accum & (1 << (bw->bits-j-1)) ? 1:0);
fprintf(out, "\n");
}
}
}
FLAC__bool FLAC__bitwriter_get_write_crc16(FLAC__BitWriter *bw, FLAC__uint16 *crc)
{
const FLAC__byte *buffer;
size_t bytes;
FLAC__ASSERT((bw->bits & 7) == 0); /* assert that we're byte-aligned */
if(!FLAC__bitwriter_get_buffer(bw, &buffer, &bytes))
return false;
*crc = (FLAC__uint16)FLAC__crc16(buffer, bytes);
FLAC__bitwriter_release_buffer(bw);
return true;
}
FLAC__bool FLAC__bitwriter_get_write_crc8(FLAC__BitWriter *bw, FLAC__byte *crc)
{
const FLAC__byte *buffer;
size_t bytes;
FLAC__ASSERT((bw->bits & 7) == 0); /* assert that we're byte-aligned */
if(!FLAC__bitwriter_get_buffer(bw, &buffer, &bytes))
return false;
*crc = FLAC__crc8(buffer, bytes);
FLAC__bitwriter_release_buffer(bw);
return true;
}
FLAC__bool FLAC__bitwriter_is_byte_aligned(const FLAC__BitWriter *bw)
{
return ((bw->bits & 7) == 0);
}
unsigned FLAC__bitwriter_get_input_bits_unconsumed(const FLAC__BitWriter *bw)
{
return FLAC__TOTAL_BITS(bw);
}
FLAC__bool FLAC__bitwriter_get_buffer(FLAC__BitWriter *bw, const FLAC__byte **buffer, size_t *bytes)
{
FLAC__ASSERT((bw->bits & 7) == 0);
/* double protection */
if(bw->bits & 7)
return false;
/* if we have bits in the accumulator we have to flush those to the buffer first */
if(bw->bits) {
FLAC__ASSERT(bw->words <= bw->capacity);
if(bw->words == bw->capacity && !bitwriter_grow_(bw, FLAC__BITS_PER_WORD))
return false;
/* append bits as complete word to buffer, but don't change bw->accum or bw->bits */
bw->buffer[bw->words] = SWAP_BE_WORD_TO_HOST(bw->accum << (FLAC__BITS_PER_WORD-bw->bits));
}
/* now we can just return what we have */
*buffer = (FLAC__byte*)bw->buffer;
*bytes = (FLAC__BYTES_PER_WORD * bw->words) + (bw->bits >> 3);
return true;
}
void FLAC__bitwriter_release_buffer(FLAC__BitWriter *bw)
{
/* nothing to do. in the future, strict checking of a 'writer-is-in-
* get-mode' flag could be added everywhere and then cleared here
*/
(void)bw;
}
inline FLAC__bool FLAC__bitwriter_write_zeroes(FLAC__BitWriter *bw, unsigned bits)
{
unsigned n;
FLAC__ASSERT(0 != bw);
FLAC__ASSERT(0 != bw->buffer);
if(bits == 0)
return true;
/* slightly pessimistic size check but faster than "<= bw->words + (bw->bits+bits+FLAC__BITS_PER_WORD-1)/FLAC__BITS_PER_WORD" */
if(bw->capacity <= bw->words + bits && !bitwriter_grow_(bw, bits))
return false;
/* first part gets to word alignment */
if(bw->bits) {
n = flac_min(FLAC__BITS_PER_WORD - bw->bits, bits);
bw->accum <<= n;
bits -= n;
bw->bits += n;
if(bw->bits == FLAC__BITS_PER_WORD) {
bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
bw->bits = 0;
}
else
return true;
}
/* do whole words */
while(bits >= FLAC__BITS_PER_WORD) {
bw->buffer[bw->words++] = 0;
bits -= FLAC__BITS_PER_WORD;
}
/* do any leftovers */
if(bits > 0) {
bw->accum = 0;
bw->bits = bits;
}
return true;
}
inline FLAC__bool FLAC__bitwriter_write_raw_uint32(FLAC__BitWriter *bw, FLAC__uint32 val, unsigned bits)
{
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;
}
inline FLAC__bool FLAC__bitwriter_write_raw_int32(FLAC__BitWriter *bw, FLAC__int32 val, unsigned bits)
{
/* zero-out unused bits */
if(bits < 32)
val &= (~(0xffffffff << bits));
return FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)val, bits);
}
inline FLAC__bool FLAC__bitwriter_write_raw_uint64(FLAC__BitWriter *bw, FLAC__uint64 val, unsigned bits)
{
/* this could be a little faster but it's not used for much */
if(bits > 32) {
return
FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)(val>>32), bits-32) &&
FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)val, 32);
}
else
return FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)val, bits);
}
inline FLAC__bool FLAC__bitwriter_write_raw_uint32_little_endian(FLAC__BitWriter *bw, FLAC__uint32 val)
{
/* this doesn't need to be that fast as currently it is only used for vorbis comments */
if(!FLAC__bitwriter_write_raw_uint32(bw, val & 0xff, 8))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, (val>>8) & 0xff, 8))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, (val>>16) & 0xff, 8))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, val>>24, 8))
return false;
return true;
}
inline FLAC__bool FLAC__bitwriter_write_byte_block(FLAC__BitWriter *bw, const FLAC__byte vals[], unsigned nvals)
{
unsigned i;
/* this could be faster but currently we don't need it to be since it's only used for writing metadata */
for(i = 0; i < nvals; i++) {
if(!FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)(vals[i]), 8))
return false;
}
return true;
}
FLAC__bool FLAC__bitwriter_write_unary_unsigned(FLAC__BitWriter *bw, unsigned val)
{
if(val < 32)
return FLAC__bitwriter_write_raw_uint32(bw, 1, ++val);
else
return
FLAC__bitwriter_write_zeroes(bw, val) &&
FLAC__bitwriter_write_raw_uint32(bw, 1, 1);
}
unsigned FLAC__bitwriter_rice_bits(FLAC__int32 val, unsigned parameter)
{
FLAC__uint32 uval;
FLAC__ASSERT(parameter < sizeof(unsigned)*8);
/* fold signed to unsigned; actual formula is: negative(v)? -2v-1 : 2v */
uval = (val<<1) ^ (val>>31);
return 1 + parameter + (uval >> parameter);
}
#if 0 /* UNUSED */
unsigned FLAC__bitwriter_golomb_bits_signed(int val, unsigned parameter)
{
unsigned bits, msbs, uval;
unsigned k;
FLAC__ASSERT(parameter > 0);
/* fold signed to unsigned */
if(val < 0)
uval = (unsigned)(((-(++val)) << 1) + 1);
else
uval = (unsigned)(val << 1);
k = FLAC__bitmath_ilog2(parameter);
if(parameter == 1u<<k) {
FLAC__ASSERT(k <= 30);
msbs = uval >> k;
bits = 1 + k + msbs;
}
else {
unsigned q, r, d;
d = (1 << (k+1)) - parameter;
q = uval / parameter;
r = uval - (q * parameter);
bits = 1 + q + k;
if(r >= d)
bits++;
}
return bits;
}
unsigned FLAC__bitwriter_golomb_bits_unsigned(unsigned uval, unsigned parameter)
{
unsigned bits, msbs;
unsigned k;
FLAC__ASSERT(parameter > 0);
k = FLAC__bitmath_ilog2(parameter);
if(parameter == 1u<<k) {
FLAC__ASSERT(k <= 30);
msbs = uval >> k;
bits = 1 + k + msbs;
}
else {
unsigned q, r, d;
d = (1 << (k+1)) - parameter;
q = uval / parameter;
r = uval - (q * parameter);
bits = 1 + q + k;
if(r >= d)
bits++;
}
return bits;
}
#endif /* UNUSED */
FLAC__bool FLAC__bitwriter_write_rice_signed(FLAC__BitWriter *bw, FLAC__int32 val, unsigned parameter)
{
unsigned total_bits, interesting_bits, msbs;
FLAC__uint32 uval, pattern;
FLAC__ASSERT(0 != bw);
FLAC__ASSERT(0 != bw->buffer);
FLAC__ASSERT(parameter < 8*sizeof(uval));
/* fold signed to unsigned; actual formula is: negative(v)? -2v-1 : 2v */
uval = (val<<1) ^ (val>>31);
msbs = uval >> parameter;
interesting_bits = 1 + parameter;
total_bits = interesting_bits + msbs;
pattern = 1 << parameter; /* the unary end bit */
pattern |= (uval & ((1<<parameter)-1)); /* the binary LSBs */
if(total_bits <= 32)
return FLAC__bitwriter_write_raw_uint32(bw, pattern, total_bits);
else
return
FLAC__bitwriter_write_zeroes(bw, msbs) && /* write the unary MSBs */
FLAC__bitwriter_write_raw_uint32(bw, pattern, interesting_bits); /* write the unary end bit and binary LSBs */
}
FLAC__bool FLAC__bitwriter_write_rice_signed_block(FLAC__BitWriter *bw, const FLAC__int32 *vals, unsigned nvals, unsigned parameter)
{
const FLAC__uint32 mask1 = FLAC__WORD_ALL_ONES << parameter; /* we val|=mask1 to set the stop bit above it... */
const FLAC__uint32 mask2 = FLAC__WORD_ALL_ONES >> (31-parameter); /* ...then mask off the bits above the stop bit with val&=mask2*/
FLAC__uint32 uval;
unsigned left;
const unsigned lsbits = 1 + parameter;
unsigned msbits;
FLAC__ASSERT(0 != bw);
FLAC__ASSERT(0 != bw->buffer);
FLAC__ASSERT(parameter < 8*sizeof(uint32_t)-1);
/* WATCHOUT: code does not work with <32bit words; we can make things much faster with this assertion */
FLAC__ASSERT(FLAC__BITS_PER_WORD >= 32);
while(nvals) {
/* fold signed to unsigned; actual formula is: negative(v)? -2v-1 : 2v */
uval = (*vals<<1) ^ (*vals>>31);
msbits = uval >> parameter;
if(bw->bits && bw->bits + msbits + lsbits < FLAC__BITS_PER_WORD) { /* i.e. if the whole thing fits in the current uint32_t */
/* ^^^ if bw->bits is 0 then we may have filled the buffer and have no free uint32_t to work in */
bw->bits = bw->bits + msbits + lsbits;
uval |= mask1; /* set stop bit */
uval &= mask2; /* mask off unused top bits */
bw->accum <<= msbits + lsbits;
bw->accum |= uval;
}
else {
/* slightly pessimistic size check but faster than "<= bw->words + (bw->bits+msbits+lsbits+FLAC__BITS_PER_WORD-1)/FLAC__BITS_PER_WORD" */
/* OPT: pessimism may cause flurry of false calls to grow_ which eat up all savings before it */
if(bw->capacity <= bw->words + bw->bits + msbits + 1/*lsbits always fit in 1 uint32_t*/ && !bitwriter_grow_(bw, msbits+lsbits))
return false;
if(msbits) {
/* first part gets to word alignment */
if(bw->bits) {
left = FLAC__BITS_PER_WORD - bw->bits;
if(msbits < left) {
bw->accum <<= msbits;
bw->bits += msbits;
goto break1;
}
else {
bw->accum <<= left;
msbits -= left;
bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
bw->bits = 0;
}
}
/* do whole words */
while(msbits >= FLAC__BITS_PER_WORD) {
bw->buffer[bw->words++] = 0;
msbits -= FLAC__BITS_PER_WORD;
}
/* do any leftovers */
if(msbits > 0) {
bw->accum = 0;
bw->bits = msbits;
}
}
break1:
uval |= mask1; /* set stop bit */
uval &= mask2; /* mask off unused top bits */
left = FLAC__BITS_PER_WORD - bw->bits;
if(lsbits < left) {
bw->accum <<= lsbits;
bw->accum |= uval;
bw->bits += lsbits;
}
else {
/* if bw->bits == 0, left==FLAC__BITS_PER_WORD which will always
* be > lsbits (because of previous assertions) so it would have
* triggered the (lsbits<left) case above.
*/
FLAC__ASSERT(bw->bits);
FLAC__ASSERT(left < FLAC__BITS_PER_WORD);
bw->accum <<= left;
bw->accum |= uval >> (bw->bits = lsbits - left);
bw->buffer[bw->words++] = SWAP_BE_WORD_TO_HOST(bw->accum);
bw->accum = uval;
}
}
vals++;
nvals--;
}
return true;
}
#if 0 /* UNUSED */
FLAC__bool FLAC__bitwriter_write_golomb_signed(FLAC__BitWriter *bw, int val, unsigned parameter)
{
unsigned total_bits, msbs, uval;
unsigned k;
FLAC__ASSERT(0 != bw);
FLAC__ASSERT(0 != bw->buffer);
FLAC__ASSERT(parameter > 0);
/* fold signed to unsigned */
if(val < 0)
uval = (unsigned)(((-(++val)) << 1) + 1);
else
uval = (unsigned)(val << 1);
k = FLAC__bitmath_ilog2(parameter);
if(parameter == 1u<<k) {
unsigned pattern;
FLAC__ASSERT(k <= 30);
msbs = uval >> k;
total_bits = 1 + k + msbs;
pattern = 1 << k; /* the unary end bit */
pattern |= (uval & ((1u<<k)-1)); /* the binary LSBs */
if(total_bits <= 32) {
if(!FLAC__bitwriter_write_raw_uint32(bw, pattern, total_bits))
return false;
}
else {
/* write the unary MSBs */
if(!FLAC__bitwriter_write_zeroes(bw, msbs))
return false;
/* write the unary end bit and binary LSBs */
if(!FLAC__bitwriter_write_raw_uint32(bw, pattern, k+1))
return false;
}
}
else {
unsigned q, r, d;
d = (1 << (k+1)) - parameter;
q = uval / parameter;
r = uval - (q * parameter);
/* write the unary MSBs */
if(!FLAC__bitwriter_write_zeroes(bw, q))
return false;
/* write the unary end bit */
if(!FLAC__bitwriter_write_raw_uint32(bw, 1, 1))
return false;
/* write the binary LSBs */
if(r >= d) {
if(!FLAC__bitwriter_write_raw_uint32(bw, r+d, k+1))
return false;
}
else {
if(!FLAC__bitwriter_write_raw_uint32(bw, r, k))
return false;
}
}
return true;
}
FLAC__bool FLAC__bitwriter_write_golomb_unsigned(FLAC__BitWriter *bw, unsigned uval, unsigned parameter)
{
unsigned total_bits, msbs;
unsigned k;
FLAC__ASSERT(0 != bw);
FLAC__ASSERT(0 != bw->buffer);
FLAC__ASSERT(parameter > 0);
k = FLAC__bitmath_ilog2(parameter);
if(parameter == 1u<<k) {
unsigned pattern;
FLAC__ASSERT(k <= 30);
msbs = uval >> k;
total_bits = 1 + k + msbs;
pattern = 1 << k; /* the unary end bit */
pattern |= (uval & ((1u<<k)-1)); /* the binary LSBs */
if(total_bits <= 32) {
if(!FLAC__bitwriter_write_raw_uint32(bw, pattern, total_bits))
return false;
}
else {
/* write the unary MSBs */
if(!FLAC__bitwriter_write_zeroes(bw, msbs))
return false;
/* write the unary end bit and binary LSBs */
if(!FLAC__bitwriter_write_raw_uint32(bw, pattern, k+1))
return false;
}
}
else {
unsigned q, r, d;
d = (1 << (k+1)) - parameter;
q = uval / parameter;
r = uval - (q * parameter);
/* write the unary MSBs */
if(!FLAC__bitwriter_write_zeroes(bw, q))
return false;
/* write the unary end bit */
if(!FLAC__bitwriter_write_raw_uint32(bw, 1, 1))
return false;
/* write the binary LSBs */
if(r >= d) {
if(!FLAC__bitwriter_write_raw_uint32(bw, r+d, k+1))
return false;
}
else {
if(!FLAC__bitwriter_write_raw_uint32(bw, r, k))
return false;
}
}
return true;
}
#endif /* UNUSED */
FLAC__bool FLAC__bitwriter_write_utf8_uint32(FLAC__BitWriter *bw, FLAC__uint32 val)
{
FLAC__bool ok = 1;
FLAC__ASSERT(0 != bw);
FLAC__ASSERT(0 != bw->buffer);
FLAC__ASSERT(!(val & 0x80000000)); /* this version only handles 31 bits */
if(val < 0x80) {
return FLAC__bitwriter_write_raw_uint32(bw, val, 8);
}
else if(val < 0x800) {
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xC0 | (val>>6), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
}
else if(val < 0x10000) {
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xE0 | (val>>12), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>6)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
}
else if(val < 0x200000) {
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xF0 | (val>>18), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>12)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>6)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
}
else if(val < 0x4000000) {
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xF8 | (val>>24), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>18)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>12)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>6)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
}
else {
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xFC | (val>>30), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>24)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>18)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>12)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | ((val>>6)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (val&0x3F), 8);
}
return ok;
}
FLAC__bool FLAC__bitwriter_write_utf8_uint64(FLAC__BitWriter *bw, FLAC__uint64 val)
{
FLAC__bool ok = 1;
FLAC__ASSERT(0 != bw);
FLAC__ASSERT(0 != bw->buffer);
FLAC__ASSERT(!(val & FLAC__U64L(0xFFFFFFF000000000))); /* this version only handles 36 bits */
if(val < 0x80) {
return FLAC__bitwriter_write_raw_uint32(bw, (FLAC__uint32)val, 8);
}
else if(val < 0x800) {
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xC0 | (FLAC__uint32)(val>>6), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
}
else if(val < 0x10000) {
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xE0 | (FLAC__uint32)(val>>12), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
}
else if(val < 0x200000) {
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xF0 | (FLAC__uint32)(val>>18), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>12)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
}
else if(val < 0x4000000) {
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xF8 | (FLAC__uint32)(val>>24), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>18)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>12)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
}
else if(val < 0x80000000) {
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xFC | (FLAC__uint32)(val>>30), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>24)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>18)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>12)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
}
else {
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0xFE, 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>30)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>24)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>18)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>12)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)((val>>6)&0x3F), 8);
ok &= FLAC__bitwriter_write_raw_uint32(bw, 0x80 | (FLAC__uint32)(val&0x3F), 8);
}
return ok;
}
FLAC__bool FLAC__bitwriter_zero_pad_to_byte_boundary(FLAC__BitWriter *bw)
{
/* 0-pad to byte boundary */
if(bw->bits & 7u)
return FLAC__bitwriter_write_zeroes(bw, 8 - (bw->bits & 7u));
else
return true;
}
/* These functions are declared inline in this file but are also callable as
* externs from elsewhere.
* According to the C99 spec, section 6.7.4, simply providing a function
* prototype in a header file without 'inline' and making the function inline
* in this file should be sufficient.
* Unfortunately, the Microsoft VS compiler doesn't pick them up externally. To
* fix that we add extern declarations here.
*/
extern FLAC__bool FLAC__bitwriter_write_zeroes(FLAC__BitWriter *bw, unsigned bits);
extern FLAC__bool FLAC__bitwriter_write_raw_int32(FLAC__BitWriter *bw, FLAC__int32 val, unsigned bits);
extern FLAC__bool FLAC__bitwriter_write_raw_uint64(FLAC__BitWriter *bw, FLAC__uint64 val, unsigned bits);
extern FLAC__bool FLAC__bitwriter_write_raw_uint32_little_endian(FLAC__BitWriter *bw, FLAC__uint32 val);
extern FLAC__bool FLAC__bitwriter_write_byte_block(FLAC__BitWriter *bw, const FLAC__byte vals[], unsigned nvals);
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/cpu.c
+490
View File
@@ -0,0 +1,490 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2001-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include "include/private/cpu.h"
#if 0
#include <stdlib.h>
#include <memory.h>
#include <stdio.h>
#endif
#if defined FLAC__CPU_IA32
# include <signal.h>
static void disable_sse(FLAC__CPUInfo *info)
{
info->ia32.sse = false;
info->ia32.sse2 = false;
info->ia32.sse3 = false;
info->ia32.ssse3 = false;
info->ia32.sse41 = false;
info->ia32.sse42 = false;
}
static void disable_avx(FLAC__CPUInfo *info)
{
info->ia32.avx = false;
info->ia32.avx2 = false;
info->ia32.fma = false;
}
#elif defined FLAC__CPU_X86_64
static void disable_avx(FLAC__CPUInfo *info)
{
info->x86.avx = false;
info->x86.avx2 = false;
info->x86.fma = false;
}
#endif
#if defined (__NetBSD__) || defined(__OpenBSD__)
#include <sys/param.h>
#include <sys/sysctl.h>
#include <machine/cpu.h>
#endif
#if defined(__FreeBSD__) || defined(__FreeBSD_kernel__) || defined(__DragonFly__)
#include <sys/types.h>
#include <sys/sysctl.h>
#endif
#if defined(__APPLE__)
/* how to get sysctlbyname()? */
#endif
#ifdef FLAC__CPU_IA32
/* these are flags in EDX of CPUID AX=00000001 */
static const unsigned FLAC__CPUINFO_IA32_CPUID_CMOV = 0x00008000;
static const unsigned FLAC__CPUINFO_IA32_CPUID_MMX = 0x00800000;
static const unsigned FLAC__CPUINFO_IA32_CPUID_FXSR = 0x01000000;
static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE = 0x02000000;
static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE2 = 0x04000000;
#endif
/* these are flags in ECX of CPUID AX=00000001 */
static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE3 = 0x00000001;
static const unsigned FLAC__CPUINFO_IA32_CPUID_SSSE3 = 0x00000200;
static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE41 = 0x00080000;
static const unsigned FLAC__CPUINFO_IA32_CPUID_SSE42 = 0x00100000;
#if defined FLAC__AVX_SUPPORTED
/* these are flags in ECX of CPUID AX=00000001 */
static const unsigned FLAC__CPUINFO_IA32_CPUID_OSXSAVE = 0x08000000;
static const unsigned FLAC__CPUINFO_IA32_CPUID_AVX = 0x10000000;
static const unsigned FLAC__CPUINFO_IA32_CPUID_FMA = 0x00001000;
/* these are flags in EBX of CPUID AX=00000007 */
static const unsigned FLAC__CPUINFO_IA32_CPUID_AVX2 = 0x00000020;
#endif
/*
* Extra stuff needed for detection of OS support for SSE on IA-32
*/
#if defined(FLAC__CPU_IA32) && !defined FLAC__NO_ASM && (defined FLAC__HAS_NASM || defined FLAC__HAS_X86INTRIN) && !defined FLAC__NO_SSE_OS && !defined FLAC__SSE_OS
# if defined(__linux__)
/*
* If the OS doesn't support SSE, we will get here with a SIGILL. We
* modify the return address to jump over the offending SSE instruction
* and also the operation following it that indicates the instruction
* executed successfully. In this way we use no global variables and
* stay thread-safe.
*
* 3 + 3 + 6:
* 3 bytes for "xorps xmm0,xmm0"
* 3 bytes for estimate of how long the follwing "inc var" instruction is
* 6 bytes extra in case our estimate is wrong
* 12 bytes puts us in the NOP "landing zone"
*/
# include <sys/ucontext.h>
static void sigill_handler_sse_os(int signal, siginfo_t *si, void *uc)
{
(void)signal, (void)si;
((ucontext_t*)uc)->uc_mcontext.gregs[14/*REG_EIP*/] += 3 + 3 + 6;
}
# elif defined(_MSC_VER)
# include <windows.h>
# endif
#endif
void FLAC__cpu_info(FLAC__CPUInfo *info)
{
/*
* IA32-specific
*/
#ifdef FLAC__CPU_IA32
FLAC__bool ia32_fxsr = false;
FLAC__bool ia32_osxsave = false;
(void) ia32_fxsr; (void) ia32_osxsave; /* to avoid warnings about unused variables */
memset(info, 0, sizeof(*info));
info->type = FLAC__CPUINFO_TYPE_IA32;
#if !defined FLAC__NO_ASM && (defined FLAC__HAS_NASM || defined FLAC__HAS_X86INTRIN)
info->use_asm = true; /* we assume a minimum of 80386 with FLAC__CPU_IA32 */
#ifdef FLAC__HAS_X86INTRIN
if(!FLAC__cpu_have_cpuid_x86())
return;
#else
if(!FLAC__cpu_have_cpuid_asm_ia32())
return;
#endif
{
/* http://www.sandpile.org/x86/cpuid.htm */
#ifdef FLAC__HAS_X86INTRIN
FLAC__uint32 flags_eax, flags_ebx, flags_ecx, flags_edx;
FLAC__cpu_info_x86(1, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
#else
FLAC__uint32 flags_ecx, flags_edx;
FLAC__cpu_info_asm_ia32(&flags_edx, &flags_ecx);
#endif
info->ia32.cmov = (flags_edx & FLAC__CPUINFO_IA32_CPUID_CMOV )? true : false;
info->ia32.mmx = (flags_edx & FLAC__CPUINFO_IA32_CPUID_MMX )? true : false;
ia32_fxsr = (flags_edx & FLAC__CPUINFO_IA32_CPUID_FXSR )? true : false;
info->ia32.sse = (flags_edx & FLAC__CPUINFO_IA32_CPUID_SSE )? true : false;
info->ia32.sse2 = (flags_edx & FLAC__CPUINFO_IA32_CPUID_SSE2 )? true : false;
info->ia32.sse3 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE3 )? true : false;
info->ia32.ssse3 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSSE3)? true : false;
info->ia32.sse41 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE41)? true : false;
info->ia32.sse42 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE42)? true : false;
#if defined FLAC__HAS_X86INTRIN && defined FLAC__AVX_SUPPORTED
ia32_osxsave = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_OSXSAVE)? true : false;
info->ia32.avx = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_AVX )? true : false;
info->ia32.fma = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_FMA )? true : false;
FLAC__cpu_info_x86(7, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
info->ia32.avx2 = (flags_ebx & FLAC__CPUINFO_IA32_CPUID_AVX2 )? true : false;
#endif
}
#ifdef DEBUG
fprintf(stderr, "CPU info (IA-32):\n");
fprintf(stderr, " CMOV ....... %c\n", info->ia32.cmov ? 'Y' : 'n');
fprintf(stderr, " MMX ........ %c\n", info->ia32.mmx ? 'Y' : 'n');
fprintf(stderr, " SSE ........ %c\n", info->ia32.sse ? 'Y' : 'n');
fprintf(stderr, " SSE2 ....... %c\n", info->ia32.sse2 ? 'Y' : 'n');
fprintf(stderr, " SSE3 ....... %c\n", info->ia32.sse3 ? 'Y' : 'n');
fprintf(stderr, " SSSE3 ...... %c\n", info->ia32.ssse3 ? 'Y' : 'n');
fprintf(stderr, " SSE41 ...... %c\n", info->ia32.sse41 ? 'Y' : 'n');
fprintf(stderr, " SSE42 ...... %c\n", info->ia32.sse42 ? 'Y' : 'n');
# if defined FLAC__HAS_X86INTRIN && defined FLAC__AVX_SUPPORTED
fprintf(stderr, " AVX ........ %c\n", info->ia32.avx ? 'Y' : 'n');
fprintf(stderr, " FMA ........ %c\n", info->ia32.fma ? 'Y' : 'n');
fprintf(stderr, " AVX2 ....... %c\n", info->ia32.avx2 ? 'Y' : 'n');
# endif
#endif
/*
* now have to check for OS support of SSE instructions
*/
if(info->ia32.sse) {
#if defined FLAC__NO_SSE_OS
/* assume user knows better than us; turn it off */
disable_sse(info);
#elif defined FLAC__SSE_OS
/* assume user knows better than us; leave as detected above */
#elif defined(__FreeBSD__) || defined(__FreeBSD_kernel__) || defined(__DragonFly__) || defined(__APPLE__)
int sse = 0;
size_t len;
/* at least one of these must work: */
len = sizeof(sse); sse = sse || (sysctlbyname("hw.instruction_sse", &sse, &len, NULL, 0) == 0 && sse);
len = sizeof(sse); sse = sse || (sysctlbyname("hw.optional.sse" , &sse, &len, NULL, 0) == 0 && sse); /* __APPLE__ ? */
if(!sse)
disable_sse(info);
#elif defined(__NetBSD__) || defined (__OpenBSD__)
# if __NetBSD_Version__ >= 105250000 || (defined __OpenBSD__)
int val = 0, mib[2] = { CTL_MACHDEP, CPU_SSE };
size_t len = sizeof(val);
if(sysctl(mib, 2, &val, &len, NULL, 0) < 0 || !val)
disable_sse(info);
else { /* double-check SSE2 */
mib[1] = CPU_SSE2;
len = sizeof(val);
if(sysctl(mib, 2, &val, &len, NULL, 0) < 0 || !val) {
disable_sse(info);
info->ia32.sse = true;
}
}
# else
disable_sse(info);
# endif
#elif defined(__linux__)
int sse = 0;
struct sigaction sigill_save;
struct sigaction sigill_sse;
sigill_sse.sa_sigaction = sigill_handler_sse_os;
__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))
{
/* http://www.ibiblio.org/gferg/ldp/GCC-Inline-Assembly-HOWTO.html */
/* see sigill_handler_sse_os() for an explanation of the following: */
asm volatile (
"xorps %%xmm0,%%xmm0\n\t" /* will cause SIGILL if unsupported by OS */
"incl %0\n\t" /* SIGILL handler will jump over this */
/* landing zone */
"nop\n\t" /* SIGILL jump lands here if "inc" is 9 bytes */
"nop\n\t"
"nop\n\t"
"nop\n\t"
"nop\n\t"
"nop\n\t"
"nop\n\t" /* SIGILL jump lands here if "inc" is 3 bytes (expected) */
"nop\n\t"
"nop" /* SIGILL jump lands here if "inc" is 1 byte */
: "=r"(sse)
: "0"(sse)
);
sigaction(SIGILL, &sigill_save, NULL);
}
if(!sse)
disable_sse(info);
#elif defined(_MSC_VER)
__try {
__asm {
xorps xmm0,xmm0
}
}
__except(EXCEPTION_EXECUTE_HANDLER) {
if (_exception_code() == STATUS_ILLEGAL_INSTRUCTION)
disable_sse(info);
}
#elif defined(__GNUC__) /* MinGW goes here */
int sse = 0;
/* Based on the idea described in Agner Fog's manual "Optimizing subroutines in assembly language" */
/* In theory, not guaranteed to detect lack of OS SSE support on some future Intel CPUs, but in practice works (see the aforementioned manual) */
if (ia32_fxsr) {
struct {
FLAC__uint32 buff[128];
} __attribute__((aligned(16))) fxsr;
FLAC__uint32 old_val, new_val;
asm volatile ("fxsave %0" : "=m" (fxsr) : "m" (fxsr));
old_val = fxsr.buff[50];
fxsr.buff[50] ^= 0x0013c0de; /* change value in the buffer */
asm volatile ("fxrstor %0" : "=m" (fxsr) : "m" (fxsr)); /* try to change SSE register */
fxsr.buff[50] = old_val; /* restore old value in the buffer */
asm volatile ("fxsave %0 " : "=m" (fxsr) : "m" (fxsr)); /* old value will be overwritten if SSE register was changed */
new_val = fxsr.buff[50]; /* == old_val if FXRSTOR didn't change SSE register and (old_val ^ 0x0013c0de) otherwise */
fxsr.buff[50] = old_val; /* again restore old value in the buffer */
asm volatile ("fxrstor %0" : "=m" (fxsr) : "m" (fxsr)); /* restore old values of registers */
if ((old_val^new_val) == 0x0013c0de)
sse = 1;
}
if(!sse)
disable_sse(info);
#else
/* no way to test, disable to be safe */
disable_sse(info);
#endif
#ifdef DEBUG
fprintf(stderr, " SSE OS sup . %c\n", info->ia32.sse ? 'Y' : 'n');
#endif
}
else /* info->ia32.sse == false */
disable_sse(info);
/*
* now have to check for OS support of AVX instructions
*/
if(info->ia32.avx && ia32_osxsave) {
FLAC__uint32 ecr = FLAC__cpu_xgetbv_x86();
if ((ecr & 0x6) != 0x6)
disable_avx(info);
#ifdef DEBUG
fprintf(stderr, " AVX OS sup . %c\n", info->ia32.avx ? 'Y' : 'n');
#endif
}
else /* no OS AVX support*/
disable_avx(info);
#else
info->use_asm = false;
#endif
/*
* x86-64-specific
*/
#elif defined FLAC__CPU_X86_64
FLAC__bool x86_osxsave = false;
(void) x86_osxsave; /* to avoid warnings about unused variables */
memset(info, 0, sizeof(*info));
info->type = FLAC__CPUINFO_TYPE_X86_64;
#if !defined FLAC__NO_ASM && defined FLAC__HAS_X86INTRIN
info->use_asm = true;
{
/* http://www.sandpile.org/x86/cpuid.htm */
FLAC__uint32 flags_eax, flags_ebx, flags_ecx, flags_edx;
FLAC__cpu_info_x86(1, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
info->x86.sse3 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE3 )? true : false;
info->x86.ssse3 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSSE3)? true : false;
info->x86.sse41 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE41)? true : false;
info->x86.sse42 = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_SSE42)? true : false;
#if defined FLAC__AVX_SUPPORTED
x86_osxsave = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_OSXSAVE)? true : false;
info->x86.avx = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_AVX )? true : false;
info->x86.fma = (flags_ecx & FLAC__CPUINFO_IA32_CPUID_FMA )? true : false;
FLAC__cpu_info_x86(7, &flags_eax, &flags_ebx, &flags_ecx, &flags_edx);
info->x86.avx2 = (flags_ebx & FLAC__CPUINFO_IA32_CPUID_AVX2 )? true : false;
#endif
}
#ifdef DEBUG
fprintf(stderr, "CPU info (x86-64):\n");
fprintf(stderr, " SSE3 ....... %c\n", info->x86.sse3 ? 'Y' : 'n');
fprintf(stderr, " SSSE3 ...... %c\n", info->x86.ssse3 ? 'Y' : 'n');
fprintf(stderr, " SSE41 ...... %c\n", info->x86.sse41 ? 'Y' : 'n');
fprintf(stderr, " SSE42 ...... %c\n", info->x86.sse42 ? 'Y' : 'n');
# if defined FLAC__AVX_SUPPORTED
fprintf(stderr, " AVX ........ %c\n", info->x86.avx ? 'Y' : 'n');
fprintf(stderr, " FMA ........ %c\n", info->x86.fma ? 'Y' : 'n');
fprintf(stderr, " AVX2 ....... %c\n", info->x86.avx2 ? 'Y' : 'n');
# endif
#endif
/*
* now have to check for OS support of AVX instructions
*/
if(info->x86.avx && x86_osxsave) {
FLAC__uint32 ecr = FLAC__cpu_xgetbv_x86();
if ((ecr & 0x6) != 0x6)
disable_avx(info);
#ifdef DEBUG
fprintf(stderr, " AVX OS sup . %c\n", info->x86.avx ? 'Y' : 'n');
#endif
}
else /* no OS AVX support*/
disable_avx(info);
#else
info->use_asm = false;
#endif
/*
* unknown CPU
*/
#else
info->type = FLAC__CPUINFO_TYPE_UNKNOWN;
info->use_asm = false;
#endif
}
#if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
#if defined _MSC_VER
#include <intrin.h> /* for __cpuid() and _xgetbv() */
#elif defined __GNUC__ && defined HAVE_CPUID_H
#include <cpuid.h> /* for __get_cpuid() and __get_cpuid_max() */
#endif
FLAC__uint32 FLAC__cpu_have_cpuid_x86(void)
{
#ifdef FLAC__CPU_X86_64
return 1;
#else
# if defined _MSC_VER || defined __INTEL_COMPILER /* Do they support CPUs w/o CPUID support (or OSes that work on those CPUs)? */
FLAC__uint32 flags1, flags2;
__asm {
pushfd
pushfd
pop eax
mov flags1, eax
xor eax, 0x200000
push eax
popfd
pushfd
pop eax
mov flags2, eax
popfd
}
if (((flags1^flags2) & 0x200000) != 0)
return 1;
else
return 0;
# elif defined __GNUC__ && defined HAVE_CPUID_H
if (__get_cpuid_max(0, 0) != 0)
return 1;
else
return 0;
# else
return 0;
# endif
#endif
}
void FLAC__cpu_info_x86(FLAC__uint32 level, FLAC__uint32 *eax, FLAC__uint32 *ebx, FLAC__uint32 *ecx, FLAC__uint32 *edx)
{
(void) level;
#if defined _MSC_VER || defined __INTEL_COMPILER
int cpuinfo[4];
int ext = level & 0x80000000;
__cpuid(cpuinfo, ext);
if((unsigned)cpuinfo[0] < level) {
*eax = *ebx = *ecx = *edx = 0;
return;
}
#if defined FLAC__AVX_SUPPORTED
__cpuidex(cpuinfo, level, 0); /* for AVX2 detection */
#else
__cpuid(cpuinfo, level); /* some old compilers don't support __cpuidex */
#endif
*eax = cpuinfo[0]; *ebx = cpuinfo[1]; *ecx = cpuinfo[2]; *edx = cpuinfo[3];
#elif defined __GNUC__ && defined HAVE_CPUID_H
FLAC__uint32 ext = level & 0x80000000;
__cpuid(ext, *eax, *ebx, *ecx, *edx);
if (*eax < level) {
*eax = *ebx = *ecx = *edx = 0;
return;
}
__cpuid_count(level, 0, *eax, *ebx, *ecx, *edx);
#else
*eax = *ebx = *ecx = *edx = 0;
#endif
}
FLAC__uint32 FLAC__cpu_xgetbv_x86(void)
{
#if (defined _MSC_VER || defined __INTEL_COMPILER) && defined FLAC__AVX_SUPPORTED
return (FLAC__uint32)_xgetbv(0);
#elif defined __GNUC__
FLAC__uint32 lo, hi;
asm volatile (".byte 0x0f, 0x01, 0xd0" : "=a"(lo), "=d"(hi) : "c" (0));
return lo;
#else
return 0;
#endif
}
#endif /* (FLAC__CPU_IA32 || FLAC__CPU_X86_64) && FLAC__HAS_X86INTRIN */
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/crc.c
+143
View File
@@ -0,0 +1,143 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include "include/private/crc.h"
/* CRC-8, poly = x^8 + x^2 + x^1 + x^0, init = 0 */
FLAC__byte const FLAC__crc8_table[256] = {
0x00, 0x07, 0x0E, 0x09, 0x1C, 0x1B, 0x12, 0x15,
0x38, 0x3F, 0x36, 0x31, 0x24, 0x23, 0x2A, 0x2D,
0x70, 0x77, 0x7E, 0x79, 0x6C, 0x6B, 0x62, 0x65,
0x48, 0x4F, 0x46, 0x41, 0x54, 0x53, 0x5A, 0x5D,
0xE0, 0xE7, 0xEE, 0xE9, 0xFC, 0xFB, 0xF2, 0xF5,
0xD8, 0xDF, 0xD6, 0xD1, 0xC4, 0xC3, 0xCA, 0xCD,
0x90, 0x97, 0x9E, 0x99, 0x8C, 0x8B, 0x82, 0x85,
0xA8, 0xAF, 0xA6, 0xA1, 0xB4, 0xB3, 0xBA, 0xBD,
0xC7, 0xC0, 0xC9, 0xCE, 0xDB, 0xDC, 0xD5, 0xD2,
0xFF, 0xF8, 0xF1, 0xF6, 0xE3, 0xE4, 0xED, 0xEA,
0xB7, 0xB0, 0xB9, 0xBE, 0xAB, 0xAC, 0xA5, 0xA2,
0x8F, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9D, 0x9A,
0x27, 0x20, 0x29, 0x2E, 0x3B, 0x3C, 0x35, 0x32,
0x1F, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0D, 0x0A,
0x57, 0x50, 0x59, 0x5E, 0x4B, 0x4C, 0x45, 0x42,
0x6F, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7D, 0x7A,
0x89, 0x8E, 0x87, 0x80, 0x95, 0x92, 0x9B, 0x9C,
0xB1, 0xB6, 0xBF, 0xB8, 0xAD, 0xAA, 0xA3, 0xA4,
0xF9, 0xFE, 0xF7, 0xF0, 0xE5, 0xE2, 0xEB, 0xEC,
0xC1, 0xC6, 0xCF, 0xC8, 0xDD, 0xDA, 0xD3, 0xD4,
0x69, 0x6E, 0x67, 0x60, 0x75, 0x72, 0x7B, 0x7C,
0x51, 0x56, 0x5F, 0x58, 0x4D, 0x4A, 0x43, 0x44,
0x19, 0x1E, 0x17, 0x10, 0x05, 0x02, 0x0B, 0x0C,
0x21, 0x26, 0x2F, 0x28, 0x3D, 0x3A, 0x33, 0x34,
0x4E, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5C, 0x5B,
0x76, 0x71, 0x78, 0x7F, 0x6A, 0x6D, 0x64, 0x63,
0x3E, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2C, 0x2B,
0x06, 0x01, 0x08, 0x0F, 0x1A, 0x1D, 0x14, 0x13,
0xAE, 0xA9, 0xA0, 0xA7, 0xB2, 0xB5, 0xBC, 0xBB,
0x96, 0x91, 0x98, 0x9F, 0x8A, 0x8D, 0x84, 0x83,
0xDE, 0xD9, 0xD0, 0xD7, 0xC2, 0xC5, 0xCC, 0xCB,
0xE6, 0xE1, 0xE8, 0xEF, 0xFA, 0xFD, 0xF4, 0xF3
};
/* CRC-16, poly = x^16 + x^15 + x^2 + x^0, init = 0 */
unsigned const FLAC__crc16_table[256] = {
0x0000, 0x8005, 0x800f, 0x000a, 0x801b, 0x001e, 0x0014, 0x8011,
0x8033, 0x0036, 0x003c, 0x8039, 0x0028, 0x802d, 0x8027, 0x0022,
0x8063, 0x0066, 0x006c, 0x8069, 0x0078, 0x807d, 0x8077, 0x0072,
0x0050, 0x8055, 0x805f, 0x005a, 0x804b, 0x004e, 0x0044, 0x8041,
0x80c3, 0x00c6, 0x00cc, 0x80c9, 0x00d8, 0x80dd, 0x80d7, 0x00d2,
0x00f0, 0x80f5, 0x80ff, 0x00fa, 0x80eb, 0x00ee, 0x00e4, 0x80e1,
0x00a0, 0x80a5, 0x80af, 0x00aa, 0x80bb, 0x00be, 0x00b4, 0x80b1,
0x8093, 0x0096, 0x009c, 0x8099, 0x0088, 0x808d, 0x8087, 0x0082,
0x8183, 0x0186, 0x018c, 0x8189, 0x0198, 0x819d, 0x8197, 0x0192,
0x01b0, 0x81b5, 0x81bf, 0x01ba, 0x81ab, 0x01ae, 0x01a4, 0x81a1,
0x01e0, 0x81e5, 0x81ef, 0x01ea, 0x81fb, 0x01fe, 0x01f4, 0x81f1,
0x81d3, 0x01d6, 0x01dc, 0x81d9, 0x01c8, 0x81cd, 0x81c7, 0x01c2,
0x0140, 0x8145, 0x814f, 0x014a, 0x815b, 0x015e, 0x0154, 0x8151,
0x8173, 0x0176, 0x017c, 0x8179, 0x0168, 0x816d, 0x8167, 0x0162,
0x8123, 0x0126, 0x012c, 0x8129, 0x0138, 0x813d, 0x8137, 0x0132,
0x0110, 0x8115, 0x811f, 0x011a, 0x810b, 0x010e, 0x0104, 0x8101,
0x8303, 0x0306, 0x030c, 0x8309, 0x0318, 0x831d, 0x8317, 0x0312,
0x0330, 0x8335, 0x833f, 0x033a, 0x832b, 0x032e, 0x0324, 0x8321,
0x0360, 0x8365, 0x836f, 0x036a, 0x837b, 0x037e, 0x0374, 0x8371,
0x8353, 0x0356, 0x035c, 0x8359, 0x0348, 0x834d, 0x8347, 0x0342,
0x03c0, 0x83c5, 0x83cf, 0x03ca, 0x83db, 0x03de, 0x03d4, 0x83d1,
0x83f3, 0x03f6, 0x03fc, 0x83f9, 0x03e8, 0x83ed, 0x83e7, 0x03e2,
0x83a3, 0x03a6, 0x03ac, 0x83a9, 0x03b8, 0x83bd, 0x83b7, 0x03b2,
0x0390, 0x8395, 0x839f, 0x039a, 0x838b, 0x038e, 0x0384, 0x8381,
0x0280, 0x8285, 0x828f, 0x028a, 0x829b, 0x029e, 0x0294, 0x8291,
0x82b3, 0x02b6, 0x02bc, 0x82b9, 0x02a8, 0x82ad, 0x82a7, 0x02a2,
0x82e3, 0x02e6, 0x02ec, 0x82e9, 0x02f8, 0x82fd, 0x82f7, 0x02f2,
0x02d0, 0x82d5, 0x82df, 0x02da, 0x82cb, 0x02ce, 0x02c4, 0x82c1,
0x8243, 0x0246, 0x024c, 0x8249, 0x0258, 0x825d, 0x8257, 0x0252,
0x0270, 0x8275, 0x827f, 0x027a, 0x826b, 0x026e, 0x0264, 0x8261,
0x0220, 0x8225, 0x822f, 0x022a, 0x823b, 0x023e, 0x0234, 0x8231,
0x8213, 0x0216, 0x021c, 0x8219, 0x0208, 0x820d, 0x8207, 0x0202
};
void FLAC__crc8_update(const FLAC__byte data, FLAC__uint8 *crc)
{
*crc = FLAC__crc8_table[*crc ^ data];
}
void FLAC__crc8_update_block(const FLAC__byte *data, unsigned len, FLAC__uint8 *crc)
{
while(len--)
*crc = FLAC__crc8_table[*crc ^ *data++];
}
FLAC__uint8 FLAC__crc8(const FLAC__byte *data, unsigned len)
{
FLAC__uint8 crc = 0;
while(len--)
crc = FLAC__crc8_table[crc ^ *data++];
return crc;
}
unsigned FLAC__crc16(const FLAC__byte *data, unsigned len)
{
unsigned crc = 0;
while(len--)
crc = ((crc<<8) ^ FLAC__crc16_table[(crc>>8) ^ *data++]) & 0xffff;
return crc;
}
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/fixed.c
+418
View File
@@ -0,0 +1,418 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <math.h>
#include <string.h>
#include "../compat.h"
#include "include/private/bitmath.h"
#include "include/private/fixed.h"
#include "../assert.h"
#ifdef local_abs
#undef local_abs
#endif
#define local_abs(x) ((unsigned)((x)<0? -(x) : (x)))
#ifdef FLAC__INTEGER_ONLY_LIBRARY
/* rbps stands for residual bits per sample
*
* (ln(2) * err)
* rbps = log (-----------)
* 2 ( n )
*/
static FLAC__fixedpoint local__compute_rbps_integerized(FLAC__uint32 err, FLAC__uint32 n)
{
FLAC__uint32 rbps;
unsigned bits; /* the number of bits required to represent a number */
int fracbits; /* the number of bits of rbps that comprise the fractional part */
FLAC__ASSERT(sizeof(rbps) == sizeof(FLAC__fixedpoint));
FLAC__ASSERT(err > 0);
FLAC__ASSERT(n > 0);
FLAC__ASSERT(n <= FLAC__MAX_BLOCK_SIZE);
if(err <= n)
return 0;
/*
* The above two things tell us 1) n fits in 16 bits; 2) err/n > 1.
* These allow us later to know we won't lose too much precision in the
* fixed-point division (err<<fracbits)/n.
*/
fracbits = (8*sizeof(err)) - (FLAC__bitmath_ilog2(err)+1);
err <<= fracbits;
err /= n;
/* err now holds err/n with fracbits fractional bits */
/*
* Whittle err down to 16 bits max. 16 significant bits is enough for
* our purposes.
*/
FLAC__ASSERT(err > 0);
bits = FLAC__bitmath_ilog2(err)+1;
if(bits > 16) {
err >>= (bits-16);
fracbits -= (bits-16);
}
rbps = (FLAC__uint32)err;
/* Multiply by fixed-point version of ln(2), with 16 fractional bits */
rbps *= FLAC__FP_LN2;
fracbits += 16;
FLAC__ASSERT(fracbits >= 0);
/* FLAC__fixedpoint_log2 requires fracbits%4 to be 0 */
{
const int f = fracbits & 3;
if(f) {
rbps >>= f;
fracbits -= f;
}
}
rbps = FLAC__fixedpoint_log2(rbps, fracbits, (unsigned)(-1));
if(rbps == 0)
return 0;
/*
* The return value must have 16 fractional bits. Since the whole part
* of the base-2 log of a 32 bit number must fit in 5 bits, and fracbits
* must be >= -3, these assertion allows us to be able to shift rbps
* left if necessary to get 16 fracbits without losing any bits of the
* whole part of rbps.
*
* There is a slight chance due to accumulated error that the whole part
* will require 6 bits, so we use 6 in the assertion. Really though as
* long as it fits in 13 bits (32 - (16 - (-3))) we are fine.
*/
FLAC__ASSERT((int)FLAC__bitmath_ilog2(rbps)+1 <= fracbits + 6);
FLAC__ASSERT(fracbits >= -3);
/* now shift the decimal point into place */
if(fracbits < 16)
return rbps << (16-fracbits);
else if(fracbits > 16)
return rbps >> (fracbits-16);
else
return rbps;
}
static FLAC__fixedpoint local__compute_rbps_wide_integerized(FLAC__uint64 err, FLAC__uint32 n)
{
FLAC__uint32 rbps;
unsigned bits; /* the number of bits required to represent a number */
int fracbits; /* the number of bits of rbps that comprise the fractional part */
FLAC__ASSERT(sizeof(rbps) == sizeof(FLAC__fixedpoint));
FLAC__ASSERT(err > 0);
FLAC__ASSERT(n > 0);
FLAC__ASSERT(n <= FLAC__MAX_BLOCK_SIZE);
if(err <= n)
return 0;
/*
* The above two things tell us 1) n fits in 16 bits; 2) err/n > 1.
* These allow us later to know we won't lose too much precision in the
* fixed-point division (err<<fracbits)/n.
*/
fracbits = (8*sizeof(err)) - (FLAC__bitmath_ilog2_wide(err)+1);
err <<= fracbits;
err /= n;
/* err now holds err/n with fracbits fractional bits */
/*
* Whittle err down to 16 bits max. 16 significant bits is enough for
* our purposes.
*/
FLAC__ASSERT(err > 0);
bits = FLAC__bitmath_ilog2_wide(err)+1;
if(bits > 16) {
err >>= (bits-16);
fracbits -= (bits-16);
}
rbps = (FLAC__uint32)err;
/* Multiply by fixed-point version of ln(2), with 16 fractional bits */
rbps *= FLAC__FP_LN2;
fracbits += 16;
FLAC__ASSERT(fracbits >= 0);
/* FLAC__fixedpoint_log2 requires fracbits%4 to be 0 */
{
const int f = fracbits & 3;
if(f) {
rbps >>= f;
fracbits -= f;
}
}
rbps = FLAC__fixedpoint_log2(rbps, fracbits, (unsigned)(-1));
if(rbps == 0)
return 0;
/*
* The return value must have 16 fractional bits. Since the whole part
* of the base-2 log of a 32 bit number must fit in 5 bits, and fracbits
* must be >= -3, these assertion allows us to be able to shift rbps
* left if necessary to get 16 fracbits without losing any bits of the
* whole part of rbps.
*
* There is a slight chance due to accumulated error that the whole part
* will require 6 bits, so we use 6 in the assertion. Really though as
* long as it fits in 13 bits (32 - (16 - (-3))) we are fine.
*/
FLAC__ASSERT((int)FLAC__bitmath_ilog2(rbps)+1 <= fracbits + 6);
FLAC__ASSERT(fracbits >= -3);
/* now shift the decimal point into place */
if(fracbits < 16)
return rbps << (16-fracbits);
else if(fracbits > 16)
return rbps >> (fracbits-16);
else
return rbps;
}
#endif
#ifndef FLAC__INTEGER_ONLY_LIBRARY
unsigned FLAC__fixed_compute_best_predictor(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
#else
unsigned FLAC__fixed_compute_best_predictor(const FLAC__int32 data[], unsigned data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
#endif
{
FLAC__int32 last_error_0 = data[-1];
FLAC__int32 last_error_1 = data[-1] - data[-2];
FLAC__int32 last_error_2 = last_error_1 - (data[-2] - data[-3]);
FLAC__int32 last_error_3 = last_error_2 - (data[-2] - 2*data[-3] + data[-4]);
FLAC__int32 error, save;
FLAC__uint32 total_error_0 = 0, total_error_1 = 0, total_error_2 = 0, total_error_3 = 0, total_error_4 = 0;
unsigned i, order;
for(i = 0; i < data_len; i++) {
error = data[i] ; total_error_0 += local_abs(error); save = error;
error -= last_error_0; total_error_1 += local_abs(error); last_error_0 = save; save = error;
error -= last_error_1; total_error_2 += local_abs(error); last_error_1 = save; save = error;
error -= last_error_2; total_error_3 += local_abs(error); last_error_2 = save; save = error;
error -= last_error_3; total_error_4 += local_abs(error); last_error_3 = save;
}
if(total_error_0 < flac_min(flac_min(flac_min(total_error_1, total_error_2), total_error_3), total_error_4))
order = 0;
else if(total_error_1 < flac_min(flac_min(total_error_2, total_error_3), total_error_4))
order = 1;
else if(total_error_2 < flac_min(total_error_3, total_error_4))
order = 2;
else if(total_error_3 < total_error_4)
order = 3;
else
order = 4;
/* Estimate the expected number of bits per residual signal sample. */
/* 'total_error*' is linearly related to the variance of the residual */
/* signal, so we use it directly to compute E(|x|) */
FLAC__ASSERT(data_len > 0 || total_error_0 == 0);
FLAC__ASSERT(data_len > 0 || total_error_1 == 0);
FLAC__ASSERT(data_len > 0 || total_error_2 == 0);
FLAC__ASSERT(data_len > 0 || total_error_3 == 0);
FLAC__ASSERT(data_len > 0 || total_error_4 == 0);
#ifndef FLAC__INTEGER_ONLY_LIBRARY
residual_bits_per_sample[0] = (FLAC__float)((total_error_0 > 0) ? log(M_LN2 * (FLAC__double)total_error_0 / (FLAC__double)data_len) / M_LN2 : 0.0);
residual_bits_per_sample[1] = (FLAC__float)((total_error_1 > 0) ? log(M_LN2 * (FLAC__double)total_error_1 / (FLAC__double)data_len) / M_LN2 : 0.0);
residual_bits_per_sample[2] = (FLAC__float)((total_error_2 > 0) ? log(M_LN2 * (FLAC__double)total_error_2 / (FLAC__double)data_len) / M_LN2 : 0.0);
residual_bits_per_sample[3] = (FLAC__float)((total_error_3 > 0) ? log(M_LN2 * (FLAC__double)total_error_3 / (FLAC__double)data_len) / M_LN2 : 0.0);
residual_bits_per_sample[4] = (FLAC__float)((total_error_4 > 0) ? log(M_LN2 * (FLAC__double)total_error_4 / (FLAC__double)data_len) / M_LN2 : 0.0);
#else
residual_bits_per_sample[0] = (total_error_0 > 0) ? local__compute_rbps_integerized(total_error_0, data_len) : 0;
residual_bits_per_sample[1] = (total_error_1 > 0) ? local__compute_rbps_integerized(total_error_1, data_len) : 0;
residual_bits_per_sample[2] = (total_error_2 > 0) ? local__compute_rbps_integerized(total_error_2, data_len) : 0;
residual_bits_per_sample[3] = (total_error_3 > 0) ? local__compute_rbps_integerized(total_error_3, data_len) : 0;
residual_bits_per_sample[4] = (total_error_4 > 0) ? local__compute_rbps_integerized(total_error_4, data_len) : 0;
#endif
return order;
}
#ifndef FLAC__INTEGER_ONLY_LIBRARY
unsigned FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
#else
unsigned FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], unsigned data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1])
#endif
{
FLAC__int32 last_error_0 = data[-1];
FLAC__int32 last_error_1 = data[-1] - data[-2];
FLAC__int32 last_error_2 = last_error_1 - (data[-2] - data[-3]);
FLAC__int32 last_error_3 = last_error_2 - (data[-2] - 2*data[-3] + data[-4]);
FLAC__int32 error, save;
/* total_error_* are 64-bits to avoid overflow when encoding
* erratic signals when the bits-per-sample and blocksize are
* large.
*/
FLAC__uint64 total_error_0 = 0, total_error_1 = 0, total_error_2 = 0, total_error_3 = 0, total_error_4 = 0;
unsigned i, order;
for(i = 0; i < data_len; i++) {
error = data[i] ; total_error_0 += local_abs(error); save = error;
error -= last_error_0; total_error_1 += local_abs(error); last_error_0 = save; save = error;
error -= last_error_1; total_error_2 += local_abs(error); last_error_1 = save; save = error;
error -= last_error_2; total_error_3 += local_abs(error); last_error_2 = save; save = error;
error -= last_error_3; total_error_4 += local_abs(error); last_error_3 = save;
}
if(total_error_0 < flac_min(flac_min(flac_min(total_error_1, total_error_2), total_error_3), total_error_4))
order = 0;
else if(total_error_1 < flac_min(flac_min(total_error_2, total_error_3), total_error_4))
order = 1;
else if(total_error_2 < flac_min(total_error_3, total_error_4))
order = 2;
else if(total_error_3 < total_error_4)
order = 3;
else
order = 4;
/* Estimate the expected number of bits per residual signal sample. */
/* 'total_error*' is linearly related to the variance of the residual */
/* signal, so we use it directly to compute E(|x|) */
FLAC__ASSERT(data_len > 0 || total_error_0 == 0);
FLAC__ASSERT(data_len > 0 || total_error_1 == 0);
FLAC__ASSERT(data_len > 0 || total_error_2 == 0);
FLAC__ASSERT(data_len > 0 || total_error_3 == 0);
FLAC__ASSERT(data_len > 0 || total_error_4 == 0);
#ifndef FLAC__INTEGER_ONLY_LIBRARY
residual_bits_per_sample[0] = (FLAC__float)((total_error_0 > 0) ? log(M_LN2 * (FLAC__double)total_error_0 / (FLAC__double)data_len) / M_LN2 : 0.0);
residual_bits_per_sample[1] = (FLAC__float)((total_error_1 > 0) ? log(M_LN2 * (FLAC__double)total_error_1 / (FLAC__double)data_len) / M_LN2 : 0.0);
residual_bits_per_sample[2] = (FLAC__float)((total_error_2 > 0) ? log(M_LN2 * (FLAC__double)total_error_2 / (FLAC__double)data_len) / M_LN2 : 0.0);
residual_bits_per_sample[3] = (FLAC__float)((total_error_3 > 0) ? log(M_LN2 * (FLAC__double)total_error_3 / (FLAC__double)data_len) / M_LN2 : 0.0);
residual_bits_per_sample[4] = (FLAC__float)((total_error_4 > 0) ? log(M_LN2 * (FLAC__double)total_error_4 / (FLAC__double)data_len) / M_LN2 : 0.0);
#else
residual_bits_per_sample[0] = (total_error_0 > 0) ? local__compute_rbps_wide_integerized(total_error_0, data_len) : 0;
residual_bits_per_sample[1] = (total_error_1 > 0) ? local__compute_rbps_wide_integerized(total_error_1, data_len) : 0;
residual_bits_per_sample[2] = (total_error_2 > 0) ? local__compute_rbps_wide_integerized(total_error_2, data_len) : 0;
residual_bits_per_sample[3] = (total_error_3 > 0) ? local__compute_rbps_wide_integerized(total_error_3, data_len) : 0;
residual_bits_per_sample[4] = (total_error_4 > 0) ? local__compute_rbps_wide_integerized(total_error_4, data_len) : 0;
#endif
return order;
}
void FLAC__fixed_compute_residual(const FLAC__int32 data[], unsigned data_len, unsigned order, FLAC__int32 residual[])
{
const int idata_len = (int)data_len;
int i;
switch(order) {
case 0:
FLAC__ASSERT(sizeof(residual[0]) == sizeof(data[0]));
memcpy(residual, data, sizeof(residual[0])*data_len);
break;
case 1:
for(i = 0; i < idata_len; i++)
residual[i] = data[i] - data[i-1];
break;
case 2:
for(i = 0; i < idata_len; i++)
#if 1 /* OPT: may be faster with some compilers on some systems */
residual[i] = data[i] - (data[i-1] << 1) + data[i-2];
#else
residual[i] = data[i] - 2*data[i-1] + data[i-2];
#endif
break;
case 3:
for(i = 0; i < idata_len; i++)
#if 1 /* OPT: may be faster with some compilers on some systems */
residual[i] = data[i] - (((data[i-1]-data[i-2])<<1) + (data[i-1]-data[i-2])) - data[i-3];
#else
residual[i] = data[i] - 3*data[i-1] + 3*data[i-2] - data[i-3];
#endif
break;
case 4:
for(i = 0; i < idata_len; i++)
#if 1 /* OPT: may be faster with some compilers on some systems */
residual[i] = data[i] - ((data[i-1]+data[i-3])<<2) + ((data[i-2]<<2) + (data[i-2]<<1)) + data[i-4];
#else
residual[i] = data[i] - 4*data[i-1] + 6*data[i-2] - 4*data[i-3] + data[i-4];
#endif
break;
default:
FLAC__ASSERT(0);
}
}
void FLAC__fixed_restore_signal(const FLAC__int32 residual[], unsigned data_len, unsigned order, FLAC__int32 data[])
{
int i, idata_len = (int)data_len;
switch(order) {
case 0:
FLAC__ASSERT(sizeof(residual[0]) == sizeof(data[0]));
memcpy(data, residual, sizeof(residual[0])*data_len);
break;
case 1:
for(i = 0; i < idata_len; i++)
data[i] = residual[i] + data[i-1];
break;
case 2:
for(i = 0; i < idata_len; i++)
#if 1 /* OPT: may be faster with some compilers on some systems */
data[i] = residual[i] + (data[i-1]<<1) - data[i-2];
#else
data[i] = residual[i] + 2*data[i-1] - data[i-2];
#endif
break;
case 3:
for(i = 0; i < idata_len; i++)
#if 1 /* OPT: may be faster with some compilers on some systems */
data[i] = residual[i] + (((data[i-1]-data[i-2])<<1) + (data[i-1]-data[i-2])) + data[i-3];
#else
data[i] = residual[i] + 3*data[i-1] - 3*data[i-2] + data[i-3];
#endif
break;
case 4:
for(i = 0; i < idata_len; i++)
#if 1 /* OPT: may be faster with some compilers on some systems */
data[i] = residual[i] + ((data[i-1]+data[i-3])<<2) - ((data[i-2]<<2) + (data[i-2]<<1)) - data[i-4];
#else
data[i] = residual[i] + 4*data[i-1] - 6*data[i-2] + 4*data[i-3] - data[i-4];
#endif
break;
default:
FLAC__ASSERT(0);
}
}
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/float.c
+302
View File
@@ -0,0 +1,302 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2004-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include "../assert.h"
#include "../compat.h"
#include "include/private/float.h"
#ifdef FLAC__INTEGER_ONLY_LIBRARY
const FLAC__fixedpoint FLAC__FP_ZERO = 0;
const FLAC__fixedpoint FLAC__FP_ONE_HALF = 0x00008000;
const FLAC__fixedpoint FLAC__FP_ONE = 0x00010000;
const FLAC__fixedpoint FLAC__FP_LN2 = 45426;
const FLAC__fixedpoint FLAC__FP_E = 178145;
/* Lookup tables for Knuth's logarithm algorithm */
#define LOG2_LOOKUP_PRECISION 16
static const FLAC__uint32 log2_lookup[][LOG2_LOOKUP_PRECISION] = {
{
/*
* 0 fraction bits
*/
/* undefined */ 0x00000000,
/* lg(2/1) = */ 0x00000001,
/* lg(4/3) = */ 0x00000000,
/* lg(8/7) = */ 0x00000000,
/* lg(16/15) = */ 0x00000000,
/* lg(32/31) = */ 0x00000000,
/* lg(64/63) = */ 0x00000000,
/* lg(128/127) = */ 0x00000000,
/* lg(256/255) = */ 0x00000000,
/* lg(512/511) = */ 0x00000000,
/* lg(1024/1023) = */ 0x00000000,
/* lg(2048/2047) = */ 0x00000000,
/* lg(4096/4095) = */ 0x00000000,
/* lg(8192/8191) = */ 0x00000000,
/* lg(16384/16383) = */ 0x00000000,
/* lg(32768/32767) = */ 0x00000000
},
{
/*
* 4 fraction bits
*/
/* undefined */ 0x00000000,
/* lg(2/1) = */ 0x00000010,
/* lg(4/3) = */ 0x00000007,
/* lg(8/7) = */ 0x00000003,
/* lg(16/15) = */ 0x00000001,
/* lg(32/31) = */ 0x00000001,
/* lg(64/63) = */ 0x00000000,
/* lg(128/127) = */ 0x00000000,
/* lg(256/255) = */ 0x00000000,
/* lg(512/511) = */ 0x00000000,
/* lg(1024/1023) = */ 0x00000000,
/* lg(2048/2047) = */ 0x00000000,
/* lg(4096/4095) = */ 0x00000000,
/* lg(8192/8191) = */ 0x00000000,
/* lg(16384/16383) = */ 0x00000000,
/* lg(32768/32767) = */ 0x00000000
},
{
/*
* 8 fraction bits
*/
/* undefined */ 0x00000000,
/* lg(2/1) = */ 0x00000100,
/* lg(4/3) = */ 0x0000006a,
/* lg(8/7) = */ 0x00000031,
/* lg(16/15) = */ 0x00000018,
/* lg(32/31) = */ 0x0000000c,
/* lg(64/63) = */ 0x00000006,
/* lg(128/127) = */ 0x00000003,
/* lg(256/255) = */ 0x00000001,
/* lg(512/511) = */ 0x00000001,
/* lg(1024/1023) = */ 0x00000000,
/* lg(2048/2047) = */ 0x00000000,
/* lg(4096/4095) = */ 0x00000000,
/* lg(8192/8191) = */ 0x00000000,
/* lg(16384/16383) = */ 0x00000000,
/* lg(32768/32767) = */ 0x00000000
},
{
/*
* 12 fraction bits
*/
/* undefined */ 0x00000000,
/* lg(2/1) = */ 0x00001000,
/* lg(4/3) = */ 0x000006a4,
/* lg(8/7) = */ 0x00000315,
/* lg(16/15) = */ 0x0000017d,
/* lg(32/31) = */ 0x000000bc,
/* lg(64/63) = */ 0x0000005d,
/* lg(128/127) = */ 0x0000002e,
/* lg(256/255) = */ 0x00000017,
/* lg(512/511) = */ 0x0000000c,
/* lg(1024/1023) = */ 0x00000006,
/* lg(2048/2047) = */ 0x00000003,
/* lg(4096/4095) = */ 0x00000001,
/* lg(8192/8191) = */ 0x00000001,
/* lg(16384/16383) = */ 0x00000000,
/* lg(32768/32767) = */ 0x00000000
},
{
/*
* 16 fraction bits
*/
/* undefined */ 0x00000000,
/* lg(2/1) = */ 0x00010000,
/* lg(4/3) = */ 0x00006a40,
/* lg(8/7) = */ 0x00003151,
/* lg(16/15) = */ 0x000017d6,
/* lg(32/31) = */ 0x00000bba,
/* lg(64/63) = */ 0x000005d1,
/* lg(128/127) = */ 0x000002e6,
/* lg(256/255) = */ 0x00000172,
/* lg(512/511) = */ 0x000000b9,
/* lg(1024/1023) = */ 0x0000005c,
/* lg(2048/2047) = */ 0x0000002e,
/* lg(4096/4095) = */ 0x00000017,
/* lg(8192/8191) = */ 0x0000000c,
/* lg(16384/16383) = */ 0x00000006,
/* lg(32768/32767) = */ 0x00000003
},
{
/*
* 20 fraction bits
*/
/* undefined */ 0x00000000,
/* lg(2/1) = */ 0x00100000,
/* lg(4/3) = */ 0x0006a3fe,
/* lg(8/7) = */ 0x00031513,
/* lg(16/15) = */ 0x00017d60,
/* lg(32/31) = */ 0x0000bb9d,
/* lg(64/63) = */ 0x00005d10,
/* lg(128/127) = */ 0x00002e59,
/* lg(256/255) = */ 0x00001721,
/* lg(512/511) = */ 0x00000b8e,
/* lg(1024/1023) = */ 0x000005c6,
/* lg(2048/2047) = */ 0x000002e3,
/* lg(4096/4095) = */ 0x00000171,
/* lg(8192/8191) = */ 0x000000b9,
/* lg(16384/16383) = */ 0x0000005c,
/* lg(32768/32767) = */ 0x0000002e
},
{
/*
* 24 fraction bits
*/
/* undefined */ 0x00000000,
/* lg(2/1) = */ 0x01000000,
/* lg(4/3) = */ 0x006a3fe6,
/* lg(8/7) = */ 0x00315130,
/* lg(16/15) = */ 0x0017d605,
/* lg(32/31) = */ 0x000bb9ca,
/* lg(64/63) = */ 0x0005d0fc,
/* lg(128/127) = */ 0x0002e58f,
/* lg(256/255) = */ 0x0001720e,
/* lg(512/511) = */ 0x0000b8d8,
/* lg(1024/1023) = */ 0x00005c61,
/* lg(2048/2047) = */ 0x00002e2d,
/* lg(4096/4095) = */ 0x00001716,
/* lg(8192/8191) = */ 0x00000b8b,
/* lg(16384/16383) = */ 0x000005c5,
/* lg(32768/32767) = */ 0x000002e3
},
{
/*
* 28 fraction bits
*/
/* undefined */ 0x00000000,
/* lg(2/1) = */ 0x10000000,
/* lg(4/3) = */ 0x06a3fe5c,
/* lg(8/7) = */ 0x03151301,
/* lg(16/15) = */ 0x017d6049,
/* lg(32/31) = */ 0x00bb9ca6,
/* lg(64/63) = */ 0x005d0fba,
/* lg(128/127) = */ 0x002e58f7,
/* lg(256/255) = */ 0x001720da,
/* lg(512/511) = */ 0x000b8d87,
/* lg(1024/1023) = */ 0x0005c60b,
/* lg(2048/2047) = */ 0x0002e2d7,
/* lg(4096/4095) = */ 0x00017160,
/* lg(8192/8191) = */ 0x0000b8ad,
/* lg(16384/16383) = */ 0x00005c56,
/* lg(32768/32767) = */ 0x00002e2b
}
};
#if 0
static const FLAC__uint64 log2_lookup_wide[] = {
{
/*
* 32 fraction bits
*/
/* undefined */ 0x00000000,
/* lg(2/1) = */ FLAC__U64L(0x100000000),
/* lg(4/3) = */ FLAC__U64L(0x6a3fe5c6),
/* lg(8/7) = */ FLAC__U64L(0x31513015),
/* lg(16/15) = */ FLAC__U64L(0x17d60497),
/* lg(32/31) = */ FLAC__U64L(0x0bb9ca65),
/* lg(64/63) = */ FLAC__U64L(0x05d0fba2),
/* lg(128/127) = */ FLAC__U64L(0x02e58f74),
/* lg(256/255) = */ FLAC__U64L(0x01720d9c),
/* lg(512/511) = */ FLAC__U64L(0x00b8d875),
/* lg(1024/1023) = */ FLAC__U64L(0x005c60aa),
/* lg(2048/2047) = */ FLAC__U64L(0x002e2d72),
/* lg(4096/4095) = */ FLAC__U64L(0x00171600),
/* lg(8192/8191) = */ FLAC__U64L(0x000b8ad2),
/* lg(16384/16383) = */ FLAC__U64L(0x0005c55d),
/* lg(32768/32767) = */ FLAC__U64L(0x0002e2ac)
},
{
/*
* 48 fraction bits
*/
/* undefined */ 0x00000000,
/* lg(2/1) = */ FLAC__U64L(0x1000000000000),
/* lg(4/3) = */ FLAC__U64L(0x6a3fe5c60429),
/* lg(8/7) = */ FLAC__U64L(0x315130157f7a),
/* lg(16/15) = */ FLAC__U64L(0x17d60496cfbb),
/* lg(32/31) = */ FLAC__U64L(0xbb9ca64ecac),
/* lg(64/63) = */ FLAC__U64L(0x5d0fba187cd),
/* lg(128/127) = */ FLAC__U64L(0x2e58f7441ee),
/* lg(256/255) = */ FLAC__U64L(0x1720d9c06a8),
/* lg(512/511) = */ FLAC__U64L(0xb8d8752173),
/* lg(1024/1023) = */ FLAC__U64L(0x5c60aa252e),
/* lg(2048/2047) = */ FLAC__U64L(0x2e2d71b0d8),
/* lg(4096/4095) = */ FLAC__U64L(0x1716001719),
/* lg(8192/8191) = */ FLAC__U64L(0xb8ad1de1b),
/* lg(16384/16383) = */ FLAC__U64L(0x5c55d640d),
/* lg(32768/32767) = */ FLAC__U64L(0x2e2abcf52)
}
};
#endif
FLAC__uint32 FLAC__fixedpoint_log2(FLAC__uint32 x, unsigned fracbits, unsigned precision)
{
const FLAC__uint32 ONE = (1u << fracbits);
const FLAC__uint32 *table = log2_lookup[fracbits >> 2];
FLAC__ASSERT(fracbits < 32);
FLAC__ASSERT((fracbits & 0x3) == 0);
if(x < ONE)
return 0;
if(precision > LOG2_LOOKUP_PRECISION)
precision = LOG2_LOOKUP_PRECISION;
/* Knuth's algorithm for computing logarithms, optimized for base-2 with lookup tables */
{
FLAC__uint32 y = 0;
FLAC__uint32 z = x >> 1, k = 1;
while (x > ONE && k < precision) {
if (x - z >= ONE) {
x -= z;
z = x >> k;
y += table[k];
}
else {
z >>= 1;
k++;
}
}
return y;
}
}
#endif /* defined FLAC__INTEGER_ONLY_LIBRARY */
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/format.c
+584
View File
@@ -0,0 +1,584 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <stdio.h>
#include <stdlib.h> /* for qsort() */
#include <string.h> /* for memset() */
#include "../assert.h"
#include "../format.h"
#include "../compat.h"
#include "include/private/format.h"
/* VERSION should come from configure */
FLAC_API const char *FLAC__VERSION_STRING = VERSION;
FLAC_API const char *FLAC__VENDOR_STRING = "reference libFLAC " VERSION " 20141125";
FLAC_API const FLAC__byte FLAC__STREAM_SYNC_STRING[4] = { 'f','L','a','C' };
FLAC_API const unsigned FLAC__STREAM_SYNC = 0x664C6143;
FLAC_API const unsigned FLAC__STREAM_SYNC_LEN = 32; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MIN_BLOCK_SIZE_LEN = 16; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MAX_BLOCK_SIZE_LEN = 16; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MIN_FRAME_SIZE_LEN = 24; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MAX_FRAME_SIZE_LEN = 24; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_SAMPLE_RATE_LEN = 20; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_CHANNELS_LEN = 3; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN = 5; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_TOTAL_SAMPLES_LEN = 36; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_STREAMINFO_MD5SUM_LEN = 128; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_APPLICATION_ID_LEN = 32; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_SEEKPOINT_SAMPLE_NUMBER_LEN = 64; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_SEEKPOINT_STREAM_OFFSET_LEN = 64; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_SEEKPOINT_FRAME_SAMPLES_LEN = 16; /* bits */
FLAC_API const FLAC__uint64 FLAC__STREAM_METADATA_SEEKPOINT_PLACEHOLDER = FLAC__U64L(0xffffffffffffffff);
FLAC_API const unsigned FLAC__STREAM_METADATA_VORBIS_COMMENT_ENTRY_LENGTH_LEN = 32; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_VORBIS_COMMENT_NUM_COMMENTS_LEN = 32; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_INDEX_OFFSET_LEN = 64; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_INDEX_NUMBER_LEN = 8; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_INDEX_RESERVED_LEN = 3*8; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_OFFSET_LEN = 64; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_NUMBER_LEN = 8; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_ISRC_LEN = 12*8; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_TYPE_LEN = 1; /* bit */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_PRE_EMPHASIS_LEN = 1; /* bit */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_RESERVED_LEN = 6+13*8; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_TRACK_NUM_INDICES_LEN = 8; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_MEDIA_CATALOG_NUMBER_LEN = 128*8; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_LEAD_IN_LEN = 64; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_IS_CD_LEN = 1; /* bit */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_RESERVED_LEN = 7+258*8; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_CUESHEET_NUM_TRACKS_LEN = 8; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_TYPE_LEN = 32; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_MIME_TYPE_LENGTH_LEN = 32; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_DESCRIPTION_LENGTH_LEN = 32; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_WIDTH_LEN = 32; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_HEIGHT_LEN = 32; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_DEPTH_LEN = 32; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_COLORS_LEN = 32; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_PICTURE_DATA_LENGTH_LEN = 32; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_IS_LAST_LEN = 1; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_TYPE_LEN = 7; /* bits */
FLAC_API const unsigned FLAC__STREAM_METADATA_LENGTH_LEN = 24; /* bits */
FLAC_API const unsigned FLAC__FRAME_HEADER_SYNC = 0x3ffe;
FLAC_API const unsigned FLAC__FRAME_HEADER_SYNC_LEN = 14; /* bits */
FLAC_API const unsigned FLAC__FRAME_HEADER_RESERVED_LEN = 1; /* bits */
FLAC_API const unsigned FLAC__FRAME_HEADER_BLOCKING_STRATEGY_LEN = 1; /* bits */
FLAC_API const unsigned FLAC__FRAME_HEADER_BLOCK_SIZE_LEN = 4; /* bits */
FLAC_API const unsigned FLAC__FRAME_HEADER_SAMPLE_RATE_LEN = 4; /* bits */
FLAC_API const unsigned FLAC__FRAME_HEADER_CHANNEL_ASSIGNMENT_LEN = 4; /* bits */
FLAC_API const unsigned FLAC__FRAME_HEADER_BITS_PER_SAMPLE_LEN = 3; /* bits */
FLAC_API const unsigned FLAC__FRAME_HEADER_ZERO_PAD_LEN = 1; /* bits */
FLAC_API const unsigned FLAC__FRAME_HEADER_CRC_LEN = 8; /* bits */
FLAC_API const unsigned FLAC__FRAME_FOOTER_CRC_LEN = 16; /* bits */
FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_TYPE_LEN = 2; /* bits */
FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_ORDER_LEN = 4; /* bits */
FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_PARAMETER_LEN = 4; /* bits */
FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_PARAMETER_LEN = 5; /* bits */
FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_RAW_LEN = 5; /* bits */
FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_ESCAPE_PARAMETER = 15; /* == (1<<FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_PARAMETER_LEN)-1 */
FLAC_API const unsigned FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_ESCAPE_PARAMETER = 31; /* == (1<<FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_PARAMETER_LEN)-1 */
FLAC_API const char * const FLAC__EntropyCodingMethodTypeString[] = {
"PARTITIONED_RICE",
"PARTITIONED_RICE2"
};
FLAC_API const unsigned FLAC__SUBFRAME_LPC_QLP_COEFF_PRECISION_LEN = 4; /* bits */
FLAC_API const unsigned FLAC__SUBFRAME_LPC_QLP_SHIFT_LEN = 5; /* bits */
FLAC_API const unsigned FLAC__SUBFRAME_ZERO_PAD_LEN = 1; /* bits */
FLAC_API const unsigned FLAC__SUBFRAME_TYPE_LEN = 6; /* bits */
FLAC_API const unsigned FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN = 1; /* bits */
FLAC_API const unsigned FLAC__SUBFRAME_TYPE_CONSTANT_BYTE_ALIGNED_MASK = 0x00;
FLAC_API const unsigned FLAC__SUBFRAME_TYPE_VERBATIM_BYTE_ALIGNED_MASK = 0x02;
FLAC_API const unsigned FLAC__SUBFRAME_TYPE_FIXED_BYTE_ALIGNED_MASK = 0x10;
FLAC_API const unsigned FLAC__SUBFRAME_TYPE_LPC_BYTE_ALIGNED_MASK = 0x40;
FLAC_API const char * const FLAC__SubframeTypeString[] = {
"CONSTANT",
"VERBATIM",
"FIXED",
"LPC"
};
FLAC_API const char * const FLAC__ChannelAssignmentString[] = {
"INDEPENDENT",
"LEFT_SIDE",
"RIGHT_SIDE",
"MID_SIDE"
};
FLAC_API const char * const FLAC__FrameNumberTypeString[] = {
"FRAME_NUMBER_TYPE_FRAME_NUMBER",
"FRAME_NUMBER_TYPE_SAMPLE_NUMBER"
};
FLAC_API const char * const FLAC__MetadataTypeString[] = {
"STREAMINFO",
"PADDING",
"APPLICATION",
"SEEKTABLE",
"VORBIS_COMMENT",
"CUESHEET",
"PICTURE"
};
FLAC_API const char * const FLAC__StreamMetadata_Picture_TypeString[] = {
"Other",
"32x32 pixels 'file icon' (PNG only)",
"Other file icon",
"Cover (front)",
"Cover (back)",
"Leaflet page",
"Media (e.g. label side of CD)",
"Lead artist/lead performer/soloist",
"Artist/performer",
"Conductor",
"Band/Orchestra",
"Composer",
"Lyricist/text writer",
"Recording Location",
"During recording",
"During performance",
"Movie/video screen capture",
"A bright coloured fish",
"Illustration",
"Band/artist logotype",
"Publisher/Studio logotype"
};
FLAC_API FLAC__bool FLAC__format_sample_rate_is_valid(unsigned sample_rate)
{
if(sample_rate == 0 || sample_rate > FLAC__MAX_SAMPLE_RATE) {
return false;
}
else
return true;
}
FLAC_API FLAC__bool FLAC__format_blocksize_is_subset(unsigned blocksize, unsigned sample_rate)
{
if(blocksize > 16384)
return false;
else if(sample_rate <= 48000 && blocksize > 4608)
return false;
else
return true;
}
FLAC_API FLAC__bool FLAC__format_sample_rate_is_subset(unsigned sample_rate)
{
if(
!FLAC__format_sample_rate_is_valid(sample_rate) ||
(
sample_rate >= (1u << 16) &&
!(sample_rate % 1000 == 0 || sample_rate % 10 == 0)
)
) {
return false;
}
else
return true;
}
/* @@@@ add to unit tests; it is already indirectly tested by the metadata_object tests */
FLAC_API FLAC__bool FLAC__format_seektable_is_legal(const FLAC__StreamMetadata_SeekTable *seek_table)
{
unsigned i;
FLAC__uint64 prev_sample_number = 0;
FLAC__bool got_prev = false;
FLAC__ASSERT(0 != seek_table);
for(i = 0; i < seek_table->num_points; i++) {
if(got_prev) {
if(
seek_table->points[i].sample_number != FLAC__STREAM_METADATA_SEEKPOINT_PLACEHOLDER &&
seek_table->points[i].sample_number <= prev_sample_number
)
return false;
}
prev_sample_number = seek_table->points[i].sample_number;
got_prev = true;
}
return true;
}
/* used as the sort predicate for qsort() */
static int seekpoint_compare_(const FLAC__StreamMetadata_SeekPoint *l, const FLAC__StreamMetadata_SeekPoint *r)
{
/* we don't just 'return l->sample_number - r->sample_number' since the result (FLAC__int64) might overflow an 'int' */
if(l->sample_number == r->sample_number)
return 0;
else if(l->sample_number < r->sample_number)
return -1;
else
return 1;
}
/* @@@@ add to unit tests; it is already indirectly tested by the metadata_object tests */
FLAC_API unsigned FLAC__format_seektable_sort(FLAC__StreamMetadata_SeekTable *seek_table)
{
unsigned i, j;
FLAC__bool first;
FLAC__ASSERT(0 != seek_table);
/* sort the seekpoints */
qsort(seek_table->points, seek_table->num_points, sizeof(FLAC__StreamMetadata_SeekPoint), (int (*)(const void *, const void *))seekpoint_compare_);
/* uniquify the seekpoints */
first = true;
for(i = j = 0; i < seek_table->num_points; i++) {
if(seek_table->points[i].sample_number != FLAC__STREAM_METADATA_SEEKPOINT_PLACEHOLDER) {
if(!first) {
if(seek_table->points[i].sample_number == seek_table->points[j-1].sample_number)
continue;
}
}
first = false;
seek_table->points[j++] = seek_table->points[i];
}
for(i = j; i < seek_table->num_points; i++) {
seek_table->points[i].sample_number = FLAC__STREAM_METADATA_SEEKPOINT_PLACEHOLDER;
seek_table->points[i].stream_offset = 0;
seek_table->points[i].frame_samples = 0;
}
return j;
}
/*
* also disallows non-shortest-form encodings, c.f.
* http://www.unicode.org/versions/corrigendum1.html
* and a more clear explanation at the end of this section:
* http://www.cl.cam.ac.uk/~mgk25/unicode.html#utf-8
*/
static unsigned utf8len_(const FLAC__byte *utf8)
{
FLAC__ASSERT(0 != utf8);
if ((utf8[0] & 0x80) == 0) {
return 1;
}
else if ((utf8[0] & 0xE0) == 0xC0 && (utf8[1] & 0xC0) == 0x80) {
if ((utf8[0] & 0xFE) == 0xC0) /* overlong sequence check */
return 0;
return 2;
}
else if ((utf8[0] & 0xF0) == 0xE0 && (utf8[1] & 0xC0) == 0x80 && (utf8[2] & 0xC0) == 0x80) {
if (utf8[0] == 0xE0 && (utf8[1] & 0xE0) == 0x80) /* overlong sequence check */
return 0;
/* illegal surrogates check (U+D800...U+DFFF and U+FFFE...U+FFFF) */
if (utf8[0] == 0xED && (utf8[1] & 0xE0) == 0xA0) /* D800-DFFF */
return 0;
if (utf8[0] == 0xEF && utf8[1] == 0xBF && (utf8[2] & 0xFE) == 0xBE) /* FFFE-FFFF */
return 0;
return 3;
}
else if ((utf8[0] & 0xF8) == 0xF0 && (utf8[1] & 0xC0) == 0x80 && (utf8[2] & 0xC0) == 0x80 && (utf8[3] & 0xC0) == 0x80) {
if (utf8[0] == 0xF0 && (utf8[1] & 0xF0) == 0x80) /* overlong sequence check */
return 0;
return 4;
}
else if ((utf8[0] & 0xFC) == 0xF8 && (utf8[1] & 0xC0) == 0x80 && (utf8[2] & 0xC0) == 0x80 && (utf8[3] & 0xC0) == 0x80 && (utf8[4] & 0xC0) == 0x80) {
if (utf8[0] == 0xF8 && (utf8[1] & 0xF8) == 0x80) /* overlong sequence check */
return 0;
return 5;
}
else if ((utf8[0] & 0xFE) == 0xFC && (utf8[1] & 0xC0) == 0x80 && (utf8[2] & 0xC0) == 0x80 && (utf8[3] & 0xC0) == 0x80 && (utf8[4] & 0xC0) == 0x80 && (utf8[5] & 0xC0) == 0x80) {
if (utf8[0] == 0xFC && (utf8[1] & 0xFC) == 0x80) /* overlong sequence check */
return 0;
return 6;
}
else {
return 0;
}
}
FLAC_API FLAC__bool FLAC__format_vorbiscomment_entry_name_is_legal(const char *name)
{
char c;
for(c = *name; c; c = *(++name))
if(c < 0x20 || c == 0x3d || c > 0x7d)
return false;
return true;
}
FLAC_API FLAC__bool FLAC__format_vorbiscomment_entry_value_is_legal(const FLAC__byte *value, unsigned length)
{
if(length == (unsigned)(-1)) {
while(*value) {
unsigned n = utf8len_(value);
if(n == 0)
return false;
value += n;
}
}
else {
const FLAC__byte *end = value + length;
while(value < end) {
unsigned n = utf8len_(value);
if(n == 0)
return false;
value += n;
}
if(value != end)
return false;
}
return true;
}
FLAC_API FLAC__bool FLAC__format_vorbiscomment_entry_is_legal(const FLAC__byte *entry, unsigned length)
{
const FLAC__byte *s, *end;
for(s = entry, end = s + length; s < end && *s != '='; s++) {
if(*s < 0x20 || *s > 0x7D)
return false;
}
if(s == end)
return false;
s++; /* skip '=' */
while(s < end) {
unsigned n = utf8len_(s);
if(n == 0)
return false;
s += n;
}
if(s != end)
return false;
return true;
}
/* @@@@ add to unit tests; it is already indirectly tested by the metadata_object tests */
FLAC_API FLAC__bool FLAC__format_cuesheet_is_legal(const FLAC__StreamMetadata_CueSheet *cue_sheet, FLAC__bool check_cd_da_subset, const char **violation)
{
unsigned i, j;
if(check_cd_da_subset) {
if(cue_sheet->lead_in < 2 * 44100) {
if(violation) *violation = "CD-DA cue sheet must have a lead-in length of at least 2 seconds";
return false;
}
if(cue_sheet->lead_in % 588 != 0) {
if(violation) *violation = "CD-DA cue sheet lead-in length must be evenly divisible by 588 samples";
return false;
}
}
if(cue_sheet->num_tracks == 0) {
if(violation) *violation = "cue sheet must have at least one track (the lead-out)";
return false;
}
if(check_cd_da_subset && cue_sheet->tracks[cue_sheet->num_tracks-1].number != 170) {
if(violation) *violation = "CD-DA cue sheet must have a lead-out track number 170 (0xAA)";
return false;
}
for(i = 0; i < cue_sheet->num_tracks; i++) {
if(cue_sheet->tracks[i].number == 0) {
if(violation) *violation = "cue sheet may not have a track number 0";
return false;
}
if(check_cd_da_subset) {
if(!((cue_sheet->tracks[i].number >= 1 && cue_sheet->tracks[i].number <= 99) || cue_sheet->tracks[i].number == 170)) {
if(violation) *violation = "CD-DA cue sheet track number must be 1-99 or 170";
return false;
}
}
if(check_cd_da_subset && cue_sheet->tracks[i].offset % 588 != 0) {
if(violation) {
if(i == cue_sheet->num_tracks-1) /* the lead-out track... */
*violation = "CD-DA cue sheet lead-out offset must be evenly divisible by 588 samples";
else
*violation = "CD-DA cue sheet track offset must be evenly divisible by 588 samples";
}
return false;
}
if(i < cue_sheet->num_tracks - 1) {
if(cue_sheet->tracks[i].num_indices == 0) {
if(violation) *violation = "cue sheet track must have at least one index point";
return false;
}
if(cue_sheet->tracks[i].indices[0].number > 1) {
if(violation) *violation = "cue sheet track's first index number must be 0 or 1";
return false;
}
}
for(j = 0; j < cue_sheet->tracks[i].num_indices; j++) {
if(check_cd_da_subset && cue_sheet->tracks[i].indices[j].offset % 588 != 0) {
if(violation) *violation = "CD-DA cue sheet track index offset must be evenly divisible by 588 samples";
return false;
}
if(j > 0) {
if(cue_sheet->tracks[i].indices[j].number != cue_sheet->tracks[i].indices[j-1].number + 1) {
if(violation) *violation = "cue sheet track index numbers must increase by 1";
return false;
}
}
}
}
return true;
}
/* @@@@ add to unit tests; it is already indirectly tested by the metadata_object tests */
FLAC_API FLAC__bool FLAC__format_picture_is_legal(const FLAC__StreamMetadata_Picture *picture, const char **violation)
{
char *p;
FLAC__byte *b;
for(p = picture->mime_type; *p; p++) {
if(*p < 0x20 || *p > 0x7e) {
if(violation) *violation = "MIME type string must contain only printable ASCII characters (0x20-0x7e)";
return false;
}
}
for(b = picture->description; *b; ) {
unsigned n = utf8len_(b);
if(n == 0) {
if(violation) *violation = "description string must be valid UTF-8";
return false;
}
b += n;
}
return true;
}
/*
* These routines are private to libFLAC
*/
unsigned FLAC__format_get_max_rice_partition_order(unsigned blocksize, unsigned predictor_order)
{
return
FLAC__format_get_max_rice_partition_order_from_blocksize_limited_max_and_predictor_order(
FLAC__format_get_max_rice_partition_order_from_blocksize(blocksize),
blocksize,
predictor_order
);
}
unsigned FLAC__format_get_max_rice_partition_order_from_blocksize(unsigned blocksize)
{
unsigned max_rice_partition_order = 0;
while(!(blocksize & 1)) {
max_rice_partition_order++;
blocksize >>= 1;
}
return flac_min(FLAC__MAX_RICE_PARTITION_ORDER, max_rice_partition_order);
}
unsigned FLAC__format_get_max_rice_partition_order_from_blocksize_limited_max_and_predictor_order(unsigned limit, unsigned blocksize, unsigned predictor_order)
{
unsigned max_rice_partition_order = limit;
while(max_rice_partition_order > 0 && (blocksize >> max_rice_partition_order) <= predictor_order)
max_rice_partition_order--;
FLAC__ASSERT(
(max_rice_partition_order == 0 && blocksize >= predictor_order) ||
(max_rice_partition_order > 0 && blocksize >> max_rice_partition_order > predictor_order)
);
return max_rice_partition_order;
}
void FLAC__format_entropy_coding_method_partitioned_rice_contents_init(FLAC__EntropyCodingMethod_PartitionedRiceContents *object)
{
FLAC__ASSERT(0 != object);
object->parameters = 0;
object->raw_bits = 0;
object->capacity_by_order = 0;
}
void FLAC__format_entropy_coding_method_partitioned_rice_contents_clear(FLAC__EntropyCodingMethod_PartitionedRiceContents *object)
{
FLAC__ASSERT(0 != object);
if(0 != object->parameters)
free(object->parameters);
if(0 != object->raw_bits)
free(object->raw_bits);
FLAC__format_entropy_coding_method_partitioned_rice_contents_init(object);
}
FLAC__bool FLAC__format_entropy_coding_method_partitioned_rice_contents_ensure_size(FLAC__EntropyCodingMethod_PartitionedRiceContents *object, unsigned max_partition_order)
{
FLAC__ASSERT(0 != object);
FLAC__ASSERT(object->capacity_by_order > 0 || (0 == object->parameters && 0 == object->raw_bits));
if(object->capacity_by_order < max_partition_order) {
if(0 == (object->parameters = (unsigned int*) realloc(object->parameters, sizeof(unsigned)*(1 << max_partition_order))))
return false;
if(0 == (object->raw_bits = (unsigned int*) realloc(object->raw_bits, sizeof(unsigned)*(1 << max_partition_order))))
return false;
memset(object->raw_bits, 0, sizeof(unsigned)*(1 << max_partition_order));
object->capacity_by_order = max_partition_order;
}
return true;
}
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/all.h
+50
View File
@@ -0,0 +1,50 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__ALL_H
#define FLAC__PRIVATE__ALL_H
#include "bitmath.h"
#include "bitreader.h"
#include "bitwriter.h"
#include "cpu.h"
#include "crc.h"
#include "fixed.h"
#include "float.h"
#include "format.h"
#include "lpc.h"
#include "md5.h"
#include "memory.h"
#include "metadata.h"
#include "stream_encoder_framing.h"
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/bitmath.h
+186
View File
@@ -0,0 +1,186 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2001-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__BITMATH_H
#define FLAC__PRIVATE__BITMATH_H
#include "../../../ordinals.h"
#include "../../../assert.h"
/* for CHAR_BIT */
#include <limits.h>
#include "../../../compat.h"
#if defined(_MSC_VER) && (_MSC_VER >= 1400)
#include <intrin.h> /* for _BitScanReverse* */
#endif
/* Will never be emitted for MSVC, GCC, Intel compilers */
static inline unsigned int FLAC__clz_soft_uint32(unsigned int word)
{
static const unsigned char byte_to_unary_table[] = {
8, 7, 6, 6, 5, 5, 5, 5, 4, 4, 4, 4, 4, 4, 4, 4,
3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3,
2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2,
1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
};
return (word) > 0xffffff ? byte_to_unary_table[(word) >> 24] :
(word) > 0xffff ? byte_to_unary_table[(word) >> 16] + 8 :
(word) > 0xff ? byte_to_unary_table[(word) >> 8] + 16 :
byte_to_unary_table[(word)] + 24;
}
static inline unsigned int FLAC__clz_uint32(FLAC__uint32 v)
{
/* Never used with input 0 */
FLAC__ASSERT(v > 0);
#if defined(__INTEL_COMPILER)
return _bit_scan_reverse(v) ^ 31U;
#elif defined(__GNUC__) && (__GNUC__ >= 4 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 4))
/* This will translate either to (bsr ^ 31U), clz , ctlz, cntlz, lzcnt depending on
* -march= setting or to a software routine in exotic machines. */
return __builtin_clz(v);
#elif defined(_MSC_VER) && (_MSC_VER >= 1400)
{
unsigned long idx;
_BitScanReverse(&idx, v);
return idx ^ 31U;
}
#else
return FLAC__clz_soft_uint32(v);
#endif
}
/* This one works with input 0 */
static inline unsigned int FLAC__clz2_uint32(FLAC__uint32 v)
{
if (!v)
return 32;
return FLAC__clz_uint32(v);
}
/* An example of what FLAC__bitmath_ilog2() computes:
*
* ilog2( 0) = assertion failure
* ilog2( 1) = 0
* ilog2( 2) = 1
* ilog2( 3) = 1
* ilog2( 4) = 2
* ilog2( 5) = 2
* ilog2( 6) = 2
* ilog2( 7) = 2
* ilog2( 8) = 3
* ilog2( 9) = 3
* ilog2(10) = 3
* ilog2(11) = 3
* ilog2(12) = 3
* ilog2(13) = 3
* ilog2(14) = 3
* ilog2(15) = 3
* ilog2(16) = 4
* ilog2(17) = 4
* ilog2(18) = 4
*/
static inline unsigned FLAC__bitmath_ilog2(FLAC__uint32 v)
{
FLAC__ASSERT(v > 0);
#if defined(__INTEL_COMPILER)
return _bit_scan_reverse(v);
#elif defined(_MSC_VER) && (_MSC_VER >= 1400)
{
unsigned long idx;
_BitScanReverse(&idx, v);
return idx;
}
#else
return sizeof(FLAC__uint32) * CHAR_BIT - 1 - FLAC__clz_uint32(v);
#endif
}
#ifdef FLAC__INTEGER_ONLY_LIBRARY /* Unused otherwise */
static inline unsigned FLAC__bitmath_ilog2_wide(FLAC__uint64 v)
{
FLAC__ASSERT(v > 0);
#if defined(__GNUC__) && (__GNUC__ >= 4 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 4))
return sizeof(FLAC__uint64) * CHAR_BIT - 1 - __builtin_clzll(v);
/* Sorry, only supported in x64/Itanium.. and both have fast FPU which makes integer-only encoder pointless */
#elif (defined(_MSC_VER) && (_MSC_VER >= 1400)) && (defined(_M_IA64) || defined(_M_X64))
{
unsigned long idx;
_BitScanReverse64(&idx, v);
return idx;
}
#else
/* Brain-damaged compilers will use the fastest possible way that is,
de Bruijn sequences (http://supertech.csail.mit.edu/papers/debruijn.pdf)
(C) Timothy B. Terriberry (tterribe@xiph.org) 2001-2009 CC0 (Public domain).
*/
{
static const unsigned char DEBRUIJN_IDX64[64]={
0, 1, 2, 7, 3,13, 8,19, 4,25,14,28, 9,34,20,40,
5,17,26,38,15,46,29,48,10,31,35,54,21,50,41,57,
63, 6,12,18,24,27,33,39,16,37,45,47,30,53,49,56,
62,11,23,32,36,44,52,55,61,22,43,51,60,42,59,58
};
v|= v>>1;
v|= v>>2;
v|= v>>4;
v|= v>>8;
v|= v>>16;
v|= v>>32;
v= (v>>1)+1;
return DEBRUIJN_IDX64[v*0x218A392CD3D5DBF>>58&0x3F];
}
#endif
}
#endif
unsigned FLAC__bitmath_silog2(int v);
unsigned FLAC__bitmath_silog2_wide(FLAC__int64 v);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/bitreader.h
+91
View File
@@ -0,0 +1,91 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__BITREADER_H
#define FLAC__PRIVATE__BITREADER_H
#include <stdio.h> /* for FILE */
#include "../../../ordinals.h"
#include "cpu.h"
/*
* opaque structure definition
*/
struct FLAC__BitReader;
typedef struct FLAC__BitReader FLAC__BitReader;
typedef FLAC__bool (*FLAC__BitReaderReadCallback)(FLAC__byte buffer[], size_t *bytes, void *client_data);
/*
* construction, deletion, initialization, etc functions
*/
FLAC__BitReader *FLAC__bitreader_new(void);
void FLAC__bitreader_delete(FLAC__BitReader *br);
FLAC__bool FLAC__bitreader_init(FLAC__BitReader *br, FLAC__BitReaderReadCallback rcb, void *cd);
void FLAC__bitreader_free(FLAC__BitReader *br); /* does not 'free(br)' */
FLAC__bool FLAC__bitreader_clear(FLAC__BitReader *br);
void FLAC__bitreader_dump(const FLAC__BitReader *br, FILE *out);
/*
* CRC functions
*/
void FLAC__bitreader_reset_read_crc16(FLAC__BitReader *br, FLAC__uint16 seed);
FLAC__uint16 FLAC__bitreader_get_read_crc16(FLAC__BitReader *br);
/*
* info functions
*/
FLAC__bool FLAC__bitreader_is_consumed_byte_aligned(const FLAC__BitReader *br);
unsigned FLAC__bitreader_bits_left_for_byte_alignment(const FLAC__BitReader *br);
unsigned FLAC__bitreader_get_input_bits_unconsumed(const FLAC__BitReader *br);
/*
* read functions
*/
FLAC__bool FLAC__bitreader_read_raw_uint32(FLAC__BitReader *br, FLAC__uint32 *val, unsigned bits);
FLAC__bool FLAC__bitreader_read_raw_int32(FLAC__BitReader *br, FLAC__int32 *val, unsigned bits);
FLAC__bool FLAC__bitreader_read_raw_uint64(FLAC__BitReader *br, FLAC__uint64 *val, unsigned bits);
FLAC__bool FLAC__bitreader_read_uint32_little_endian(FLAC__BitReader *br, FLAC__uint32 *val); /*only for bits=32*/
FLAC__bool FLAC__bitreader_skip_bits_no_crc(FLAC__BitReader *br, unsigned bits); /* WATCHOUT: does not CRC the skipped data! */ /*@@@@ add to unit tests */
FLAC__bool FLAC__bitreader_skip_byte_block_aligned_no_crc(FLAC__BitReader *br, unsigned nvals); /* WATCHOUT: does not CRC the read data! */
FLAC__bool FLAC__bitreader_read_byte_block_aligned_no_crc(FLAC__BitReader *br, FLAC__byte *val, unsigned nvals); /* WATCHOUT: does not CRC the read data! */
FLAC__bool FLAC__bitreader_read_unary_unsigned(FLAC__BitReader *br, unsigned *val);
FLAC__bool FLAC__bitreader_read_rice_signed(FLAC__BitReader *br, int *val, unsigned parameter);
FLAC__bool FLAC__bitreader_read_rice_signed_block(FLAC__BitReader *br, int vals[], unsigned nvals, unsigned parameter);
#if 0 /* UNUSED */
FLAC__bool FLAC__bitreader_read_golomb_signed(FLAC__BitReader *br, int *val, unsigned parameter);
FLAC__bool FLAC__bitreader_read_golomb_unsigned(FLAC__BitReader *br, unsigned *val, unsigned parameter);
#endif
FLAC__bool FLAC__bitreader_read_utf8_uint32(FLAC__BitReader *br, FLAC__uint32 *val, FLAC__byte *raw, unsigned *rawlen);
FLAC__bool FLAC__bitreader_read_utf8_uint64(FLAC__BitReader *br, FLAC__uint64 *val, FLAC__byte *raw, unsigned *rawlen);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/bitwriter.h
+104
View File
@@ -0,0 +1,104 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__BITWRITER_H
#define FLAC__PRIVATE__BITWRITER_H
#include <stdio.h> /* for FILE */
#include "../../../ordinals.h"
/*
* opaque structure definition
*/
struct FLAC__BitWriter;
typedef struct FLAC__BitWriter FLAC__BitWriter;
/*
* construction, deletion, initialization, etc functions
*/
FLAC__BitWriter *FLAC__bitwriter_new(void);
void FLAC__bitwriter_delete(FLAC__BitWriter *bw);
FLAC__bool FLAC__bitwriter_init(FLAC__BitWriter *bw);
void FLAC__bitwriter_free(FLAC__BitWriter *bw); /* does not 'free(buffer)' */
void FLAC__bitwriter_clear(FLAC__BitWriter *bw);
void FLAC__bitwriter_dump(const FLAC__BitWriter *bw, FILE *out);
/*
* CRC functions
*
* non-const *bw because they have to cal FLAC__bitwriter_get_buffer()
*/
FLAC__bool FLAC__bitwriter_get_write_crc16(FLAC__BitWriter *bw, FLAC__uint16 *crc);
FLAC__bool FLAC__bitwriter_get_write_crc8(FLAC__BitWriter *bw, FLAC__byte *crc);
/*
* info functions
*/
FLAC__bool FLAC__bitwriter_is_byte_aligned(const FLAC__BitWriter *bw);
unsigned FLAC__bitwriter_get_input_bits_unconsumed(const FLAC__BitWriter *bw); /* can be called anytime, returns total # of bits unconsumed */
/*
* direct buffer access
*
* there may be no calls on the bitwriter between get and release.
* the bitwriter continues to own the returned buffer.
* before get, bitwriter MUST be byte aligned: check with FLAC__bitwriter_is_byte_aligned()
*/
FLAC__bool FLAC__bitwriter_get_buffer(FLAC__BitWriter *bw, const FLAC__byte **buffer, size_t *bytes);
void FLAC__bitwriter_release_buffer(FLAC__BitWriter *bw);
/*
* write functions
*/
FLAC__bool FLAC__bitwriter_write_zeroes(FLAC__BitWriter *bw, unsigned bits);
FLAC__bool FLAC__bitwriter_write_raw_uint32(FLAC__BitWriter *bw, FLAC__uint32 val, unsigned bits);
FLAC__bool FLAC__bitwriter_write_raw_int32(FLAC__BitWriter *bw, FLAC__int32 val, unsigned bits);
FLAC__bool FLAC__bitwriter_write_raw_uint64(FLAC__BitWriter *bw, FLAC__uint64 val, unsigned bits);
FLAC__bool FLAC__bitwriter_write_raw_uint32_little_endian(FLAC__BitWriter *bw, FLAC__uint32 val); /*only for bits=32*/
FLAC__bool FLAC__bitwriter_write_byte_block(FLAC__BitWriter *bw, const FLAC__byte vals[], unsigned nvals);
FLAC__bool FLAC__bitwriter_write_unary_unsigned(FLAC__BitWriter *bw, unsigned val);
unsigned FLAC__bitwriter_rice_bits(FLAC__int32 val, unsigned parameter);
#if 0 /* UNUSED */
unsigned FLAC__bitwriter_golomb_bits_signed(int val, unsigned parameter);
unsigned FLAC__bitwriter_golomb_bits_unsigned(unsigned val, unsigned parameter);
#endif
FLAC__bool FLAC__bitwriter_write_rice_signed(FLAC__BitWriter *bw, FLAC__int32 val, unsigned parameter);
FLAC__bool FLAC__bitwriter_write_rice_signed_block(FLAC__BitWriter *bw, const FLAC__int32 *vals, unsigned nvals, unsigned parameter);
#if 0 /* UNUSED */
FLAC__bool FLAC__bitwriter_write_golomb_signed(FLAC__BitWriter *bw, int val, unsigned parameter);
FLAC__bool FLAC__bitwriter_write_golomb_unsigned(FLAC__BitWriter *bw, unsigned val, unsigned parameter);
#endif
FLAC__bool FLAC__bitwriter_write_utf8_uint32(FLAC__BitWriter *bw, FLAC__uint32 val);
FLAC__bool FLAC__bitwriter_write_utf8_uint64(FLAC__BitWriter *bw, FLAC__uint64 val);
FLAC__bool FLAC__bitwriter_zero_pad_to_byte_boundary(FLAC__BitWriter *bw);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/cpu.h
+99
View File
@@ -0,0 +1,99 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2001-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__CPU_H
#define FLAC__PRIVATE__CPU_H
#include "../../../ordinals.h"
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
typedef enum {
FLAC__CPUINFO_TYPE_IA32,
FLAC__CPUINFO_TYPE_X86_64,
FLAC__CPUINFO_TYPE_UNKNOWN
} FLAC__CPUInfo_Type;
#if defined FLAC__CPU_IA32
typedef struct {
FLAC__bool cmov;
FLAC__bool mmx;
FLAC__bool sse;
FLAC__bool sse2;
FLAC__bool sse3;
FLAC__bool ssse3;
FLAC__bool sse41;
FLAC__bool sse42;
FLAC__bool avx;
FLAC__bool avx2;
FLAC__bool fma;
} FLAC__CPUInfo_IA32;
#elif defined FLAC__CPU_X86_64
typedef struct {
FLAC__bool sse3;
FLAC__bool ssse3;
FLAC__bool sse41;
FLAC__bool sse42;
FLAC__bool avx;
FLAC__bool avx2;
FLAC__bool fma;
} FLAC__CPUInfo_x86;
#endif
typedef struct {
FLAC__bool use_asm;
FLAC__CPUInfo_Type type;
#if defined FLAC__CPU_IA32
FLAC__CPUInfo_IA32 ia32;
#elif defined FLAC__CPU_X86_64
FLAC__CPUInfo_x86 x86;
#endif
} FLAC__CPUInfo;
void FLAC__cpu_info(FLAC__CPUInfo *info);
#ifndef FLAC__NO_ASM
# if defined FLAC__CPU_IA32 && defined FLAC__HAS_NASM
FLAC__uint32 FLAC__cpu_have_cpuid_asm_ia32(void);
void FLAC__cpu_info_asm_ia32(FLAC__uint32 *flags_edx, FLAC__uint32 *flags_ecx);
# endif
# if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
FLAC__uint32 FLAC__cpu_have_cpuid_x86(void);
void FLAC__cpu_info_x86(FLAC__uint32 level, FLAC__uint32 *eax, FLAC__uint32 *ebx, FLAC__uint32 *ecx, FLAC__uint32 *edx);
FLAC__uint32 FLAC__cpu_xgetbv_x86(void);
# endif
#endif
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/crc.h
+62
View File
@@ -0,0 +1,62 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__CRC_H
#define FLAC__PRIVATE__CRC_H
#include "../../../ordinals.h"
/* 8 bit CRC generator, MSB shifted first
** polynomial = x^8 + x^2 + x^1 + x^0
** init = 0
*/
extern FLAC__byte const FLAC__crc8_table[256];
#define FLAC__CRC8_UPDATE(data, crc) (crc) = FLAC__crc8_table[(crc) ^ (data)];
void FLAC__crc8_update(const FLAC__byte data, FLAC__uint8 *crc);
void FLAC__crc8_update_block(const FLAC__byte *data, unsigned len, FLAC__uint8 *crc);
FLAC__uint8 FLAC__crc8(const FLAC__byte *data, unsigned len);
/* 16 bit CRC generator, MSB shifted first
** polynomial = x^16 + x^15 + x^2 + x^0
** init = 0
*/
extern unsigned const FLAC__crc16_table[256];
#define FLAC__CRC16_UPDATE(data, crc) ((((crc)<<8) & 0xffff) ^ FLAC__crc16_table[((crc)>>8) ^ (data)])
/* this alternate may be faster on some systems/compilers */
#if 0
#define FLAC__CRC16_UPDATE(data, crc) ((((crc)<<8) ^ FLAC__crc16_table[((crc)>>8) ^ (data)]) & 0xffff)
#endif
unsigned FLAC__crc16(const FLAC__byte *data, unsigned len);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/fixed.h
+107
View File
@@ -0,0 +1,107 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__FIXED_H
#define FLAC__PRIVATE__FIXED_H
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include "cpu.h"
#include "float.h"
#include "../../../format.h"
/*
* FLAC__fixed_compute_best_predictor()
* --------------------------------------------------------------------
* Compute the best fixed predictor and the expected bits-per-sample
* of the residual signal for each order. The _wide() version uses
* 64-bit integers which is statistically necessary when bits-per-
* sample + log2(blocksize) > 30
*
* IN data[0,data_len-1]
* IN data_len
* OUT residual_bits_per_sample[0,FLAC__MAX_FIXED_ORDER]
*/
#ifndef FLAC__INTEGER_ONLY_LIBRARY
unsigned FLAC__fixed_compute_best_predictor(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
unsigned FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
# ifndef FLAC__NO_ASM
# if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
# ifdef FLAC__SSE2_SUPPORTED
unsigned FLAC__fixed_compute_best_predictor_intrin_sse2(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER + 1]);
unsigned FLAC__fixed_compute_best_predictor_wide_intrin_sse2(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER + 1]);
# endif
# ifdef FLAC__SSSE3_SUPPORTED
unsigned FLAC__fixed_compute_best_predictor_intrin_ssse3(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
unsigned FLAC__fixed_compute_best_predictor_wide_intrin_ssse3(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER + 1]);
# endif
# endif
# if defined FLAC__CPU_IA32 && defined FLAC__HAS_NASM
unsigned FLAC__fixed_compute_best_predictor_asm_ia32_mmx_cmov(const FLAC__int32 data[], unsigned data_len, FLAC__float residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
# endif
# endif
#else
unsigned FLAC__fixed_compute_best_predictor(const FLAC__int32 data[], unsigned data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
unsigned FLAC__fixed_compute_best_predictor_wide(const FLAC__int32 data[], unsigned data_len, FLAC__fixedpoint residual_bits_per_sample[FLAC__MAX_FIXED_ORDER+1]);
#endif
/*
* FLAC__fixed_compute_residual()
* --------------------------------------------------------------------
* Compute the residual signal obtained from sutracting the predicted
* signal from the original.
*
* IN data[-order,data_len-1] original signal (NOTE THE INDICES!)
* IN data_len length of original signal
* IN order <= FLAC__MAX_FIXED_ORDER fixed-predictor order
* OUT residual[0,data_len-1] residual signal
*/
void FLAC__fixed_compute_residual(const FLAC__int32 data[], unsigned data_len, unsigned order, FLAC__int32 residual[]);
/*
* FLAC__fixed_restore_signal()
* --------------------------------------------------------------------
* Restore the original signal by summing the residual and the
* predictor.
*
* IN residual[0,data_len-1] residual signal
* IN data_len length of original signal
* IN order <= FLAC__MAX_FIXED_ORDER fixed-predictor order
* *** IMPORTANT: the caller must pass in the historical samples:
* IN data[-order,-1] previously-reconstructed historical samples
* OUT data[0,data_len-1] original signal
*/
void FLAC__fixed_restore_signal(const FLAC__int32 residual[], unsigned data_len, unsigned order, FLAC__int32 data[]);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/float.h
+98
View File
@@ -0,0 +1,98 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2004-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__FLOAT_H
#define FLAC__PRIVATE__FLOAT_H
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include "../../../ordinals.h"
/*
* These typedefs make it easier to ensure that integer versions of
* the library really only contain integer operations. All the code
* in libFLAC should use FLAC__float and FLAC__double in place of
* float and double, and be protected by checks of the macro
* FLAC__INTEGER_ONLY_LIBRARY.
*
* FLAC__real is the basic floating point type used in LPC analysis.
*/
#ifndef FLAC__INTEGER_ONLY_LIBRARY
typedef double FLAC__double;
typedef float FLAC__float;
/*
* WATCHOUT: changing FLAC__real will change the signatures of many
* functions that have assembly language equivalents and break them.
*/
typedef float FLAC__real;
#else
/*
* The convention for FLAC__fixedpoint is to use the upper 16 bits
* for the integer part and lower 16 bits for the fractional part.
*/
typedef FLAC__int32 FLAC__fixedpoint;
extern const FLAC__fixedpoint FLAC__FP_ZERO;
extern const FLAC__fixedpoint FLAC__FP_ONE_HALF;
extern const FLAC__fixedpoint FLAC__FP_ONE;
extern const FLAC__fixedpoint FLAC__FP_LN2;
extern const FLAC__fixedpoint FLAC__FP_E;
#define FLAC__fixedpoint_trunc(x) ((x)>>16)
#define FLAC__fixedpoint_mul(x, y) ( (FLAC__fixedpoint) ( ((FLAC__int64)(x)*(FLAC__int64)(y)) >> 16 ) )
#define FLAC__fixedpoint_div(x, y) ( (FLAC__fixedpoint) ( ( ((FLAC__int64)(x)<<32) / (FLAC__int64)(y) ) >> 16 ) )
/*
* FLAC__fixedpoint_log2()
* --------------------------------------------------------------------
* Returns the base-2 logarithm of the fixed-point number 'x' using an
* algorithm by Knuth for x >= 1.0
*
* 'fracbits' is the number of fractional bits of 'x'. 'fracbits' must
* be < 32 and evenly divisible by 4 (0 is OK but not very precise).
*
* 'precision' roughly limits the number of iterations that are done;
* use (unsigned)(-1) for maximum precision.
*
* If 'x' is less than one -- that is, x < (1<<fracbits) -- then this
* function will punt and return 0.
*
* The return value will also have 'fracbits' fractional bits.
*/
FLAC__uint32 FLAC__fixedpoint_log2(FLAC__uint32 x, unsigned fracbits, unsigned precision);
#endif
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/format.h
+45
View File
@@ -0,0 +1,45 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__FORMAT_H
#define FLAC__PRIVATE__FORMAT_H
#include "../../../format.h"
unsigned FLAC__format_get_max_rice_partition_order(unsigned blocksize, unsigned predictor_order);
unsigned FLAC__format_get_max_rice_partition_order_from_blocksize(unsigned blocksize);
unsigned FLAC__format_get_max_rice_partition_order_from_blocksize_limited_max_and_predictor_order(unsigned limit, unsigned blocksize, unsigned predictor_order);
void FLAC__format_entropy_coding_method_partitioned_rice_contents_init(FLAC__EntropyCodingMethod_PartitionedRiceContents *object);
void FLAC__format_entropy_coding_method_partitioned_rice_contents_clear(FLAC__EntropyCodingMethod_PartitionedRiceContents *object);
FLAC__bool FLAC__format_entropy_coding_method_partitioned_rice_contents_ensure_size(FLAC__EntropyCodingMethod_PartitionedRiceContents *object, unsigned max_partition_order);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/lpc.h
+246
View File
@@ -0,0 +1,246 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__LPC_H
#define FLAC__PRIVATE__LPC_H
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include "cpu.h"
#include "float.h"
#include "../../../format.h"
#ifndef FLAC__INTEGER_ONLY_LIBRARY
/*
* FLAC__lpc_window_data()
* --------------------------------------------------------------------
* Applies the given window to the data.
* OPT: asm implementation
*
* IN in[0,data_len-1]
* IN window[0,data_len-1]
* OUT out[0,lag-1]
* IN data_len
*/
void FLAC__lpc_window_data(const FLAC__int32 in[], const FLAC__real window[], FLAC__real out[], unsigned data_len);
/*
* FLAC__lpc_compute_autocorrelation()
* --------------------------------------------------------------------
* Compute the autocorrelation for lags between 0 and lag-1.
* Assumes data[] outside of [0,data_len-1] == 0.
* Asserts that lag > 0.
*
* IN data[0,data_len-1]
* IN data_len
* IN 0 < lag <= data_len
* OUT autoc[0,lag-1]
*/
void FLAC__lpc_compute_autocorrelation(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
#ifndef FLAC__NO_ASM
# ifdef FLAC__CPU_IA32
# ifdef FLAC__HAS_NASM
void FLAC__lpc_compute_autocorrelation_asm_ia32(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
void FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_4(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
void FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_8(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
void FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_12(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
void FLAC__lpc_compute_autocorrelation_asm_ia32_sse_lag_16(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
# endif
# endif
# if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
# ifdef FLAC__SSE_SUPPORTED
void FLAC__lpc_compute_autocorrelation_intrin_sse_lag_4(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
void FLAC__lpc_compute_autocorrelation_intrin_sse_lag_8(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
void FLAC__lpc_compute_autocorrelation_intrin_sse_lag_12(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
void FLAC__lpc_compute_autocorrelation_intrin_sse_lag_16(const FLAC__real data[], unsigned data_len, unsigned lag, FLAC__real autoc[]);
# endif
# endif
#endif
/*
* FLAC__lpc_compute_lp_coefficients()
* --------------------------------------------------------------------
* Computes LP coefficients for orders 1..max_order.
* Do not call if autoc[0] == 0.0. This means the signal is zero
* and there is no point in calculating a predictor.
*
* IN autoc[0,max_order] autocorrelation values
* IN 0 < max_order <= FLAC__MAX_LPC_ORDER max LP order to compute
* OUT lp_coeff[0,max_order-1][0,max_order-1] LP coefficients for each order
* *** IMPORTANT:
* *** lp_coeff[0,max_order-1][max_order,FLAC__MAX_LPC_ORDER-1] are untouched
* OUT error[0,max_order-1] error for each order (more
* specifically, the variance of
* the error signal times # of
* samples in the signal)
*
* Example: if max_order is 9, the LP coefficients for order 9 will be
* in lp_coeff[8][0,8], the LP coefficients for order 8 will be
* in lp_coeff[7][0,7], etc.
*/
void FLAC__lpc_compute_lp_coefficients(const FLAC__real autoc[], unsigned *max_order, FLAC__real lp_coeff[][FLAC__MAX_LPC_ORDER], FLAC__double error[]);
/*
* FLAC__lpc_quantize_coefficients()
* --------------------------------------------------------------------
* Quantizes the LP coefficients. NOTE: precision + bits_per_sample
* must be less than 32 (sizeof(FLAC__int32)*8).
*
* IN lp_coeff[0,order-1] LP coefficients
* IN order LP order
* IN FLAC__MIN_QLP_COEFF_PRECISION < precision
* desired precision (in bits, including sign
* bit) of largest coefficient
* OUT qlp_coeff[0,order-1] quantized coefficients
* OUT shift # of bits to shift right to get approximated
* LP coefficients. NOTE: could be negative.
* RETURN 0 => quantization OK
* 1 => coefficients require too much shifting for *shift to
* fit in the LPC subframe header. 'shift' is unset.
* 2 => coefficients are all zero, which is bad. 'shift' is
* unset.
*/
int FLAC__lpc_quantize_coefficients(const FLAC__real lp_coeff[], unsigned order, unsigned precision, FLAC__int32 qlp_coeff[], int *shift);
/*
* FLAC__lpc_compute_residual_from_qlp_coefficients()
* --------------------------------------------------------------------
* Compute the residual signal obtained from sutracting the predicted
* signal from the original.
*
* IN data[-order,data_len-1] original signal (NOTE THE INDICES!)
* IN data_len length of original signal
* IN qlp_coeff[0,order-1] quantized LP coefficients
* IN order > 0 LP order
* IN lp_quantization quantization of LP coefficients in bits
* OUT residual[0,data_len-1] residual signal
*/
void FLAC__lpc_compute_residual_from_qlp_coefficients(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
void FLAC__lpc_compute_residual_from_qlp_coefficients_wide(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
#ifndef FLAC__NO_ASM
# ifdef FLAC__CPU_IA32
# ifdef FLAC__HAS_NASM
void FLAC__lpc_compute_residual_from_qlp_coefficients_asm_ia32(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
void FLAC__lpc_compute_residual_from_qlp_coefficients_asm_ia32_mmx(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
void FLAC__lpc_compute_residual_from_qlp_coefficients_wide_asm_ia32(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
# endif
# endif
# if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
# ifdef FLAC__SSE2_SUPPORTED
void FLAC__lpc_compute_residual_from_qlp_coefficients_16_intrin_sse2(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
void FLAC__lpc_compute_residual_from_qlp_coefficients_intrin_sse2(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
# endif
# ifdef FLAC__SSE4_1_SUPPORTED
void FLAC__lpc_compute_residual_from_qlp_coefficients_intrin_sse41(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
void FLAC__lpc_compute_residual_from_qlp_coefficients_wide_intrin_sse41(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
# endif
# ifdef FLAC__AVX2_SUPPORTED
void FLAC__lpc_compute_residual_from_qlp_coefficients_16_intrin_avx2(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
void FLAC__lpc_compute_residual_from_qlp_coefficients_intrin_avx2(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
void FLAC__lpc_compute_residual_from_qlp_coefficients_wide_intrin_avx2(const FLAC__int32 *data, unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 residual[]);
# endif
# endif
#endif
#endif /* !defined FLAC__INTEGER_ONLY_LIBRARY */
/*
* FLAC__lpc_restore_signal()
* --------------------------------------------------------------------
* Restore the original signal by summing the residual and the
* predictor.
*
* IN residual[0,data_len-1] residual signal
* IN data_len length of original signal
* IN qlp_coeff[0,order-1] quantized LP coefficients
* IN order > 0 LP order
* IN lp_quantization quantization of LP coefficients in bits
* *** IMPORTANT: the caller must pass in the historical samples:
* IN data[-order,-1] previously-reconstructed historical samples
* OUT data[0,data_len-1] original signal
*/
void FLAC__lpc_restore_signal(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
void FLAC__lpc_restore_signal_wide(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
#ifndef FLAC__NO_ASM
# ifdef FLAC__CPU_IA32
# ifdef FLAC__HAS_NASM
void FLAC__lpc_restore_signal_asm_ia32(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
void FLAC__lpc_restore_signal_asm_ia32_mmx(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
void FLAC__lpc_restore_signal_wide_asm_ia32(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
# endif /* FLAC__HAS_NASM */
# endif /* FLAC__CPU_IA32 */
# if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
# ifdef FLAC__SSE2_SUPPORTED
void FLAC__lpc_restore_signal_16_intrin_sse2(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
# endif
# ifdef FLAC__SSE4_1_SUPPORTED
void FLAC__lpc_restore_signal_wide_intrin_sse41(const FLAC__int32 residual[], unsigned data_len, const FLAC__int32 qlp_coeff[], unsigned order, int lp_quantization, FLAC__int32 data[]);
# endif
# endif
#endif /* FLAC__NO_ASM */
#ifndef FLAC__INTEGER_ONLY_LIBRARY
/*
* FLAC__lpc_compute_expected_bits_per_residual_sample()
* --------------------------------------------------------------------
* Compute the expected number of bits per residual signal sample
* based on the LP error (which is related to the residual variance).
*
* IN lpc_error >= 0.0 error returned from calculating LP coefficients
* IN total_samples > 0 # of samples in residual signal
* RETURN expected bits per sample
*/
FLAC__double FLAC__lpc_compute_expected_bits_per_residual_sample(FLAC__double lpc_error, unsigned total_samples);
FLAC__double FLAC__lpc_compute_expected_bits_per_residual_sample_with_error_scale(FLAC__double lpc_error, FLAC__double error_scale);
/*
* FLAC__lpc_compute_best_order()
* --------------------------------------------------------------------
* Compute the best order from the array of signal errors returned
* during coefficient computation.
*
* IN lpc_error[0,max_order-1] >= 0.0 error returned from calculating LP coefficients
* IN max_order > 0 max LP order
* IN total_samples > 0 # of samples in residual signal
* IN overhead_bits_per_order # of bits overhead for each increased LP order
* (includes warmup sample size and quantized LP coefficient)
* RETURN [1,max_order] best order
*/
unsigned FLAC__lpc_compute_best_order(const FLAC__double lpc_error[], unsigned max_order, unsigned total_samples, unsigned overhead_bits_per_order);
#endif /* !defined FLAC__INTEGER_ONLY_LIBRARY */
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/md5.h
+50
View File
@@ -0,0 +1,50 @@
#ifndef FLAC__PRIVATE__MD5_H
#define FLAC__PRIVATE__MD5_H
/*
* This is the header file for the MD5 message-digest algorithm.
* The algorithm is due to Ron Rivest. This code was
* written by Colin Plumb in 1993, no copyright is claimed.
* This code is in the public domain; do with it what you wish.
*
* Equivalent code is available from RSA Data Security, Inc.
* This code has been tested against that, and is equivalent,
* except that you don't need to include two pages of legalese
* with every copy.
*
* To compute the message digest of a chunk of bytes, declare an
* MD5Context structure, pass it to MD5Init, call MD5Update as
* needed on buffers full of bytes, and then call MD5Final, which
* will fill a supplied 16-byte array with the digest.
*
* Changed so as no longer to depend on Colin Plumb's `usual.h'
* header definitions; now uses stuff from dpkg's config.h
* - Ian Jackson <ijackson@nyx.cs.du.edu>.
* Still in the public domain.
*
* Josh Coalson: made some changes to integrate with libFLAC.
* Still in the public domain, with no warranty.
*/
#include "../../../ordinals.h"
typedef union {
FLAC__byte *p8;
FLAC__int16 *p16;
FLAC__int32 *p32;
} FLAC__multibyte;
typedef struct {
FLAC__uint32 in[16];
FLAC__uint32 buf[4];
FLAC__uint32 bytes[2];
FLAC__multibyte internal_buf;
size_t capacity;
} FLAC__MD5Context;
void FLAC__MD5Init(FLAC__MD5Context *context);
void FLAC__MD5Final(FLAC__byte digest[16], FLAC__MD5Context *context);
FLAC__bool FLAC__MD5Accumulate(FLAC__MD5Context *ctx, const FLAC__int32 * const signal[], unsigned channels, unsigned samples, unsigned bytes_per_sample);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/memory.h
+58
View File
@@ -0,0 +1,58 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2001-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__MEMORY_H
#define FLAC__PRIVATE__MEMORY_H
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <stdlib.h> /* for size_t */
#include "float.h"
#include "../../../ordinals.h" /* for FLAC__bool */
/* Returns the unaligned address returned by malloc.
* Use free() on this address to deallocate.
*/
void *FLAC__memory_alloc_aligned(size_t bytes, void **aligned_address);
FLAC__bool FLAC__memory_alloc_aligned_int32_array(size_t elements, FLAC__int32 **unaligned_pointer, FLAC__int32 **aligned_pointer);
FLAC__bool FLAC__memory_alloc_aligned_uint32_array(size_t elements, FLAC__uint32 **unaligned_pointer, FLAC__uint32 **aligned_pointer);
FLAC__bool FLAC__memory_alloc_aligned_uint64_array(size_t elements, FLAC__uint64 **unaligned_pointer, FLAC__uint64 **aligned_pointer);
FLAC__bool FLAC__memory_alloc_aligned_unsigned_array(size_t elements, unsigned **unaligned_pointer, unsigned **aligned_pointer);
#ifndef FLAC__INTEGER_ONLY_LIBRARY
FLAC__bool FLAC__memory_alloc_aligned_real_array(size_t elements, FLAC__real **unaligned_pointer, FLAC__real **aligned_pointer);
#endif
void *safe_malloc_mul_2op_p(size_t size1, size_t size2);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/metadata.h
+46
View File
@@ -0,0 +1,46 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2002-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__METADATA_H
#define FLAC__PRIVATE__METADATA_H
#include "FLAC/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
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/stream_encoder.h
+67
View File
@@ -0,0 +1,67 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__STREAM_ENCODER_H
#define FLAC__PRIVATE__STREAM_ENCODER_H
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
/*
* This is used to avoid overflow with unusual signals in 32-bit
* accumulator in the *precompute_partition_info_sums_* functions.
*/
#define FLAC__MAX_EXTRA_RESIDUAL_BPS 4
#if (defined FLAC__CPU_IA32 || defined FLAC__CPU_X86_64) && defined FLAC__HAS_X86INTRIN
#include "cpu.h"
#include "../../../format.h"
#ifdef FLAC__SSE2_SUPPORTED
extern void FLAC__precompute_partition_info_sums_intrin_sse2(const FLAC__int32 residual[], FLAC__uint64 abs_residual_partition_sums[],
unsigned residual_samples, unsigned predictor_order, unsigned min_partition_order, unsigned max_partition_order, unsigned bps);
#endif
#ifdef FLAC__SSSE3_SUPPORTED
extern void FLAC__precompute_partition_info_sums_intrin_ssse3(const FLAC__int32 residual[], FLAC__uint64 abs_residual_partition_sums[],
unsigned residual_samples, unsigned predictor_order, unsigned min_partition_order, unsigned max_partition_order, unsigned bps);
#endif
#ifdef FLAC__AVX2_SUPPORTED
extern void FLAC__precompute_partition_info_sums_intrin_avx2(const FLAC__int32 residual[], FLAC__uint64 abs_residual_partition_sums[],
unsigned residual_samples, unsigned predictor_order, unsigned min_partition_order, unsigned max_partition_order, unsigned bps);
#endif
#endif
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/stream_encoder_framing.h
+46
View File
@@ -0,0 +1,46 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__STREAM_ENCODER_FRAMING_H
#define FLAC__PRIVATE__STREAM_ENCODER_FRAMING_H
#include "../../../format.h"
#include "bitwriter.h"
FLAC__bool FLAC__add_metadata_block(const FLAC__StreamMetadata *metadata, FLAC__BitWriter *bw);
FLAC__bool FLAC__frame_add_header(const FLAC__FrameHeader *header, FLAC__BitWriter *bw);
FLAC__bool FLAC__subframe_add_constant(const FLAC__Subframe_Constant *subframe, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw);
FLAC__bool FLAC__subframe_add_fixed(const FLAC__Subframe_Fixed *subframe, unsigned residual_samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw);
FLAC__bool FLAC__subframe_add_lpc(const FLAC__Subframe_LPC *subframe, unsigned residual_samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw);
FLAC__bool FLAC__subframe_add_verbatim(const FLAC__Subframe_Verbatim *subframe, unsigned samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/private/window.h
+74
View File
@@ -0,0 +1,74 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2006-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PRIVATE__WINDOW_H
#define FLAC__PRIVATE__WINDOW_H
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include "float.h"
#include "../../../format.h"
#ifndef FLAC__INTEGER_ONLY_LIBRARY
/*
* FLAC__window_*()
* --------------------------------------------------------------------
* Calculates window coefficients according to different apodization
* functions.
*
* OUT window[0,L-1]
* IN L (number of points in window)
*/
void FLAC__window_bartlett(FLAC__real *window, const FLAC__int32 L);
void FLAC__window_bartlett_hann(FLAC__real *window, const FLAC__int32 L);
void FLAC__window_blackman(FLAC__real *window, const FLAC__int32 L);
void FLAC__window_blackman_harris_4term_92db_sidelobe(FLAC__real *window, const FLAC__int32 L);
void FLAC__window_connes(FLAC__real *window, const FLAC__int32 L);
void FLAC__window_flattop(FLAC__real *window, const FLAC__int32 L);
void FLAC__window_gauss(FLAC__real *window, const FLAC__int32 L, const FLAC__real stddev); /* 0.0 < stddev <= 0.5 */
void FLAC__window_hamming(FLAC__real *window, const FLAC__int32 L);
void FLAC__window_hann(FLAC__real *window, const FLAC__int32 L);
void FLAC__window_kaiser_bessel(FLAC__real *window, const FLAC__int32 L);
void FLAC__window_nuttall(FLAC__real *window, const FLAC__int32 L);
void FLAC__window_rectangle(FLAC__real *window, const FLAC__int32 L);
void FLAC__window_triangle(FLAC__real *window, const FLAC__int32 L);
void FLAC__window_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p);
void FLAC__window_partial_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p, const FLAC__real start, const FLAC__real end);
void FLAC__window_punchout_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p, const FLAC__real start, const FLAC__real end);
void FLAC__window_welch(FLAC__real *window, const FLAC__int32 L);
#endif /* !defined FLAC__INTEGER_ONLY_LIBRARY */
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/protected/all.h
+39
View File
@@ -0,0 +1,39 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2001-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PROTECTED__ALL_H
#define FLAC__PROTECTED__ALL_H
#include "stream_decoder.h"
#include "stream_encoder.h"
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/protected/stream_decoder.h
+60
View File
@@ -0,0 +1,60 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PROTECTED__STREAM_DECODER_H
#define FLAC__PROTECTED__STREAM_DECODER_H
#include "../../../stream_decoder.h"
#if FLAC__HAS_OGG
#include "../private/ogg_decoder_aspect.h"
#endif
typedef struct FLAC__StreamDecoderProtected {
FLAC__StreamDecoderState state;
FLAC__StreamDecoderInitStatus initstate;
unsigned channels;
FLAC__ChannelAssignment channel_assignment;
unsigned bits_per_sample;
unsigned sample_rate; /* in Hz */
unsigned blocksize; /* in samples (per channel) */
FLAC__bool md5_checking; /* if true, generate MD5 signature of decoded data and compare against signature in the STREAMINFO metadata block */
#if FLAC__HAS_OGG
FLAC__OggDecoderAspect ogg_decoder_aspect;
#endif
} FLAC__StreamDecoderProtected;
/*
* return the number of input bytes consumed
*/
unsigned FLAC__stream_decoder_get_input_bytes_unconsumed(const FLAC__StreamDecoder *decoder);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/include/protected/stream_encoder.h
+118
View File
@@ -0,0 +1,118 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2001-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__PROTECTED__STREAM_ENCODER_H
#define FLAC__PROTECTED__STREAM_ENCODER_H
#include "../../../stream_encoder.h"
#if FLAC__HAS_OGG
#include "../private/ogg_encoder_aspect.h"
#endif
#ifndef FLAC__INTEGER_ONLY_LIBRARY
#include "../private/float.h"
#define FLAC__MAX_APODIZATION_FUNCTIONS 32
typedef enum {
FLAC__APODIZATION_BARTLETT,
FLAC__APODIZATION_BARTLETT_HANN,
FLAC__APODIZATION_BLACKMAN,
FLAC__APODIZATION_BLACKMAN_HARRIS_4TERM_92DB_SIDELOBE,
FLAC__APODIZATION_CONNES,
FLAC__APODIZATION_FLATTOP,
FLAC__APODIZATION_GAUSS,
FLAC__APODIZATION_HAMMING,
FLAC__APODIZATION_HANN,
FLAC__APODIZATION_KAISER_BESSEL,
FLAC__APODIZATION_NUTTALL,
FLAC__APODIZATION_RECTANGLE,
FLAC__APODIZATION_TRIANGLE,
FLAC__APODIZATION_TUKEY,
FLAC__APODIZATION_PARTIAL_TUKEY,
FLAC__APODIZATION_PUNCHOUT_TUKEY,
FLAC__APODIZATION_WELCH
} FLAC__ApodizationFunction;
typedef struct {
FLAC__ApodizationFunction type;
union {
struct {
FLAC__real stddev;
} gauss;
struct {
FLAC__real p;
} tukey;
struct {
FLAC__real p;
FLAC__real start;
FLAC__real end;
} multiple_tukey;
} parameters;
} FLAC__ApodizationSpecification;
#endif // #ifndef FLAC__INTEGER_ONLY_LIBRARY
typedef struct FLAC__StreamEncoderProtected {
FLAC__StreamEncoderState state;
FLAC__bool verify;
FLAC__bool streamable_subset;
FLAC__bool do_md5;
FLAC__bool do_mid_side_stereo;
FLAC__bool loose_mid_side_stereo;
unsigned channels;
unsigned bits_per_sample;
unsigned sample_rate;
unsigned blocksize;
#ifndef FLAC__INTEGER_ONLY_LIBRARY
unsigned num_apodizations;
FLAC__ApodizationSpecification apodizations[FLAC__MAX_APODIZATION_FUNCTIONS];
#endif
unsigned max_lpc_order;
unsigned qlp_coeff_precision;
FLAC__bool do_qlp_coeff_prec_search;
FLAC__bool do_exhaustive_model_search;
FLAC__bool do_escape_coding;
unsigned min_residual_partition_order;
unsigned max_residual_partition_order;
unsigned rice_parameter_search_dist;
FLAC__uint64 total_samples_estimate;
FLAC__StreamMetadata **metadata;
unsigned num_metadata_blocks;
FLAC__uint64 streaminfo_offset, seektable_offset, audio_offset;
#if FLAC__HAS_OGG
FLAC__OggEncoderAspect ogg_encoder_aspect;
#endif
} FLAC__StreamEncoderProtected;
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/lpc_flac.c
+1356
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/md5.c
+518
View File
@@ -0,0 +1,518 @@
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <stdlib.h> /* for malloc() */
#include <string.h> /* for memcpy() */
#include "include/private/md5.h"
#include "../alloc.h"
#include "../endswap.h"
/*
* This code implements the MD5 message-digest algorithm.
* The algorithm is due to Ron Rivest. This code was
* written by Colin Plumb in 1993, no copyright is claimed.
* This code is in the public domain; do with it what you wish.
*
* Equivalent code is available from RSA Data Security, Inc.
* This code has been tested against that, and is equivalent,
* except that you don't need to include two pages of legalese
* with every copy.
*
* To compute the message digest of a chunk of bytes, declare an
* MD5Context structure, pass it to MD5Init, call MD5Update as
* needed on buffers full of bytes, and then call MD5Final, which
* will fill a supplied 16-byte array with the digest.
*
* Changed so as no longer to depend on Colin Plumb's `usual.h' header
* definitions; now uses stuff from dpkg's config.h.
* - Ian Jackson <ijackson@nyx.cs.du.edu>.
* Still in the public domain.
*
* Josh Coalson: made some changes to integrate with libFLAC.
* Still in the public domain.
*/
/* The four core functions - F1 is optimized somewhat */
/* #define F1(x, y, z) (x & y | ~x & z) */
#define F1(x, y, z) (z ^ (x & (y ^ z)))
#define F2(x, y, z) F1(z, x, y)
#define F3(x, y, z) (x ^ y ^ z)
#define F4(x, y, z) (y ^ (x | ~z))
/* This is the central step in the MD5 algorithm. */
#define MD5STEP(f,w,x,y,z,in,s) \
(w += f(x,y,z) + in, w = (w<<s | w>>(32-s)) + x)
/*
* The core of the MD5 algorithm, this alters an existing MD5 hash to
* reflect the addition of 16 longwords of new data. MD5Update blocks
* the data and converts bytes into longwords for this routine.
*/
static void FLAC__MD5Transform(FLAC__uint32 buf[4], FLAC__uint32 const in[16])
{
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.p8= 0;
ctx->capacity = 0;
}
/*
* Final wrapup - pad to 64-byte boundary with the bit pattern
* 1 0* (64-bit count of bits processed, MSB-first)
*/
void FLAC__MD5Final(FLAC__byte digest[16], FLAC__MD5Context *ctx)
{
int count = ctx->bytes[0] & 0x3f; /* Number of bytes in ctx->in */
FLAC__byte *p = (FLAC__byte *)ctx->in + count;
/* Set the first char of padding to 0x80. There is always room. */
*p++ = 0x80;
/* Bytes of padding needed to make 56 bytes (-8..55) */
count = 56 - 1 - count;
if (count < 0) { /* Padding forces an extra block */
memset(p, 0, count + 8);
byteSwapX16(ctx->in);
FLAC__MD5Transform(ctx->buf, ctx->in);
p = (FLAC__byte *)ctx->in;
count = 56;
}
memset(p, 0, count);
byteSwap(ctx->in, 14);
/* Append length in bits and transform */
ctx->in[14] = ctx->bytes[0] << 3;
ctx->in[15] = ctx->bytes[1] << 3 | ctx->bytes[0] >> 29;
FLAC__MD5Transform(ctx->buf, ctx->in);
byteSwap(ctx->buf, 4);
memcpy(digest, ctx->buf, 16);
if (0 != ctx->internal_buf.p8) {
free(ctx->internal_buf.p8);
ctx->internal_buf.p8= 0;
ctx->capacity = 0;
}
memset(ctx, 0, sizeof(*ctx)); /* In case it's sensitive */
}
/*
* Convert the incoming audio signal to a byte stream
*/
static void format_input_(FLAC__multibyte *mbuf, const FLAC__int32 * const signal[], unsigned channels, unsigned samples, unsigned bytes_per_sample)
{
FLAC__byte *buf_ = mbuf->p8;
FLAC__int16 *buf16 = mbuf->p16;
FLAC__int32 *buf32 = mbuf->p32;
FLAC__int32 a_word;
unsigned channel, sample;
/* Storage in the output buffer, buf, is little endian. */
#define BYTES_CHANNEL_SELECTOR(bytes, channels) (bytes * 100 + channels)
/* First do the most commonly used combinations. */
switch (BYTES_CHANNEL_SELECTOR (bytes_per_sample, channels)) {
/* One byte per sample. */
case (BYTES_CHANNEL_SELECTOR (1, 1)):
for (sample = 0; sample < samples; sample++)
*buf_++ = signal[0][sample];
return;
case (BYTES_CHANNEL_SELECTOR (1, 2)):
for (sample = 0; sample < samples; sample++) {
*buf_++ = signal[0][sample];
*buf_++ = signal[1][sample];
}
return;
case (BYTES_CHANNEL_SELECTOR (1, 4)):
for (sample = 0; sample < samples; sample++) {
*buf_++ = signal[0][sample];
*buf_++ = signal[1][sample];
*buf_++ = signal[2][sample];
*buf_++ = signal[3][sample];
}
return;
case (BYTES_CHANNEL_SELECTOR (1, 6)):
for (sample = 0; sample < samples; sample++) {
*buf_++ = signal[0][sample];
*buf_++ = signal[1][sample];
*buf_++ = signal[2][sample];
*buf_++ = signal[3][sample];
*buf_++ = signal[4][sample];
*buf_++ = signal[5][sample];
}
return;
case (BYTES_CHANNEL_SELECTOR (1, 8)):
for (sample = 0; sample < samples; sample++) {
*buf_++ = signal[0][sample];
*buf_++ = signal[1][sample];
*buf_++ = signal[2][sample];
*buf_++ = signal[3][sample];
*buf_++ = signal[4][sample];
*buf_++ = signal[5][sample];
*buf_++ = signal[6][sample];
*buf_++ = signal[7][sample];
}
return;
/* Two bytes per sample. */
case (BYTES_CHANNEL_SELECTOR (2, 1)):
for (sample = 0; sample < samples; sample++)
*buf16++ = H2LE_16(signal[0][sample]);
return;
case (BYTES_CHANNEL_SELECTOR (2, 2)):
for (sample = 0; sample < samples; sample++) {
*buf16++ = H2LE_16(signal[0][sample]);
*buf16++ = H2LE_16(signal[1][sample]);
}
return;
case (BYTES_CHANNEL_SELECTOR (2, 4)):
for (sample = 0; sample < samples; sample++) {
*buf16++ = H2LE_16(signal[0][sample]);
*buf16++ = H2LE_16(signal[1][sample]);
*buf16++ = H2LE_16(signal[2][sample]);
*buf16++ = H2LE_16(signal[3][sample]);
}
return;
case (BYTES_CHANNEL_SELECTOR (2, 6)):
for (sample = 0; sample < samples; sample++) {
*buf16++ = H2LE_16(signal[0][sample]);
*buf16++ = H2LE_16(signal[1][sample]);
*buf16++ = H2LE_16(signal[2][sample]);
*buf16++ = H2LE_16(signal[3][sample]);
*buf16++ = H2LE_16(signal[4][sample]);
*buf16++ = H2LE_16(signal[5][sample]);
}
return;
case (BYTES_CHANNEL_SELECTOR (2, 8)):
for (sample = 0; sample < samples; sample++) {
*buf16++ = H2LE_16(signal[0][sample]);
*buf16++ = H2LE_16(signal[1][sample]);
*buf16++ = H2LE_16(signal[2][sample]);
*buf16++ = H2LE_16(signal[3][sample]);
*buf16++ = H2LE_16(signal[4][sample]);
*buf16++ = H2LE_16(signal[5][sample]);
*buf16++ = H2LE_16(signal[6][sample]);
*buf16++ = H2LE_16(signal[7][sample]);
}
return;
/* Three bytes per sample. */
case (BYTES_CHANNEL_SELECTOR (3, 1)):
for (sample = 0; sample < samples; sample++) {
a_word = signal[0][sample];
*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
*buf_++ = (FLAC__byte)a_word;
}
return;
case (BYTES_CHANNEL_SELECTOR (3, 2)):
for (sample = 0; sample < samples; sample++) {
a_word = signal[0][sample];
*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
*buf_++ = (FLAC__byte)a_word;
a_word = signal[1][sample];
*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
*buf_++ = (FLAC__byte)a_word;
}
return;
/* Four bytes per sample. */
case (BYTES_CHANNEL_SELECTOR (4, 1)):
for (sample = 0; sample < samples; sample++)
*buf32++ = H2LE_32(signal[0][sample]);
return;
case (BYTES_CHANNEL_SELECTOR (4, 2)):
for (sample = 0; sample < samples; sample++) {
*buf32++ = H2LE_32(signal[0][sample]);
*buf32++ = H2LE_32(signal[1][sample]);
}
return;
case (BYTES_CHANNEL_SELECTOR (4, 4)):
for (sample = 0; sample < samples; sample++) {
*buf32++ = H2LE_32(signal[0][sample]);
*buf32++ = H2LE_32(signal[1][sample]);
*buf32++ = H2LE_32(signal[2][sample]);
*buf32++ = H2LE_32(signal[3][sample]);
}
return;
case (BYTES_CHANNEL_SELECTOR (4, 6)):
for (sample = 0; sample < samples; sample++) {
*buf32++ = H2LE_32(signal[0][sample]);
*buf32++ = H2LE_32(signal[1][sample]);
*buf32++ = H2LE_32(signal[2][sample]);
*buf32++ = H2LE_32(signal[3][sample]);
*buf32++ = H2LE_32(signal[4][sample]);
*buf32++ = H2LE_32(signal[5][sample]);
}
return;
case (BYTES_CHANNEL_SELECTOR (4, 8)):
for (sample = 0; sample < samples; sample++) {
*buf32++ = H2LE_32(signal[0][sample]);
*buf32++ = H2LE_32(signal[1][sample]);
*buf32++ = H2LE_32(signal[2][sample]);
*buf32++ = H2LE_32(signal[3][sample]);
*buf32++ = H2LE_32(signal[4][sample]);
*buf32++ = H2LE_32(signal[5][sample]);
*buf32++ = H2LE_32(signal[6][sample]);
*buf32++ = H2LE_32(signal[7][sample]);
}
return;
default:
break;
}
/* General version. */
switch (bytes_per_sample) {
case 1:
for (sample = 0; sample < samples; sample++)
for (channel = 0; channel < channels; channel++)
*buf_++ = signal[channel][sample];
return;
case 2:
for (sample = 0; sample < samples; sample++)
for (channel = 0; channel < channels; channel++)
*buf16++ = H2LE_16(signal[channel][sample]);
return;
case 3:
for (sample = 0; sample < samples; sample++)
for (channel = 0; channel < channels; channel++) {
a_word = signal[channel][sample];
*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
*buf_++ = (FLAC__byte)a_word; a_word >>= 8;
*buf_++ = (FLAC__byte)a_word;
}
return;
case 4:
for (sample = 0; sample < samples; sample++)
for (channel = 0; channel < channels; channel++)
*buf32++ = H2LE_32(signal[channel][sample]);
return;
default:
break;
}
}
/*
* Convert the incoming audio signal to a byte stream and FLAC__MD5Update it.
*/
FLAC__bool FLAC__MD5Accumulate(FLAC__MD5Context *ctx, const FLAC__int32 * const signal[], unsigned channels, unsigned samples, unsigned bytes_per_sample)
{
const size_t bytes_needed = (size_t)channels * (size_t)samples * (size_t)bytes_per_sample;
/* overflow check */
if ((size_t)channels > SIZE_MAX / (size_t)bytes_per_sample)
return false;
if ((size_t)channels * (size_t)bytes_per_sample > SIZE_MAX / (size_t)samples)
return false;
if (ctx->capacity < bytes_needed) {
FLAC__byte *tmp = (FLAC__byte*) realloc(ctx->internal_buf.p8, bytes_needed);
if (0 == tmp) {
free(ctx->internal_buf.p8);
if (0 == (ctx->internal_buf.p8= (FLAC__byte*) safe_malloc_(bytes_needed)))
return false;
}
else
ctx->internal_buf.p8= tmp;
ctx->capacity = bytes_needed;
}
format_input_(&ctx->internal_buf, signal, channels, samples, bytes_per_sample);
FLAC__MD5Update(ctx, ctx->internal_buf.p8, bytes_needed);
return true;
}
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/memory.c
+218
View File
@@ -0,0 +1,218 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2001-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#ifdef HAVE_STDINT_H
#include <stdint.h>
#endif
#include "include/private/memory.h"
#include "../assert.h"
#include "../alloc.h"
void *FLAC__memory_alloc_aligned(size_t bytes, void **aligned_address)
{
void *x;
FLAC__ASSERT(0 != aligned_address);
#ifdef FLAC__ALIGN_MALLOC_DATA
/* align on 32-byte (256-bit) boundary */
x = safe_malloc_add_2op_(bytes, /*+*/31L);
*aligned_address = (void*)(((uintptr_t)x + 31L) & -32L);
#else
x = safe_malloc_(bytes);
*aligned_address = x;
#endif
return x;
}
FLAC__bool FLAC__memory_alloc_aligned_int32_array(size_t elements, FLAC__int32 **unaligned_pointer, FLAC__int32 **aligned_pointer)
{
FLAC__int32 *pu; /* unaligned pointer */
union { /* union needed to comply with C99 pointer aliasing rules */
FLAC__int32 *pa; /* aligned pointer */
void *pv; /* aligned pointer alias */
} u;
FLAC__ASSERT(elements > 0);
FLAC__ASSERT(0 != unaligned_pointer);
FLAC__ASSERT(0 != aligned_pointer);
FLAC__ASSERT(unaligned_pointer != aligned_pointer);
if(elements > SIZE_MAX / sizeof(*pu)) /* overflow check */
return false;
pu = (FLAC__int32*) FLAC__memory_alloc_aligned(sizeof(*pu) * elements, &u.pv);
if(0 == pu) {
return false;
}
else {
if(*unaligned_pointer != 0)
free(*unaligned_pointer);
*unaligned_pointer = pu;
*aligned_pointer = u.pa;
return true;
}
}
FLAC__bool FLAC__memory_alloc_aligned_uint32_array(size_t elements, FLAC__uint32 **unaligned_pointer, FLAC__uint32 **aligned_pointer)
{
FLAC__uint32 *pu; /* unaligned pointer */
union { /* union needed to comply with C99 pointer aliasing rules */
FLAC__uint32 *pa; /* aligned pointer */
void *pv; /* aligned pointer alias */
} u;
FLAC__ASSERT(elements > 0);
FLAC__ASSERT(0 != unaligned_pointer);
FLAC__ASSERT(0 != aligned_pointer);
FLAC__ASSERT(unaligned_pointer != aligned_pointer);
if(elements > SIZE_MAX / sizeof(*pu)) /* overflow check */
return false;
pu = (FLAC__uint32*) FLAC__memory_alloc_aligned(sizeof(*pu) * elements, &u.pv);
if(0 == pu) {
return false;
}
else {
if(*unaligned_pointer != 0)
free(*unaligned_pointer);
*unaligned_pointer = pu;
*aligned_pointer = u.pa;
return true;
}
}
FLAC__bool FLAC__memory_alloc_aligned_uint64_array(size_t elements, FLAC__uint64 **unaligned_pointer, FLAC__uint64 **aligned_pointer)
{
FLAC__uint64 *pu; /* unaligned pointer */
union { /* union needed to comply with C99 pointer aliasing rules */
FLAC__uint64 *pa; /* aligned pointer */
void *pv; /* aligned pointer alias */
} u;
FLAC__ASSERT(elements > 0);
FLAC__ASSERT(0 != unaligned_pointer);
FLAC__ASSERT(0 != aligned_pointer);
FLAC__ASSERT(unaligned_pointer != aligned_pointer);
if(elements > SIZE_MAX / sizeof(*pu)) /* overflow check */
return false;
pu = (FLAC__uint64*) FLAC__memory_alloc_aligned(sizeof(*pu) * elements, &u.pv);
if(0 == pu) {
return false;
}
else {
if(*unaligned_pointer != 0)
free(*unaligned_pointer);
*unaligned_pointer = pu;
*aligned_pointer = u.pa;
return true;
}
}
FLAC__bool FLAC__memory_alloc_aligned_unsigned_array(size_t elements, unsigned **unaligned_pointer, unsigned **aligned_pointer)
{
unsigned *pu; /* unaligned pointer */
union { /* union needed to comply with C99 pointer aliasing rules */
unsigned *pa; /* aligned pointer */
void *pv; /* aligned pointer alias */
} u;
FLAC__ASSERT(elements > 0);
FLAC__ASSERT(0 != unaligned_pointer);
FLAC__ASSERT(0 != aligned_pointer);
FLAC__ASSERT(unaligned_pointer != aligned_pointer);
if(elements > SIZE_MAX / sizeof(*pu)) /* overflow check */
return false;
pu = (unsigned int*) FLAC__memory_alloc_aligned(sizeof(*pu) * elements, &u.pv);
if(0 == pu) {
return false;
}
else {
if(*unaligned_pointer != 0)
free(*unaligned_pointer);
*unaligned_pointer = pu;
*aligned_pointer = u.pa;
return true;
}
}
#ifndef FLAC__INTEGER_ONLY_LIBRARY
FLAC__bool FLAC__memory_alloc_aligned_real_array(size_t elements, FLAC__real **unaligned_pointer, FLAC__real **aligned_pointer)
{
FLAC__real *pu; /* unaligned pointer */
union { /* union needed to comply with C99 pointer aliasing rules */
FLAC__real *pa; /* aligned pointer */
void *pv; /* aligned pointer alias */
} u;
FLAC__ASSERT(elements > 0);
FLAC__ASSERT(0 != unaligned_pointer);
FLAC__ASSERT(0 != aligned_pointer);
FLAC__ASSERT(unaligned_pointer != aligned_pointer);
if(elements > SIZE_MAX / sizeof(*pu)) /* overflow check */
return false;
pu = (FLAC__real*) FLAC__memory_alloc_aligned(sizeof(*pu) * elements, &u.pv);
if(0 == pu) {
return false;
}
else {
if(*unaligned_pointer != 0)
free(*unaligned_pointer);
*unaligned_pointer = pu;
*aligned_pointer = u.pa;
return true;
}
}
#endif
void *safe_malloc_mul_2op_p(size_t size1, size_t size2)
{
if(!size1 || !size2)
return malloc(1); /* malloc(0) is undefined; FLAC src convention is to always allocate */
if(size1 > SIZE_MAX / size2)
return 0;
return malloc(size1*size2);
}
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/stream_decoder.c
+3395
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/stream_encoder.c
+4527
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/stream_encoder_framing.c
+549
View File
@@ -0,0 +1,549 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <stdio.h>
#include <string.h> /* for strlen() */
#include "include/private/stream_encoder_framing.h"
#include "include/private/crc.h"
#include "../assert.h"
static FLAC__bool add_entropy_coding_method_(FLAC__BitWriter *bw, const FLAC__EntropyCodingMethod *method);
static FLAC__bool add_residual_partitioned_rice_(FLAC__BitWriter *bw, const FLAC__int32 residual[], const unsigned residual_samples, const unsigned predictor_order, const unsigned rice_parameters[], const unsigned raw_bits[], const unsigned partition_order, const FLAC__bool is_extended);
FLAC__bool FLAC__add_metadata_block(const FLAC__StreamMetadata *metadata, FLAC__BitWriter *bw)
{
unsigned i, j;
const unsigned vendor_string_length = (unsigned)strlen(FLAC__VENDOR_STRING);
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->is_last, FLAC__STREAM_METADATA_IS_LAST_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->type, FLAC__STREAM_METADATA_TYPE_LEN))
return false;
/*
* First, for VORBIS_COMMENTs, adjust the length to reflect our vendor string
*/
i = metadata->length;
if(metadata->type == FLAC__METADATA_TYPE_VORBIS_COMMENT) {
FLAC__ASSERT(metadata->data.vorbis_comment.vendor_string.length == 0 || 0 != metadata->data.vorbis_comment.vendor_string.entry);
i -= metadata->data.vorbis_comment.vendor_string.length;
i += vendor_string_length;
}
FLAC__ASSERT(i < (1u << FLAC__STREAM_METADATA_LENGTH_LEN));
if(!FLAC__bitwriter_write_raw_uint32(bw, i, FLAC__STREAM_METADATA_LENGTH_LEN))
return false;
switch(metadata->type) {
case FLAC__METADATA_TYPE_STREAMINFO:
FLAC__ASSERT(metadata->data.stream_info.min_blocksize < (1u << FLAC__STREAM_METADATA_STREAMINFO_MIN_BLOCK_SIZE_LEN));
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.min_blocksize, FLAC__STREAM_METADATA_STREAMINFO_MIN_BLOCK_SIZE_LEN))
return false;
FLAC__ASSERT(metadata->data.stream_info.max_blocksize < (1u << FLAC__STREAM_METADATA_STREAMINFO_MAX_BLOCK_SIZE_LEN));
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.max_blocksize, FLAC__STREAM_METADATA_STREAMINFO_MAX_BLOCK_SIZE_LEN))
return false;
FLAC__ASSERT(metadata->data.stream_info.min_framesize < (1u << FLAC__STREAM_METADATA_STREAMINFO_MIN_FRAME_SIZE_LEN));
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.min_framesize, FLAC__STREAM_METADATA_STREAMINFO_MIN_FRAME_SIZE_LEN))
return false;
FLAC__ASSERT(metadata->data.stream_info.max_framesize < (1u << FLAC__STREAM_METADATA_STREAMINFO_MAX_FRAME_SIZE_LEN));
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.max_framesize, FLAC__STREAM_METADATA_STREAMINFO_MAX_FRAME_SIZE_LEN))
return false;
FLAC__ASSERT(FLAC__format_sample_rate_is_valid(metadata->data.stream_info.sample_rate));
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.sample_rate, FLAC__STREAM_METADATA_STREAMINFO_SAMPLE_RATE_LEN))
return false;
FLAC__ASSERT(metadata->data.stream_info.channels > 0);
FLAC__ASSERT(metadata->data.stream_info.channels <= (1u << FLAC__STREAM_METADATA_STREAMINFO_CHANNELS_LEN));
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.channels-1, FLAC__STREAM_METADATA_STREAMINFO_CHANNELS_LEN))
return false;
FLAC__ASSERT(metadata->data.stream_info.bits_per_sample > 0);
FLAC__ASSERT(metadata->data.stream_info.bits_per_sample <= (1u << FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN));
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.stream_info.bits_per_sample-1, FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint64(bw, metadata->data.stream_info.total_samples, FLAC__STREAM_METADATA_STREAMINFO_TOTAL_SAMPLES_LEN))
return false;
if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.stream_info.md5sum, 16))
return false;
break;
case FLAC__METADATA_TYPE_PADDING:
if(!FLAC__bitwriter_write_zeroes(bw, metadata->length * 8))
return false;
break;
case FLAC__METADATA_TYPE_APPLICATION:
if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.application.id, FLAC__STREAM_METADATA_APPLICATION_ID_LEN / 8))
return false;
if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.application.data, metadata->length - (FLAC__STREAM_METADATA_APPLICATION_ID_LEN / 8)))
return false;
break;
case FLAC__METADATA_TYPE_SEEKTABLE:
for(i = 0; i < metadata->data.seek_table.num_points; i++) {
if(!FLAC__bitwriter_write_raw_uint64(bw, metadata->data.seek_table.points[i].sample_number, FLAC__STREAM_METADATA_SEEKPOINT_SAMPLE_NUMBER_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint64(bw, metadata->data.seek_table.points[i].stream_offset, FLAC__STREAM_METADATA_SEEKPOINT_STREAM_OFFSET_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.seek_table.points[i].frame_samples, FLAC__STREAM_METADATA_SEEKPOINT_FRAME_SAMPLES_LEN))
return false;
}
break;
case FLAC__METADATA_TYPE_VORBIS_COMMENT:
if(!FLAC__bitwriter_write_raw_uint32_little_endian(bw, vendor_string_length))
return false;
if(!FLAC__bitwriter_write_byte_block(bw, (const FLAC__byte*)FLAC__VENDOR_STRING, vendor_string_length))
return false;
if(!FLAC__bitwriter_write_raw_uint32_little_endian(bw, metadata->data.vorbis_comment.num_comments))
return false;
for(i = 0; i < metadata->data.vorbis_comment.num_comments; i++) {
if(!FLAC__bitwriter_write_raw_uint32_little_endian(bw, metadata->data.vorbis_comment.comments[i].length))
return false;
if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.vorbis_comment.comments[i].entry, metadata->data.vorbis_comment.comments[i].length))
return false;
}
break;
case FLAC__METADATA_TYPE_CUESHEET:
FLAC__ASSERT(FLAC__STREAM_METADATA_CUESHEET_MEDIA_CATALOG_NUMBER_LEN % 8 == 0);
if(!FLAC__bitwriter_write_byte_block(bw, (const FLAC__byte*)metadata->data.cue_sheet.media_catalog_number, FLAC__STREAM_METADATA_CUESHEET_MEDIA_CATALOG_NUMBER_LEN/8))
return false;
if(!FLAC__bitwriter_write_raw_uint64(bw, metadata->data.cue_sheet.lead_in, FLAC__STREAM_METADATA_CUESHEET_LEAD_IN_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.cue_sheet.is_cd? 1 : 0, FLAC__STREAM_METADATA_CUESHEET_IS_CD_LEN))
return false;
if(!FLAC__bitwriter_write_zeroes(bw, FLAC__STREAM_METADATA_CUESHEET_RESERVED_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.cue_sheet.num_tracks, FLAC__STREAM_METADATA_CUESHEET_NUM_TRACKS_LEN))
return false;
for(i = 0; i < metadata->data.cue_sheet.num_tracks; i++) {
const FLAC__StreamMetadata_CueSheet_Track *track = metadata->data.cue_sheet.tracks + i;
if(!FLAC__bitwriter_write_raw_uint64(bw, track->offset, FLAC__STREAM_METADATA_CUESHEET_TRACK_OFFSET_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, track->number, FLAC__STREAM_METADATA_CUESHEET_TRACK_NUMBER_LEN))
return false;
FLAC__ASSERT(FLAC__STREAM_METADATA_CUESHEET_TRACK_ISRC_LEN % 8 == 0);
if(!FLAC__bitwriter_write_byte_block(bw, (const FLAC__byte*)track->isrc, FLAC__STREAM_METADATA_CUESHEET_TRACK_ISRC_LEN/8))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, track->type, FLAC__STREAM_METADATA_CUESHEET_TRACK_TYPE_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, track->pre_emphasis, FLAC__STREAM_METADATA_CUESHEET_TRACK_PRE_EMPHASIS_LEN))
return false;
if(!FLAC__bitwriter_write_zeroes(bw, FLAC__STREAM_METADATA_CUESHEET_TRACK_RESERVED_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, track->num_indices, FLAC__STREAM_METADATA_CUESHEET_TRACK_NUM_INDICES_LEN))
return false;
for(j = 0; j < track->num_indices; j++) {
const FLAC__StreamMetadata_CueSheet_Index *indx = track->indices + j;
if(!FLAC__bitwriter_write_raw_uint64(bw, indx->offset, FLAC__STREAM_METADATA_CUESHEET_INDEX_OFFSET_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, indx->number, FLAC__STREAM_METADATA_CUESHEET_INDEX_NUMBER_LEN))
return false;
if(!FLAC__bitwriter_write_zeroes(bw, FLAC__STREAM_METADATA_CUESHEET_INDEX_RESERVED_LEN))
return false;
}
}
break;
case FLAC__METADATA_TYPE_PICTURE:
{
size_t len;
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.type, FLAC__STREAM_METADATA_PICTURE_TYPE_LEN))
return false;
len = strlen(metadata->data.picture.mime_type);
if(!FLAC__bitwriter_write_raw_uint32(bw, len, FLAC__STREAM_METADATA_PICTURE_MIME_TYPE_LENGTH_LEN))
return false;
if(!FLAC__bitwriter_write_byte_block(bw, (const FLAC__byte*)metadata->data.picture.mime_type, len))
return false;
len = strlen((const char *)metadata->data.picture.description);
if(!FLAC__bitwriter_write_raw_uint32(bw, len, FLAC__STREAM_METADATA_PICTURE_DESCRIPTION_LENGTH_LEN))
return false;
if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.picture.description, len))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.width, FLAC__STREAM_METADATA_PICTURE_WIDTH_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.height, FLAC__STREAM_METADATA_PICTURE_HEIGHT_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.depth, FLAC__STREAM_METADATA_PICTURE_DEPTH_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.colors, FLAC__STREAM_METADATA_PICTURE_COLORS_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, metadata->data.picture.data_length, FLAC__STREAM_METADATA_PICTURE_DATA_LENGTH_LEN))
return false;
if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.picture.data, metadata->data.picture.data_length))
return false;
}
break;
default:
if(!FLAC__bitwriter_write_byte_block(bw, metadata->data.unknown.data, metadata->length))
return false;
break;
}
FLAC__ASSERT(FLAC__bitwriter_is_byte_aligned(bw));
return true;
}
FLAC__bool FLAC__frame_add_header(const FLAC__FrameHeader *header, FLAC__BitWriter *bw)
{
unsigned u, blocksize_hint, sample_rate_hint;
FLAC__byte crc;
FLAC__ASSERT(FLAC__bitwriter_is_byte_aligned(bw));
if(!FLAC__bitwriter_write_raw_uint32(bw, FLAC__FRAME_HEADER_SYNC, FLAC__FRAME_HEADER_SYNC_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, 0, FLAC__FRAME_HEADER_RESERVED_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, (header->number_type == FLAC__FRAME_NUMBER_TYPE_FRAME_NUMBER)? 0 : 1, FLAC__FRAME_HEADER_BLOCKING_STRATEGY_LEN))
return false;
FLAC__ASSERT(header->blocksize > 0 && header->blocksize <= FLAC__MAX_BLOCK_SIZE);
/* when this assertion holds true, any legal blocksize can be expressed in the frame header */
FLAC__ASSERT(FLAC__MAX_BLOCK_SIZE <= 65535u);
blocksize_hint = 0;
switch(header->blocksize) {
case 192: u = 1; break;
case 576: u = 2; break;
case 1152: u = 3; break;
case 2304: u = 4; break;
case 4608: u = 5; break;
case 256: u = 8; break;
case 512: u = 9; break;
case 1024: u = 10; break;
case 2048: u = 11; break;
case 4096: u = 12; break;
case 8192: u = 13; break;
case 16384: u = 14; break;
case 32768: u = 15; break;
default:
if(header->blocksize <= 0x100)
blocksize_hint = u = 6;
else
blocksize_hint = u = 7;
break;
}
if(!FLAC__bitwriter_write_raw_uint32(bw, u, FLAC__FRAME_HEADER_BLOCK_SIZE_LEN))
return false;
FLAC__ASSERT(FLAC__format_sample_rate_is_valid(header->sample_rate));
sample_rate_hint = 0;
switch(header->sample_rate) {
case 88200: u = 1; break;
case 176400: u = 2; break;
case 192000: u = 3; break;
case 8000: u = 4; break;
case 16000: u = 5; break;
case 22050: u = 6; break;
case 24000: u = 7; break;
case 32000: u = 8; break;
case 44100: u = 9; break;
case 48000: u = 10; break;
case 96000: u = 11; break;
default:
if(header->sample_rate <= 255000 && header->sample_rate % 1000 == 0)
sample_rate_hint = u = 12;
else if(header->sample_rate % 10 == 0)
sample_rate_hint = u = 14;
else if(header->sample_rate <= 0xffff)
sample_rate_hint = u = 13;
else
u = 0;
break;
}
if(!FLAC__bitwriter_write_raw_uint32(bw, u, FLAC__FRAME_HEADER_SAMPLE_RATE_LEN))
return false;
FLAC__ASSERT(header->channels > 0 && header->channels <= (1u << FLAC__STREAM_METADATA_STREAMINFO_CHANNELS_LEN) && header->channels <= FLAC__MAX_CHANNELS);
switch(header->channel_assignment) {
case FLAC__CHANNEL_ASSIGNMENT_INDEPENDENT:
u = header->channels - 1;
break;
case FLAC__CHANNEL_ASSIGNMENT_LEFT_SIDE:
FLAC__ASSERT(header->channels == 2);
u = 8;
break;
case FLAC__CHANNEL_ASSIGNMENT_RIGHT_SIDE:
FLAC__ASSERT(header->channels == 2);
u = 9;
break;
case FLAC__CHANNEL_ASSIGNMENT_MID_SIDE:
FLAC__ASSERT(header->channels == 2);
u = 10;
break;
default:
FLAC__ASSERT(0);
}
if(!FLAC__bitwriter_write_raw_uint32(bw, u, FLAC__FRAME_HEADER_CHANNEL_ASSIGNMENT_LEN))
return false;
FLAC__ASSERT(header->bits_per_sample > 0 && header->bits_per_sample <= (1u << FLAC__STREAM_METADATA_STREAMINFO_BITS_PER_SAMPLE_LEN));
switch(header->bits_per_sample) {
case 8 : u = 1; break;
case 12: u = 2; break;
case 16: u = 4; break;
case 20: u = 5; break;
case 24: u = 6; break;
default: u = 0; break;
}
if(!FLAC__bitwriter_write_raw_uint32(bw, u, FLAC__FRAME_HEADER_BITS_PER_SAMPLE_LEN))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, 0, FLAC__FRAME_HEADER_ZERO_PAD_LEN))
return false;
if(header->number_type == FLAC__FRAME_NUMBER_TYPE_FRAME_NUMBER) {
if(!FLAC__bitwriter_write_utf8_uint32(bw, header->number.frame_number))
return false;
}
else {
if(!FLAC__bitwriter_write_utf8_uint64(bw, header->number.sample_number))
return false;
}
if(blocksize_hint)
if(!FLAC__bitwriter_write_raw_uint32(bw, header->blocksize-1, (blocksize_hint==6)? 8:16))
return false;
switch(sample_rate_hint) {
case 12:
if(!FLAC__bitwriter_write_raw_uint32(bw, header->sample_rate / 1000, 8))
return false;
break;
case 13:
if(!FLAC__bitwriter_write_raw_uint32(bw, header->sample_rate, 16))
return false;
break;
case 14:
if(!FLAC__bitwriter_write_raw_uint32(bw, header->sample_rate / 10, 16))
return false;
break;
}
/* write the CRC */
if(!FLAC__bitwriter_get_write_crc8(bw, &crc))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, crc, FLAC__FRAME_HEADER_CRC_LEN))
return false;
return true;
}
FLAC__bool FLAC__subframe_add_constant(const FLAC__Subframe_Constant *subframe, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw)
{
FLAC__bool ok;
ok =
FLAC__bitwriter_write_raw_uint32(bw, FLAC__SUBFRAME_TYPE_CONSTANT_BYTE_ALIGNED_MASK | (wasted_bits? 1:0), FLAC__SUBFRAME_ZERO_PAD_LEN + FLAC__SUBFRAME_TYPE_LEN + FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN) &&
(wasted_bits? FLAC__bitwriter_write_unary_unsigned(bw, wasted_bits-1) : true) &&
FLAC__bitwriter_write_raw_int32(bw, subframe->value, subframe_bps)
;
return ok;
}
FLAC__bool FLAC__subframe_add_fixed(const FLAC__Subframe_Fixed *subframe, unsigned residual_samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw)
{
unsigned i;
if(!FLAC__bitwriter_write_raw_uint32(bw, FLAC__SUBFRAME_TYPE_FIXED_BYTE_ALIGNED_MASK | (subframe->order<<1) | (wasted_bits? 1:0), FLAC__SUBFRAME_ZERO_PAD_LEN + FLAC__SUBFRAME_TYPE_LEN + FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN))
return false;
if(wasted_bits)
if(!FLAC__bitwriter_write_unary_unsigned(bw, wasted_bits-1))
return false;
for(i = 0; i < subframe->order; i++)
if(!FLAC__bitwriter_write_raw_int32(bw, subframe->warmup[i], subframe_bps))
return false;
if(!add_entropy_coding_method_(bw, &subframe->entropy_coding_method))
return false;
switch(subframe->entropy_coding_method.type) {
case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE:
case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2:
if(!add_residual_partitioned_rice_(
bw,
subframe->residual,
residual_samples,
subframe->order,
subframe->entropy_coding_method.data.partitioned_rice.contents->parameters,
subframe->entropy_coding_method.data.partitioned_rice.contents->raw_bits,
subframe->entropy_coding_method.data.partitioned_rice.order,
/*is_extended=*/subframe->entropy_coding_method.type == FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2
))
return false;
break;
default:
FLAC__ASSERT(0);
}
return true;
}
FLAC__bool FLAC__subframe_add_lpc(const FLAC__Subframe_LPC *subframe, unsigned residual_samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw)
{
unsigned i;
if(!FLAC__bitwriter_write_raw_uint32(bw, FLAC__SUBFRAME_TYPE_LPC_BYTE_ALIGNED_MASK | ((subframe->order-1)<<1) | (wasted_bits? 1:0), FLAC__SUBFRAME_ZERO_PAD_LEN + FLAC__SUBFRAME_TYPE_LEN + FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN))
return false;
if(wasted_bits)
if(!FLAC__bitwriter_write_unary_unsigned(bw, wasted_bits-1))
return false;
for(i = 0; i < subframe->order; i++)
if(!FLAC__bitwriter_write_raw_int32(bw, subframe->warmup[i], subframe_bps))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, subframe->qlp_coeff_precision-1, FLAC__SUBFRAME_LPC_QLP_COEFF_PRECISION_LEN))
return false;
if(!FLAC__bitwriter_write_raw_int32(bw, subframe->quantization_level, FLAC__SUBFRAME_LPC_QLP_SHIFT_LEN))
return false;
for(i = 0; i < subframe->order; i++)
if(!FLAC__bitwriter_write_raw_int32(bw, subframe->qlp_coeff[i], subframe->qlp_coeff_precision))
return false;
if(!add_entropy_coding_method_(bw, &subframe->entropy_coding_method))
return false;
switch(subframe->entropy_coding_method.type) {
case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE:
case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2:
if(!add_residual_partitioned_rice_(
bw,
subframe->residual,
residual_samples,
subframe->order,
subframe->entropy_coding_method.data.partitioned_rice.contents->parameters,
subframe->entropy_coding_method.data.partitioned_rice.contents->raw_bits,
subframe->entropy_coding_method.data.partitioned_rice.order,
/*is_extended=*/subframe->entropy_coding_method.type == FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2
))
return false;
break;
default:
FLAC__ASSERT(0);
}
return true;
}
FLAC__bool FLAC__subframe_add_verbatim(const FLAC__Subframe_Verbatim *subframe, unsigned samples, unsigned subframe_bps, unsigned wasted_bits, FLAC__BitWriter *bw)
{
unsigned i;
const FLAC__int32 *signal = subframe->data;
if(!FLAC__bitwriter_write_raw_uint32(bw, FLAC__SUBFRAME_TYPE_VERBATIM_BYTE_ALIGNED_MASK | (wasted_bits? 1:0), FLAC__SUBFRAME_ZERO_PAD_LEN + FLAC__SUBFRAME_TYPE_LEN + FLAC__SUBFRAME_WASTED_BITS_FLAG_LEN))
return false;
if(wasted_bits)
if(!FLAC__bitwriter_write_unary_unsigned(bw, wasted_bits-1))
return false;
for(i = 0; i < samples; i++)
if(!FLAC__bitwriter_write_raw_int32(bw, signal[i], subframe_bps))
return false;
return true;
}
FLAC__bool add_entropy_coding_method_(FLAC__BitWriter *bw, const FLAC__EntropyCodingMethod *method)
{
if(!FLAC__bitwriter_write_raw_uint32(bw, method->type, FLAC__ENTROPY_CODING_METHOD_TYPE_LEN))
return false;
switch(method->type) {
case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE:
case FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2:
if(!FLAC__bitwriter_write_raw_uint32(bw, method->data.partitioned_rice.order, FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_ORDER_LEN))
return false;
break;
default:
FLAC__ASSERT(0);
}
return true;
}
FLAC__bool add_residual_partitioned_rice_(FLAC__BitWriter *bw, const FLAC__int32 residual[], const unsigned residual_samples, const unsigned predictor_order, const unsigned rice_parameters[], const unsigned raw_bits[], const unsigned partition_order, const FLAC__bool is_extended)
{
const unsigned plen = is_extended? FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_PARAMETER_LEN : FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_PARAMETER_LEN;
const unsigned pesc = is_extended? FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE2_ESCAPE_PARAMETER : FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_ESCAPE_PARAMETER;
if(partition_order == 0) {
unsigned i;
if(raw_bits[0] == 0) {
if(!FLAC__bitwriter_write_raw_uint32(bw, rice_parameters[0], plen))
return false;
if(!FLAC__bitwriter_write_rice_signed_block(bw, residual, residual_samples, rice_parameters[0]))
return false;
}
else {
FLAC__ASSERT(rice_parameters[0] == 0);
if(!FLAC__bitwriter_write_raw_uint32(bw, pesc, plen))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, raw_bits[0], FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_RAW_LEN))
return false;
for(i = 0; i < residual_samples; i++) {
if(!FLAC__bitwriter_write_raw_int32(bw, residual[i], raw_bits[0]))
return false;
}
}
return true;
}
else {
unsigned i, j, k = 0, k_last = 0;
unsigned partition_samples;
const unsigned default_partition_samples = (residual_samples+predictor_order) >> partition_order;
for(i = 0; i < (1u<<partition_order); i++) {
partition_samples = default_partition_samples;
if(i == 0)
partition_samples -= predictor_order;
k += partition_samples;
if(raw_bits[i] == 0) {
if(!FLAC__bitwriter_write_raw_uint32(bw, rice_parameters[i], plen))
return false;
if(!FLAC__bitwriter_write_rice_signed_block(bw, residual+k_last, k-k_last, rice_parameters[i]))
return false;
}
else {
if(!FLAC__bitwriter_write_raw_uint32(bw, pesc, plen))
return false;
if(!FLAC__bitwriter_write_raw_uint32(bw, raw_bits[i], FLAC__ENTROPY_CODING_METHOD_PARTITIONED_RICE_RAW_LEN))
return false;
for(j = k_last; j < k; j++) {
if(!FLAC__bitwriter_write_raw_int32(bw, residual[j], raw_bits[i]))
return false;
}
}
k_last = k;
}
return true;
}
}
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/libFLAC/window_flac.c
+281
View File
@@ -0,0 +1,281 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2006-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
#include <math.h>
#include "../assert.h"
#include "../format.h"
#include "include/private/window.h"
#ifndef FLAC__INTEGER_ONLY_LIBRARY
void FLAC__window_bartlett(FLAC__real *window, const FLAC__int32 L)
{
const FLAC__int32 N = L - 1;
FLAC__int32 n;
if (L & 1) {
for (n = 0; n <= N/2; n++)
window[n] = 2.0f * n / (float)N;
for (; n <= N; n++)
window[n] = 2.0f - 2.0f * n / (float)N;
}
else {
for (n = 0; n <= L/2-1; n++)
window[n] = 2.0f * n / (float)N;
for (; n <= N; n++)
window[n] = 2.0f - 2.0f * n / (float)N;
}
}
void FLAC__window_bartlett_hann(FLAC__real *window, const FLAC__int32 L)
{
const FLAC__int32 N = L - 1;
FLAC__int32 n;
for (n = 0; n < L; n++)
window[n] = (FLAC__real)(0.62f - 0.48f * fabs((float)n/(float)N-0.5f) - 0.38f * cos(2.0f * M_PI * ((float)n/(float)N)));
}
void FLAC__window_blackman(FLAC__real *window, const FLAC__int32 L)
{
const FLAC__int32 N = L - 1;
FLAC__int32 n;
for (n = 0; n < L; n++)
window[n] = (FLAC__real)(0.42f - 0.5f * cos(2.0f * M_PI * n / N) + 0.08f * cos(4.0f * M_PI * n / N));
}
/* 4-term -92dB side-lobe */
void FLAC__window_blackman_harris_4term_92db_sidelobe(FLAC__real *window, const FLAC__int32 L)
{
const FLAC__int32 N = L - 1;
FLAC__int32 n;
for (n = 0; n <= N; n++)
window[n] = (FLAC__real)(0.35875f - 0.48829f * cos(2.0f * M_PI * n / N) + 0.14128f * cos(4.0f * M_PI * n / N) - 0.01168f * cos(6.0f * M_PI * n / N));
}
void FLAC__window_connes(FLAC__real *window, const FLAC__int32 L)
{
const FLAC__int32 N = L - 1;
const double N2 = (double)N / 2.;
FLAC__int32 n;
for (n = 0; n <= N; n++) {
double k = ((double)n - N2) / N2;
k = 1.0f - k * k;
window[n] = (FLAC__real)(k * k);
}
}
void FLAC__window_flattop(FLAC__real *window, const FLAC__int32 L)
{
const FLAC__int32 N = L - 1;
FLAC__int32 n;
for (n = 0; n < L; n++)
window[n] = (FLAC__real)(1.0f - 1.93f * cos(2.0f * M_PI * n / N) + 1.29f * cos(4.0f * M_PI * n / N) - 0.388f * cos(6.0f * M_PI * n / N) + 0.0322f * cos(8.0f * M_PI * n / N));
}
void FLAC__window_gauss(FLAC__real *window, const FLAC__int32 L, const FLAC__real stddev)
{
const FLAC__int32 N = L - 1;
const double N2 = (double)N / 2.;
FLAC__int32 n;
for (n = 0; n <= N; n++) {
const double k = ((double)n - N2) / (stddev * N2);
window[n] = (FLAC__real)exp(-0.5f * k * k);
}
}
void FLAC__window_hamming(FLAC__real *window, const FLAC__int32 L)
{
const FLAC__int32 N = L - 1;
FLAC__int32 n;
for (n = 0; n < L; n++)
window[n] = (FLAC__real)(0.54f - 0.46f * cos(2.0f * M_PI * n / N));
}
void FLAC__window_hann(FLAC__real *window, const FLAC__int32 L)
{
const FLAC__int32 N = L - 1;
FLAC__int32 n;
for (n = 0; n < L; n++)
window[n] = (FLAC__real)(0.5f - 0.5f * cos(2.0f * M_PI * n / N));
}
void FLAC__window_kaiser_bessel(FLAC__real *window, const FLAC__int32 L)
{
const FLAC__int32 N = L - 1;
FLAC__int32 n;
for (n = 0; n < L; n++)
window[n] = (FLAC__real)(0.402f - 0.498f * cos(2.0f * M_PI * n / N) + 0.098f * cos(4.0f * M_PI * n / N) - 0.001f * cos(6.0f * M_PI * n / N));
}
void FLAC__window_nuttall(FLAC__real *window, const FLAC__int32 L)
{
const FLAC__int32 N = L - 1;
FLAC__int32 n;
for (n = 0; n < L; n++)
window[n] = (FLAC__real)(0.3635819f - 0.4891775f*cos(2.0f*M_PI*n/N) + 0.1365995f*cos(4.0f*M_PI*n/N) - 0.0106411f*cos(6.0f*M_PI*n/N));
}
void FLAC__window_rectangle(FLAC__real *window, const FLAC__int32 L)
{
FLAC__int32 n;
for (n = 0; n < L; n++)
window[n] = 1.0f;
}
void FLAC__window_triangle(FLAC__real *window, const FLAC__int32 L)
{
FLAC__int32 n;
if (L & 1) {
for (n = 1; n <= (L+1)/2; n++)
window[n-1] = 2.0f * n / ((float)L + 1.0f);
for (; n <= L; n++)
window[n-1] = (float)(2 * (L - n + 1)) / ((float)L + 1.0f);
}
else {
for (n = 1; n <= L/2; n++)
window[n-1] = 2.0f * n / ((float)L + 1.0f);
for (; n <= L; n++)
window[n-1] = (float)(2 * (L - n + 1)) / ((float)L + 1.0f);
}
}
void FLAC__window_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p)
{
if (p <= 0.0)
FLAC__window_rectangle(window, L);
else if (p >= 1.0)
FLAC__window_hann(window, L);
else {
const FLAC__int32 Np = (FLAC__int32)(p / 2.0f * L) - 1;
FLAC__int32 n;
/* start with rectangle... */
FLAC__window_rectangle(window, L);
/* ...replace ends with hann */
if (Np > 0) {
for (n = 0; n <= Np; n++) {
window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * n / Np));
window[L-Np-1+n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * (n+Np) / Np));
}
}
}
}
void FLAC__window_partial_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p, const FLAC__real start, const FLAC__real end)
{
const FLAC__int32 start_n = (FLAC__int32)(start * L);
const FLAC__int32 end_n = (FLAC__int32)(end * L);
const FLAC__int32 N = end_n - start_n;
FLAC__int32 Np, n, i;
if (p <= 0.0f)
FLAC__window_partial_tukey(window, L, 0.05f, start, end);
else if (p >= 1.0f)
FLAC__window_partial_tukey(window, L, 0.95f, start, end);
else {
Np = (FLAC__int32)(p / 2.0f * N);
for (n = 0; n < start_n && n < L; n++)
window[n] = 0.0f;
for (i = 1; n < (start_n+Np) && n < L; n++, i++)
window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Np));
for (; n < (end_n-Np) && n < L; n++)
window[n] = 1.0f;
for (i = Np; n < end_n && n < L; n++, i--)
window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Np));
for (; n < L; n++)
window[n] = 0.0f;
}
}
void FLAC__window_punchout_tukey(FLAC__real *window, const FLAC__int32 L, const FLAC__real p, const FLAC__real start, const FLAC__real end)
{
const FLAC__int32 start_n = (FLAC__int32)(start * L);
const FLAC__int32 end_n = (FLAC__int32)(end * L);
FLAC__int32 Ns, Ne, n, i;
if (p <= 0.0f)
FLAC__window_punchout_tukey(window, L, 0.05f, start, end);
else if (p >= 1.0f)
FLAC__window_punchout_tukey(window, L, 0.95f, start, end);
else {
Ns = (FLAC__int32)(p / 2.0f * start_n);
Ne = (FLAC__int32)(p / 2.0f * (L - end_n));
for (n = 0, i = 1; n < Ns && n < L; n++, i++)
window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Ns));
for (; n < start_n-Ns && n < L; n++)
window[n] = 1.0f;
for (i = Ns; n < start_n && n < L; n++, i--)
window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Ns));
for (; n < end_n && n < L; n++)
window[n] = 0.0f;
for (i = 1; n < end_n+Ne && n < L; n++, i++)
window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Ne));
for (; n < L - (Ne) && n < L; n++)
window[n] = 1.0f;
for (i = Ne; n < L; n++, i--)
window[n] = (FLAC__real)(0.5f - 0.5f * cos(M_PI * i / Ne));
}
}
void FLAC__window_welch(FLAC__real *window, const FLAC__int32 L)
{
const FLAC__int32 N = L - 1;
const double N2 = (double)N / 2.;
FLAC__int32 n;
for (n = 0; n <= N; n++) {
const double k = ((double)n - N2) / N2;
window[n] = (FLAC__real)(1.0f - k * k);
}
}
#endif /* !defined FLAC__INTEGER_ONLY_LIBRARY */
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/metadata.h
+2182
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/ordinals.h
+86
View File
@@ -0,0 +1,86 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2000-2009 Josh Coalson
* Copyright (C) 2011-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef FLAC__ORDINALS_H
#define FLAC__ORDINALS_H
#if defined(_MSC_VER) && _MSC_VER < 1600
/* Microsoft Visual Studio earlier than the 2010 version did not provide
* the 1999 ISO C Standard header file <stdint.h>.
*/
typedef __int8 FLAC__int8;
typedef unsigned __int8 FLAC__uint8;
typedef __int16 FLAC__int16;
typedef __int32 FLAC__int32;
typedef __int64 FLAC__int64;
typedef unsigned __int16 FLAC__uint16;
typedef unsigned __int32 FLAC__uint32;
typedef unsigned __int64 FLAC__uint64;
#else
/* For MSVC 2010 and everything else which provides <stdint.h>. */
#include <stdint.h>
typedef int8_t FLAC__int8;
typedef uint8_t FLAC__uint8;
typedef int16_t FLAC__int16;
typedef int32_t FLAC__int32;
typedef int64_t FLAC__int64;
typedef uint16_t FLAC__uint16;
typedef uint32_t FLAC__uint32;
typedef uint64_t FLAC__uint64;
#endif
typedef int FLAC__bool;
typedef FLAC__uint8 FLAC__byte;
#ifdef true
#undef true
#endif
#ifdef false
#undef false
#endif
#ifndef __cplusplus
#define true 1
#define false 0
#endif
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/stream_decoder.h
+1560
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/stream_encoder.h
+1790
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/flac/win_utf8_io.h
+69
View File
@@ -0,0 +1,69 @@
/* libFLAC - Free Lossless Audio Codec library
* Copyright (C) 2013-2014 Xiph.Org Foundation
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* - Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
*
* - Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* - Neither the name of the Xiph.org Foundation nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
* LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
* A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR
* CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
* EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
* PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
* LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
* NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
* SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifdef _WIN32
#ifndef flac__win_utf8_io_h
#define flac__win_utf8_io_h
#ifdef __cplusplus
extern "C" {
#endif
#include <stdio.h>
#include <sys/stat.h>
#include <stdarg.h>
#include <windows.h>
int get_utf8_argv(int *argc, char ***argv);
int printf_utf8(const char *format, ...);
int fprintf_utf8(FILE *stream, const char *format, ...);
int vfprintf_utf8(FILE *stream, const char *format, va_list argptr);
FILE *fopen_utf8(const char *filename, const char *mode);
int stat_utf8(const char *path, struct stat *buffer);
int _stat64_utf8(const char *path, struct __stat64 *buffer);
int chmod_utf8(const char *filename, int pmode);
int utime_utf8(const char *filename, struct utimbuf *times);
int unlink_utf8(const char *filename);
int rename_utf8(const char *oldname, const char *newname);
size_t strlen_utf8(const char *str);
int win_get_console_width(void);
int print_console(FILE *stream, const wchar_t *text, size_t len);
HANDLE WINAPI CreateFile_utf8(const char *lpFileName, DWORD dwDesiredAccess, DWORD dwShareMode, LPSECURITY_ATTRIBUTES lpSecurityAttributes, DWORD dwCreationDisposition, DWORD dwFlagsAndAttributes, HANDLE hTemplateFile);
#ifdef __cplusplus
} /* extern "C" */
#endif
#endif
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_AiffAudioFormat.cpp
+954
View File
@@ -0,0 +1,954 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
static const char* const aiffFormatName = "AIFF file";
//==============================================================================
const char* const AiffAudioFormat::appleOneShot = "apple one shot";
const char* const AiffAudioFormat::appleRootSet = "apple root set";
const char* const AiffAudioFormat::appleRootNote = "apple root note";
const char* const AiffAudioFormat::appleBeats = "apple beats";
const char* const AiffAudioFormat::appleDenominator = "apple denominator";
const char* const AiffAudioFormat::appleNumerator = "apple numerator";
const char* const AiffAudioFormat::appleTag = "apple tag";
const char* const AiffAudioFormat::appleKey = "apple key";
//==============================================================================
namespace AiffFileHelpers
{
inline int chunkName (const char* name) noexcept { return (int) ByteOrder::littleEndianInt (name); }
#if JUCE_MSVC
#pragma pack (push, 1)
#endif
//==============================================================================
struct InstChunk
{
struct Loop
{
uint16 type; // these are different in AIFF and WAV
uint16 startIdentifier;
uint16 endIdentifier;
} JUCE_PACKED;
int8 baseNote;
int8 detune;
int8 lowNote;
int8 highNote;
int8 lowVelocity;
int8 highVelocity;
int16 gain;
Loop sustainLoop;
Loop releaseLoop;
void copyTo (StringPairArray& values) const
{
values.set ("MidiUnityNote", String (baseNote));
values.set ("Detune", String (detune));
values.set ("LowNote", String (lowNote));
values.set ("HighNote", String (highNote));
values.set ("LowVelocity", String (lowVelocity));
values.set ("HighVelocity", String (highVelocity));
values.set ("Gain", String ((int16) ByteOrder::swapIfLittleEndian ((uint16) gain)));
values.set ("NumSampleLoops", String (2)); // always 2 with AIFF, WAV can have more
values.set ("Loop0Type", String (ByteOrder::swapIfLittleEndian (sustainLoop.type)));
values.set ("Loop0StartIdentifier", String (ByteOrder::swapIfLittleEndian (sustainLoop.startIdentifier)));
values.set ("Loop0EndIdentifier", String (ByteOrder::swapIfLittleEndian (sustainLoop.endIdentifier)));
values.set ("Loop1Type", String (ByteOrder::swapIfLittleEndian (releaseLoop.type)));
values.set ("Loop1StartIdentifier", String (ByteOrder::swapIfLittleEndian (releaseLoop.startIdentifier)));
values.set ("Loop1EndIdentifier", String (ByteOrder::swapIfLittleEndian (releaseLoop.endIdentifier)));
}
static uint16 getValue16 (const StringPairArray& values, const char* name, const char* def)
{
return ByteOrder::swapIfLittleEndian ((uint16) values.getValue (name, def).getIntValue());
}
static int8 getValue8 (const StringPairArray& values, const char* name, const char* def)
{
return (int8) values.getValue (name, def).getIntValue();
}
static void create (MemoryBlock& block, const StringPairArray& values)
{
if (values.getAllKeys().contains ("MidiUnityNote", true))
{
block.setSize ((sizeof (InstChunk) + 3) & ~(size_t) 3, true);
InstChunk& inst = *static_cast<InstChunk*> (block.getData());
inst.baseNote = getValue8 (values, "MidiUnityNote", "60");
inst.detune = getValue8 (values, "Detune", "0");
inst.lowNote = getValue8 (values, "LowNote", "0");
inst.highNote = getValue8 (values, "HighNote", "127");
inst.lowVelocity = getValue8 (values, "LowVelocity", "1");
inst.highVelocity = getValue8 (values, "HighVelocity", "127");
inst.gain = (int16) getValue16 (values, "Gain", "0");
inst.sustainLoop.type = getValue16 (values, "Loop0Type", "0");
inst.sustainLoop.startIdentifier = getValue16 (values, "Loop0StartIdentifier", "0");
inst.sustainLoop.endIdentifier = getValue16 (values, "Loop0EndIdentifier", "0");
inst.releaseLoop.type = getValue16 (values, "Loop1Type", "0");
inst.releaseLoop.startIdentifier = getValue16 (values, "Loop1StartIdentifier", "0");
inst.releaseLoop.endIdentifier = getValue16 (values, "Loop1EndIdentifier", "0");
}
}
} JUCE_PACKED;
//==============================================================================
struct BASCChunk
{
enum Key
{
minor = 1,
major = 2,
neither = 3,
both = 4
};
BASCChunk (InputStream& input)
{
zerostruct (*this);
flags = (uint32) input.readIntBigEndian();
numBeats = (uint32) input.readIntBigEndian();
rootNote = (uint16) input.readShortBigEndian();
key = (uint16) input.readShortBigEndian();
timeSigNum = (uint16) input.readShortBigEndian();
timeSigDen = (uint16) input.readShortBigEndian();
oneShot = (uint16) input.readShortBigEndian();
input.read (unknown, sizeof (unknown));
}
void addToMetadata (StringPairArray& metadata) const
{
const bool rootNoteSet = rootNote != 0;
setBoolFlag (metadata, AiffAudioFormat::appleOneShot, oneShot == 2);
setBoolFlag (metadata, AiffAudioFormat::appleRootSet, rootNoteSet);
if (rootNoteSet)
metadata.set (AiffAudioFormat::appleRootNote, String (rootNote));
metadata.set (AiffAudioFormat::appleBeats, String (numBeats));
metadata.set (AiffAudioFormat::appleDenominator, String (timeSigDen));
metadata.set (AiffAudioFormat::appleNumerator, String (timeSigNum));
const char* keyString = nullptr;
switch (key)
{
case minor: keyString = "major"; break;
case major: keyString = "major"; break;
case neither: keyString = "neither"; break;
case both: keyString = "both"; break;
}
if (keyString != nullptr)
metadata.set (AiffAudioFormat::appleKey, keyString);
}
void setBoolFlag (StringPairArray& values, const char* name, bool shouldBeSet) const
{
values.set (name, shouldBeSet ? "1" : "0");
}
uint32 flags;
uint32 numBeats;
uint16 rootNote;
uint16 key;
uint16 timeSigNum;
uint16 timeSigDen;
uint16 oneShot;
uint8 unknown[66];
} JUCE_PACKED;
#if JUCE_MSVC
#pragma pack (pop)
#endif
//==============================================================================
static String readCATEChunk (InputStream& input, const uint32 length)
{
MemoryBlock mb;
input.skipNextBytes (4);
input.readIntoMemoryBlock (mb, (ssize_t) length - 4);
static const char* appleGenres[] =
{
"Rock/Blues",
"Electronic/Dance",
"Jazz",
"Urban",
"World/Ethnic",
"Cinematic/New Age",
"Orchestral",
"Country/Folk",
"Experimental",
"Other Genre",
nullptr
};
const StringArray genres (appleGenres);
StringArray tagsArray;
int bytesLeft = (int) mb.getSize();
const char* data = static_cast<const char*> (mb.getData());
while (bytesLeft > 0)
{
const String tag (CharPointer_UTF8 (data),
CharPointer_UTF8 (data + bytesLeft));
if (tag.isNotEmpty())
tagsArray.add (data);
const int numBytesInTag = genres.contains (tag) ? 118 : 50;
data += numBytesInTag;
bytesLeft -= numBytesInTag;
}
return tagsArray.joinIntoString (";");
}
//==============================================================================
namespace MarkChunk
{
static bool metaDataContainsZeroIdentifiers (const StringPairArray& values)
{
// (zero cue identifiers are valid for WAV but not for AIFF)
const String cueString ("Cue");
const String noteString ("CueNote");
const String identifierString ("Identifier");
const StringArray& keys = values.getAllKeys();
for (int i = 0; i < keys.size(); ++i)
{
const String key (keys[i]);
if (key.startsWith (noteString))
continue; // zero identifier IS valid in a COMT chunk
if (key.startsWith (cueString) && key.contains (identifierString))
{
const int value = values.getValue (key, "-1").getIntValue();
if (value == 0)
return true;
}
}
return false;
}
static void create (MemoryBlock& block, const StringPairArray& values)
{
const int numCues = values.getValue ("NumCuePoints", "0").getIntValue();
if (numCues > 0)
{
MemoryOutputStream out (block, false);
out.writeShortBigEndian ((short) numCues);
const int numCueLabels = values.getValue ("NumCueLabels", "0").getIntValue();
const int idOffset = metaDataContainsZeroIdentifiers (values) ? 1 : 0; // can't have zero IDs in AIFF
#if JUCE_DEBUG
Array<int> identifiers;
#endif
for (int i = 0; i < numCues; ++i)
{
const String prefixCue ("Cue" + String (i));
const int identifier = idOffset + values.getValue (prefixCue + "Identifier", "1").getIntValue();
#if JUCE_DEBUG
jassert (! identifiers.contains (identifier));
identifiers.add (identifier);
#endif
const int offset = values.getValue (prefixCue + "Offset", "0").getIntValue();
String label ("CueLabel" + String (i));
for (int labelIndex = 0; labelIndex < numCueLabels; ++labelIndex)
{
const String prefixLabel ("CueLabel" + String (labelIndex));
const int labelIdentifier = idOffset + values.getValue (prefixLabel + "Identifier", "1").getIntValue();
if (labelIdentifier == identifier)
{
label = values.getValue (prefixLabel + "Text", label);
break;
}
}
out.writeShortBigEndian ((short) identifier);
out.writeIntBigEndian (offset);
const size_t labelLength = jmin ((size_t) 254, label.getNumBytesAsUTF8()); // seems to need null terminator even though it's a pstring
out.writeByte ((char) labelLength + 1);
out.write (label.toUTF8(), labelLength);
out.writeByte (0);
}
if ((out.getDataSize() & 1) != 0)
out.writeByte (0);
}
}
}
//==============================================================================
namespace COMTChunk
{
static void create (MemoryBlock& block, const StringPairArray& values)
{
const int numNotes = values.getValue ("NumCueNotes", "0").getIntValue();
if (numNotes > 0)
{
MemoryOutputStream out (block, false);
out.writeShortBigEndian ((short) numNotes);
for (int i = 0; i < numNotes; ++i)
{
const String prefix ("CueNote" + String (i));
out.writeIntBigEndian (values.getValue (prefix + "TimeStamp", "0").getIntValue());
out.writeShortBigEndian ((short) values.getValue (prefix + "Identifier", "0").getIntValue());
const String comment (values.getValue (prefix + "Text", String()));
const size_t commentLength = jmin (comment.getNumBytesAsUTF8(), (size_t) 65534);
out.writeShortBigEndian ((short) commentLength + 1);
out.write (comment.toUTF8(), commentLength);
out.writeByte (0);
if ((out.getDataSize() & 1) != 0)
out.writeByte (0);
}
}
}
}
}
//==============================================================================
class AiffAudioFormatReader : public AudioFormatReader
{
public:
AiffAudioFormatReader (InputStream* in)
: AudioFormatReader (in, aiffFormatName)
{
using namespace AiffFileHelpers;
if (input->readInt() == chunkName ("FORM"))
{
const int len = input->readIntBigEndian();
const int64 end = input->getPosition() + len;
const int nextType = input->readInt();
if (nextType == chunkName ("AIFF") || nextType == chunkName ("AIFC"))
{
bool hasGotVer = false;
bool hasGotData = false;
bool hasGotType = false;
while (input->getPosition() < end)
{
const int type = input->readInt();
const uint32 length = (uint32) input->readIntBigEndian();
const int64 chunkEnd = input->getPosition() + length;
if (type == chunkName ("FVER"))
{
hasGotVer = true;
const int ver = input->readIntBigEndian();
if (ver != 0 && ver != (int) 0xa2805140)
break;
}
else if (type == chunkName ("COMM"))
{
hasGotType = true;
numChannels = (unsigned int) input->readShortBigEndian();
lengthInSamples = input->readIntBigEndian();
bitsPerSample = (unsigned int) input->readShortBigEndian();
bytesPerFrame = (int) ((numChannels * bitsPerSample) >> 3);
unsigned char sampleRateBytes[10];
input->read (sampleRateBytes, 10);
const int byte0 = sampleRateBytes[0];
if ((byte0 & 0x80) != 0
|| byte0 <= 0x3F || byte0 > 0x40
|| (byte0 == 0x40 && sampleRateBytes[1] > 0x1C))
break;
unsigned int sampRate = ByteOrder::bigEndianInt (sampleRateBytes + 2);
sampRate >>= (16414 - ByteOrder::bigEndianShort (sampleRateBytes));
sampleRate = (int) sampRate;
if (length <= 18)
{
// some types don't have a chunk large enough to include a compression
// type, so assume it's just big-endian pcm
littleEndian = false;
}
else
{
const int compType = input->readInt();
if (compType == chunkName ("NONE") || compType == chunkName ("twos"))
{
littleEndian = false;
}
else if (compType == chunkName ("sowt"))
{
littleEndian = true;
}
else if (compType == chunkName ("fl32") || compType == chunkName ("FL32"))
{
littleEndian = false;
usesFloatingPointData = true;
}
else
{
sampleRate = 0;
break;
}
}
}
else if (type == chunkName ("SSND"))
{
hasGotData = true;
const int offset = input->readIntBigEndian();
dataChunkStart = input->getPosition() + 4 + offset;
lengthInSamples = (bytesPerFrame > 0) ? jmin (lengthInSamples, ((int64) length) / (int64) bytesPerFrame) : 0;
}
else if (type == chunkName ("MARK"))
{
const uint16 numCues = (uint16) input->readShortBigEndian();
// these two are always the same for AIFF-read files
metadataValues.set ("NumCuePoints", String (numCues));
metadataValues.set ("NumCueLabels", String (numCues));
for (uint16 i = 0; i < numCues; ++i)
{
uint16 identifier = (uint16) input->readShortBigEndian();
uint32 offset = (uint32) input->readIntBigEndian();
uint8 stringLength = (uint8) input->readByte();
MemoryBlock textBlock;
input->readIntoMemoryBlock (textBlock, stringLength);
// if the stringLength is even then read one more byte as the
// string needs to be an even number of bytes INCLUDING the
// leading length character in the pascal string
if ((stringLength & 1) == 0)
input->readByte();
const String prefixCue ("Cue" + String (i));
metadataValues.set (prefixCue + "Identifier", String (identifier));
metadataValues.set (prefixCue + "Offset", String (offset));
const String prefixLabel ("CueLabel" + String (i));
metadataValues.set (prefixLabel + "Identifier", String (identifier));
metadataValues.set (prefixLabel + "Text", textBlock.toString());
}
}
else if (type == chunkName ("COMT"))
{
const uint16 numNotes = (uint16) input->readShortBigEndian();
metadataValues.set ("NumCueNotes", String (numNotes));
for (uint16 i = 0; i < numNotes; ++i)
{
uint32 timestamp = (uint32) input->readIntBigEndian();
uint16 identifier = (uint16) input->readShortBigEndian(); // may be zero in this case
uint16 stringLength = (uint16) input->readShortBigEndian();
MemoryBlock textBlock;
input->readIntoMemoryBlock (textBlock, stringLength + (stringLength & 1));
const String prefix ("CueNote" + String (i));
metadataValues.set (prefix + "TimeStamp", String (timestamp));
metadataValues.set (prefix + "Identifier", String (identifier));
metadataValues.set (prefix + "Text", textBlock.toString());
}
}
else if (type == chunkName ("INST"))
{
HeapBlock <InstChunk> inst;
inst.calloc (jmax ((size_t) length + 1, sizeof (InstChunk)), 1);
input->read (inst, (int) length);
inst->copyTo (metadataValues);
}
else if (type == chunkName ("basc"))
{
AiffFileHelpers::BASCChunk (*input).addToMetadata (metadataValues);
}
else if (type == chunkName ("cate"))
{
metadataValues.set (AiffAudioFormat::appleTag,
AiffFileHelpers::readCATEChunk (*input, length));;
}
else if ((hasGotVer && hasGotData && hasGotType)
|| chunkEnd < input->getPosition()
|| input->isExhausted())
{
break;
}
input->setPosition (chunkEnd + (chunkEnd & 1)); // (chunks should be aligned to an even byte address)
}
}
}
if (metadataValues.size() > 0)
metadataValues.set ("MetaDataSource", "AIFF");
}
//==============================================================================
bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
int64 startSampleInFile, int numSamples) override
{
clearSamplesBeyondAvailableLength (destSamples, numDestChannels, startOffsetInDestBuffer,
startSampleInFile, numSamples, lengthInSamples);
if (numSamples <= 0)
return true;
input->setPosition (dataChunkStart + startSampleInFile * bytesPerFrame);
while (numSamples > 0)
{
const int tempBufSize = 480 * 3 * 4; // (keep this a multiple of 3)
char tempBuffer [tempBufSize];
const int numThisTime = jmin (tempBufSize / bytesPerFrame, numSamples);
const int bytesRead = input->read (tempBuffer, numThisTime * bytesPerFrame);
if (bytesRead < numThisTime * bytesPerFrame)
{
jassert (bytesRead >= 0);
zeromem (tempBuffer + bytesRead, (size_t) (numThisTime * bytesPerFrame - bytesRead));
}
if (littleEndian)
copySampleData<AudioData::LittleEndian> (bitsPerSample, usesFloatingPointData,
destSamples, startOffsetInDestBuffer, numDestChannels,
tempBuffer, (int) numChannels, numThisTime);
else
copySampleData<AudioData::BigEndian> (bitsPerSample, usesFloatingPointData,
destSamples, startOffsetInDestBuffer, numDestChannels,
tempBuffer, (int) numChannels, numThisTime);
startOffsetInDestBuffer += numThisTime;
numSamples -= numThisTime;
}
return true;
}
template <typename Endianness>
static void copySampleData (unsigned int bitsPerSample, const bool usesFloatingPointData,
int* const* destSamples, int startOffsetInDestBuffer, int numDestChannels,
const void* sourceData, int numChannels, int numSamples) noexcept
{
switch (bitsPerSample)
{
case 8: ReadHelper<AudioData::Int32, AudioData::Int8, Endianness>::read (destSamples, startOffsetInDestBuffer, numDestChannels, sourceData, numChannels, numSamples); break;
case 16: ReadHelper<AudioData::Int32, AudioData::Int16, Endianness>::read (destSamples, startOffsetInDestBuffer, numDestChannels, sourceData, numChannels, numSamples); break;
case 24: ReadHelper<AudioData::Int32, AudioData::Int24, Endianness>::read (destSamples, startOffsetInDestBuffer, numDestChannels, sourceData, numChannels, numSamples); break;
case 32: if (usesFloatingPointData) ReadHelper<AudioData::Float32, AudioData::Float32, Endianness>::read (destSamples, startOffsetInDestBuffer, numDestChannels, sourceData, numChannels, numSamples);
else ReadHelper<AudioData::Int32, AudioData::Int32, Endianness>::read (destSamples, startOffsetInDestBuffer, numDestChannels, sourceData, numChannels, numSamples); break;
default: jassertfalse; break;
}
}
int bytesPerFrame;
int64 dataChunkStart;
bool littleEndian;
private:
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AiffAudioFormatReader)
};
//==============================================================================
class AiffAudioFormatWriter : public AudioFormatWriter
{
public:
AiffAudioFormatWriter (OutputStream* out, double rate,
unsigned int numChans, unsigned int bits,
const StringPairArray& metadataValues)
: AudioFormatWriter (out, aiffFormatName, rate, numChans, bits),
lengthInSamples (0),
bytesWritten (0),
writeFailed (false)
{
using namespace AiffFileHelpers;
if (metadataValues.size() > 0)
{
// The meta data should have been santised for the AIFF format.
// If it was originally sourced from a WAV file the MetaDataSource
// key should be removed (or set to "AIFF") once this has been done
jassert (metadataValues.getValue ("MetaDataSource", "None") != "WAV");
MarkChunk::create (markChunk, metadataValues);
COMTChunk::create (comtChunk, metadataValues);
InstChunk::create (instChunk, metadataValues);
}
headerPosition = out->getPosition();
writeHeader();
}
~AiffAudioFormatWriter()
{
if ((bytesWritten & 1) != 0)
output->writeByte (0);
writeHeader();
}
//==============================================================================
bool write (const int** data, int numSamples) override
{
jassert (data != nullptr && *data != nullptr); // the input must contain at least one channel!
if (writeFailed)
return false;
const size_t bytes = (size_t) numSamples * numChannels * bitsPerSample / 8;
tempBlock.ensureSize ((size_t) bytes, false);
switch (bitsPerSample)
{
case 8: WriteHelper<AudioData::Int8, AudioData::Int32, AudioData::BigEndian>::write (tempBlock.getData(), (int) numChannels, data, numSamples); break;
case 16: WriteHelper<AudioData::Int16, AudioData::Int32, AudioData::BigEndian>::write (tempBlock.getData(), (int) numChannels, data, numSamples); break;
case 24: WriteHelper<AudioData::Int24, AudioData::Int32, AudioData::BigEndian>::write (tempBlock.getData(), (int) numChannels, data, numSamples); break;
case 32: WriteHelper<AudioData::Int32, AudioData::Int32, AudioData::BigEndian>::write (tempBlock.getData(), (int) numChannels, data, numSamples); break;
default: jassertfalse; break;
}
if (bytesWritten + bytes >= (size_t) 0xfff00000
|| ! output->write (tempBlock.getData(), bytes))
{
// failed to write to disk, so let's try writing the header.
// If it's just run out of disk space, then if it does manage
// to write the header, we'll still have a useable file..
writeHeader();
writeFailed = true;
return false;
}
else
{
bytesWritten += bytes;
lengthInSamples += (uint64) numSamples;
return true;
}
}
private:
MemoryBlock tempBlock, markChunk, comtChunk, instChunk;
uint64 lengthInSamples, bytesWritten;
int64 headerPosition;
bool writeFailed;
void writeHeader()
{
using namespace AiffFileHelpers;
const bool couldSeekOk = output->setPosition (headerPosition);
(void) couldSeekOk;
// if this fails, you've given it an output stream that can't seek! It needs
// to be able to seek back to write the header
jassert (couldSeekOk);
const int headerLen = (int) (54 + (markChunk.getSize() > 0 ? markChunk.getSize() + 8 : 0)
+ (comtChunk.getSize() > 0 ? comtChunk.getSize() + 8 : 0)
+ (instChunk.getSize() > 0 ? instChunk.getSize() + 8 : 0));
int audioBytes = (int) (lengthInSamples * ((bitsPerSample * numChannels) / 8));
audioBytes += (audioBytes & 1);
output->writeInt (chunkName ("FORM"));
output->writeIntBigEndian (headerLen + audioBytes - 8);
output->writeInt (chunkName ("AIFF"));
output->writeInt (chunkName ("COMM"));
output->writeIntBigEndian (18);
output->writeShortBigEndian ((short) numChannels);
output->writeIntBigEndian ((int) lengthInSamples);
output->writeShortBigEndian ((short) bitsPerSample);
uint8 sampleRateBytes[10] = { 0 };
if (sampleRate <= 1)
{
sampleRateBytes[0] = 0x3f;
sampleRateBytes[1] = 0xff;
sampleRateBytes[2] = 0x80;
}
else
{
int mask = 0x40000000;
sampleRateBytes[0] = 0x40;
if (sampleRate >= mask)
{
jassertfalse;
sampleRateBytes[1] = 0x1d;
}
else
{
int n = (int) sampleRate;
int i;
for (i = 0; i <= 32 ; ++i)
{
if ((n & mask) != 0)
break;
mask >>= 1;
}
n = n << (i + 1);
sampleRateBytes[1] = (uint8) (29 - i);
sampleRateBytes[2] = (uint8) ((n >> 24) & 0xff);
sampleRateBytes[3] = (uint8) ((n >> 16) & 0xff);
sampleRateBytes[4] = (uint8) ((n >> 8) & 0xff);
sampleRateBytes[5] = (uint8) (n & 0xff);
}
}
output->write (sampleRateBytes, 10);
if (markChunk.getSize() > 0)
{
output->writeInt (chunkName ("MARK"));
output->writeIntBigEndian ((int) markChunk.getSize());
*output << markChunk;
}
if (comtChunk.getSize() > 0)
{
output->writeInt (chunkName ("COMT"));
output->writeIntBigEndian ((int) comtChunk.getSize());
*output << comtChunk;
}
if (instChunk.getSize() > 0)
{
output->writeInt (chunkName ("INST"));
output->writeIntBigEndian ((int) instChunk.getSize());
*output << instChunk;
}
output->writeInt (chunkName ("SSND"));
output->writeIntBigEndian (audioBytes + 8);
output->writeInt (0);
output->writeInt (0);
jassert (output->getPosition() == headerLen);
}
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AiffAudioFormatWriter)
};
//==============================================================================
class MemoryMappedAiffReader : public MemoryMappedAudioFormatReader
{
public:
MemoryMappedAiffReader (const File& f, const AiffAudioFormatReader& reader)
: MemoryMappedAudioFormatReader (f, reader, reader.dataChunkStart,
reader.bytesPerFrame * reader.lengthInSamples, reader.bytesPerFrame),
littleEndian (reader.littleEndian)
{
}
bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
int64 startSampleInFile, int numSamples) override
{
clearSamplesBeyondAvailableLength (destSamples, numDestChannels, startOffsetInDestBuffer,
startSampleInFile, numSamples, lengthInSamples);
if (map == nullptr || ! mappedSection.contains (Range<int64> (startSampleInFile, startSampleInFile + numSamples)))
{
jassertfalse; // you must make sure that the window contains all the samples you're going to attempt to read.
return false;
}
if (littleEndian)
AiffAudioFormatReader::copySampleData<AudioData::LittleEndian>
(bitsPerSample, usesFloatingPointData, destSamples, startOffsetInDestBuffer,
numDestChannels, sampleToPointer (startSampleInFile), (int) numChannels, numSamples);
else
AiffAudioFormatReader::copySampleData<AudioData::BigEndian>
(bitsPerSample, usesFloatingPointData, destSamples, startOffsetInDestBuffer,
numDestChannels, sampleToPointer (startSampleInFile), (int) numChannels, numSamples);
return true;
}
void readMaxLevels (int64 startSampleInFile, int64 numSamples,
float& min0, float& max0, float& min1, float& max1)
{
if (numSamples <= 0)
{
min0 = max0 = min1 = max1 = 0;
return;
}
if (map == nullptr || ! mappedSection.contains (Range<int64> (startSampleInFile, startSampleInFile + numSamples)))
{
jassertfalse; // you must make sure that the window contains all the samples you're going to attempt to read.
min0 = max0 = min1 = max1 = 0;
return;
}
switch (bitsPerSample)
{
case 8: scanMinAndMax<AudioData::UInt8> (startSampleInFile, numSamples, min0, max0, min1, max1); break;
case 16: scanMinAndMax<AudioData::Int16> (startSampleInFile, numSamples, min0, max0, min1, max1); break;
case 24: scanMinAndMax<AudioData::Int24> (startSampleInFile, numSamples, min0, max0, min1, max1); break;
case 32: if (usesFloatingPointData) scanMinAndMax<AudioData::Float32> (startSampleInFile, numSamples, min0, max0, min1, max1);
else scanMinAndMax<AudioData::Int32> (startSampleInFile, numSamples, min0, max0, min1, max1); break;
default: jassertfalse; break;
}
}
private:
const bool littleEndian;
template <typename SampleType>
void scanMinAndMax (int64 startSampleInFile, int64 numSamples,
float& min0, float& max0, float& min1, float& max1) const noexcept
{
scanMinAndMax2<SampleType> (0, startSampleInFile, numSamples, min0, max0);
if (numChannels > 1)
scanMinAndMax2<SampleType> (1, startSampleInFile, numSamples, min1, max1);
else
min1 = max1 = 0;
}
template <typename SampleType>
void scanMinAndMax2 (int channel, int64 startSampleInFile, int64 numSamples, float& mn, float& mx) const noexcept
{
if (littleEndian)
scanMinAndMaxInterleaved<SampleType, AudioData::LittleEndian> (channel, startSampleInFile, numSamples, mn, mx);
else
scanMinAndMaxInterleaved<SampleType, AudioData::BigEndian> (channel, startSampleInFile, numSamples, mn, mx);
}
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (MemoryMappedAiffReader)
};
//==============================================================================
AiffAudioFormat::AiffAudioFormat() : AudioFormat (aiffFormatName, ".aiff .aif")
{
}
AiffAudioFormat::~AiffAudioFormat()
{
}
Array<int> AiffAudioFormat::getPossibleSampleRates()
{
const int rates[] = { 22050, 32000, 44100, 48000, 88200, 96000, 176400, 192000, 0 };
return Array<int> (rates);
}
Array<int> AiffAudioFormat::getPossibleBitDepths()
{
const int depths[] = { 8, 16, 24, 0 };
return Array<int> (depths);
}
bool AiffAudioFormat::canDoStereo() { return true; }
bool AiffAudioFormat::canDoMono() { return true; }
#if JUCE_MAC
bool AiffAudioFormat::canHandleFile (const File& f)
{
if (AudioFormat::canHandleFile (f))
return true;
const OSType type = f.getMacOSType();
// (NB: written as hex to avoid four-char-constant warnings)
return type == 0x41494646 /* AIFF */ || type == 0x41494643 /* AIFC */
|| type == 0x61696666 /* aiff */ || type == 0x61696663 /* aifc */;
}
#endif
AudioFormatReader* AiffAudioFormat::createReaderFor (InputStream* sourceStream, const bool deleteStreamIfOpeningFails)
{
ScopedPointer <AiffAudioFormatReader> w (new AiffAudioFormatReader (sourceStream));
if (w->sampleRate > 0 && w->numChannels > 0)
return w.release();
if (! deleteStreamIfOpeningFails)
w->input = nullptr;
return nullptr;
}
MemoryMappedAudioFormatReader* AiffAudioFormat::createMemoryMappedReader (const File& file)
{
if (FileInputStream* fin = file.createInputStream())
{
AiffAudioFormatReader reader (fin);
if (reader.lengthInSamples > 0)
return new MemoryMappedAiffReader (file, reader);
}
return nullptr;
}
AudioFormatWriter* AiffAudioFormat::createWriterFor (OutputStream* out,
double sampleRate,
unsigned int numberOfChannels,
int bitsPerSample,
const StringPairArray& metadataValues,
int /*qualityOptionIndex*/)
{
if (getPossibleBitDepths().contains (bitsPerSample))
return new AiffAudioFormatWriter (out, sampleRate, numberOfChannels, (unsigned int) bitsPerSample, metadataValues);
return nullptr;
}
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_AiffAudioFormat.h
+84
View File
@@ -0,0 +1,84 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
//==============================================================================
/**
Reads and Writes AIFF format audio files.
@see AudioFormat
*/
class JUCE_API AiffAudioFormat : public AudioFormat
{
public:
//==============================================================================
/** Creates an format object. */
AiffAudioFormat();
/** Destructor. */
~AiffAudioFormat();
//==============================================================================
/** Metadata property name used when reading a aiff file with a basc chunk. */
static const char* const appleOneShot;
/** Metadata property name used when reading a aiff file with a basc chunk. */
static const char* const appleRootSet;
/** Metadata property name used when reading a aiff file with a basc chunk. */
static const char* const appleRootNote;
/** Metadata property name used when reading a aiff file with a basc chunk. */
static const char* const appleBeats;
/** Metadata property name used when reading a aiff file with a basc chunk. */
static const char* const appleDenominator;
/** Metadata property name used when reading a aiff file with a basc chunk. */
static const char* const appleNumerator;
/** Metadata property name used when reading a aiff file with a basc chunk. */
static const char* const appleTag;
/** Metadata property name used when reading a aiff file with a basc chunk. */
static const char* const appleKey;
//==============================================================================
Array<int> getPossibleSampleRates() override;
Array<int> getPossibleBitDepths() override;
bool canDoStereo() override;
bool canDoMono() override;
#if JUCE_MAC
bool canHandleFile (const File& fileToTest) override;
#endif
//==============================================================================
AudioFormatReader* createReaderFor (InputStream* sourceStream,
bool deleteStreamIfOpeningFails) override;
MemoryMappedAudioFormatReader* createMemoryMappedReader (const File&) override;
AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
double sampleRateToUse,
unsigned int numberOfChannels,
int bitsPerSample,
const StringPairArray& metadataValues,
int qualityOptionIndex) override;
private:
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR(AiffAudioFormat)
};
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_CoreAudioFormat.cpp
+528
View File
@@ -0,0 +1,528 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
#if JUCE_MAC || JUCE_IOS
//==============================================================================
namespace
{
const char* const coreAudioFormatName = "CoreAudio supported file";
StringArray findFileExtensionsForCoreAudioCodecs()
{
StringArray extensionsArray;
CFArrayRef extensions = nullptr;
UInt32 sizeOfArray = sizeof (extensions);
if (AudioFileGetGlobalInfo (kAudioFileGlobalInfo_AllExtensions, 0, 0, &sizeOfArray, &extensions) == noErr)
{
const CFIndex numValues = CFArrayGetCount (extensions);
for (CFIndex i = 0; i < numValues; ++i)
extensionsArray.add ("." + String::fromCFString ((CFStringRef) CFArrayGetValueAtIndex (extensions, i)));
CFRelease (extensions);
}
return extensionsArray;
}
}
//==============================================================================
const char* const CoreAudioFormat::midiDataBase64 = "midiDataBase64";
const char* const CoreAudioFormat::tempo = "tempo";
const char* const CoreAudioFormat::timeSig = "time signature";
const char* const CoreAudioFormat::keySig = "key signature";
//==============================================================================
struct CoreAudioFormatMetatdata
{
static uint32 chunkName (const char* const name) noexcept { return ByteOrder::bigEndianInt (name); }
//==============================================================================
struct FileHeader
{
FileHeader (InputStream& input)
{
fileType = (uint32) input.readIntBigEndian();
fileVersion = (uint16) input.readShortBigEndian();
fileFlags = (uint16) input.readShortBigEndian();
}
uint32 fileType;
uint16 fileVersion;
uint16 fileFlags;
};
//==============================================================================
struct ChunkHeader
{
ChunkHeader (InputStream& input)
{
chunkType = (uint32) input.readIntBigEndian();
chunkSize = (int64) input.readInt64BigEndian();
}
uint32 chunkType;
int64 chunkSize;
};
//==============================================================================
struct AudioDescriptionChunk
{
AudioDescriptionChunk (InputStream& input)
{
sampleRate = input.readDoubleBigEndian();
formatID = (uint32) input.readIntBigEndian();
formatFlags = (uint32) input.readIntBigEndian();
bytesPerPacket = (uint32) input.readIntBigEndian();
framesPerPacket = (uint32) input.readIntBigEndian();
channelsPerFrame = (uint32) input.readIntBigEndian();
bitsPerChannel = (uint32) input.readIntBigEndian();
}
double sampleRate;
uint32 formatID;
uint32 formatFlags;
uint32 bytesPerPacket;
uint32 framesPerPacket;
uint32 channelsPerFrame;
uint32 bitsPerChannel;
};
//==============================================================================
struct UserDefinedChunk
{
UserDefinedChunk (InputStream& input, int64 size)
{
// a user defined chunk contains 16 bytes of a UUID first
uuid[1] = input.readInt64BigEndian();
uuid[0] = input.readInt64BigEndian();
input.skipNextBytes (size - 16);
}
int64 uuid[2];
};
//==============================================================================
static StringPairArray parseMidiChunk (InputStream& input, int64 size)
{
const int64 originalPosition = input.getPosition();
MemoryBlock midiBlock;
input.readIntoMemoryBlock (midiBlock, (ssize_t) size);
MemoryInputStream midiInputStream (midiBlock, false);
StringPairArray midiMetadata;
MidiFile midiFile;
if (midiFile.readFrom (midiInputStream))
{
midiMetadata.set (CoreAudioFormat::midiDataBase64, midiBlock.toBase64Encoding());
findTempoEvents (midiFile, midiMetadata);
findTimeSigEvents (midiFile, midiMetadata);
findKeySigEvents (midiFile, midiMetadata);
}
input.setPosition (originalPosition + size);
return midiMetadata;
}
static void findTempoEvents (MidiFile& midiFile, StringPairArray& midiMetadata)
{
MidiMessageSequence tempoEvents;
midiFile.findAllTempoEvents (tempoEvents);
const int numTempoEvents = tempoEvents.getNumEvents();
MemoryOutputStream tempoSequence;
for (int i = 0; i < numTempoEvents; ++i)
{
const double tempo = getTempoFromTempoMetaEvent (tempoEvents.getEventPointer (i));
if (tempo > 0.0)
{
if (i == 0)
midiMetadata.set (CoreAudioFormat::tempo, String (tempo));
if (numTempoEvents > 1)
tempoSequence << String (tempo) << ',' << tempoEvents.getEventTime (i) << ';';
}
}
if (tempoSequence.getDataSize() > 0)
midiMetadata.set ("tempo sequence", tempoSequence.toUTF8());
}
static double getTempoFromTempoMetaEvent (MidiMessageSequence::MidiEventHolder* holder)
{
if (holder != nullptr)
{
const MidiMessage& midiMessage = holder->message;
if (midiMessage.isTempoMetaEvent())
{
const double tempoSecondsPerQuarterNote = midiMessage.getTempoSecondsPerQuarterNote();
if (tempoSecondsPerQuarterNote > 0.0)
return 60.0 / tempoSecondsPerQuarterNote;
}
}
return 0.0;
}
static void findTimeSigEvents (MidiFile& midiFile, StringPairArray& midiMetadata)
{
MidiMessageSequence timeSigEvents;
midiFile.findAllTimeSigEvents (timeSigEvents);
const int numTimeSigEvents = timeSigEvents.getNumEvents();
MemoryOutputStream timeSigSequence;
for (int i = 0; i < numTimeSigEvents; ++i)
{
int numerator, denominator;
timeSigEvents.getEventPointer(i)->message.getTimeSignatureInfo (numerator, denominator);
String timeSigString;
timeSigString << numerator << '/' << denominator;
if (i == 0)
midiMetadata.set (CoreAudioFormat::timeSig, timeSigString);
if (numTimeSigEvents > 1)
timeSigSequence << timeSigString << ',' << timeSigEvents.getEventTime (i) << ';';
}
if (timeSigSequence.getDataSize() > 0)
midiMetadata.set ("time signature sequence", timeSigSequence.toUTF8());
}
static void findKeySigEvents (MidiFile& midiFile, StringPairArray& midiMetadata)
{
MidiMessageSequence keySigEvents;
midiFile.findAllKeySigEvents (keySigEvents);
const int numKeySigEvents = keySigEvents.getNumEvents();
MemoryOutputStream keySigSequence;
for (int i = 0; i < numKeySigEvents; ++i)
{
const MidiMessage& message (keySigEvents.getEventPointer (i)->message);
const int key = jlimit (0, 14, message.getKeySignatureNumberOfSharpsOrFlats() + 7);
const bool isMajor = message.isKeySignatureMajorKey();
static const char* majorKeys[] = { "Cb", "Gb", "Db", "Ab", "Eb", "Bb", "F", "C", "G", "D", "A", "E", "B", "F#", "C#" };
static const char* minorKeys[] = { "Ab", "Eb", "Bb", "F", "C", "G", "D", "A", "E", "B", "F#", "C#", "G#", "D#", "A#" };
String keySigString (isMajor ? majorKeys[key]
: minorKeys[key]);
if (! isMajor)
keySigString << 'm';
if (i == 0)
midiMetadata.set (CoreAudioFormat::keySig, keySigString);
if (numKeySigEvents > 1)
keySigSequence << keySigString << ',' << keySigEvents.getEventTime (i) << ';';
}
if (keySigSequence.getDataSize() > 0)
midiMetadata.set ("key signature sequence", keySigSequence.toUTF8());
}
//==============================================================================
static StringPairArray parseInformationChunk (InputStream& input)
{
StringPairArray infoStrings;
const uint32 numEntries = (uint32) input.readIntBigEndian();
for (uint32 i = 0; i < numEntries; ++i)
infoStrings.set (input.readString(), input.readString());
return infoStrings;
}
//==============================================================================
static bool read (InputStream& input, StringPairArray& metadataValues)
{
const int64 originalPos = input.getPosition();
const FileHeader cafFileHeader (input);
const bool isCafFile = cafFileHeader.fileType == chunkName ("caff");
if (isCafFile)
{
while (! input.isExhausted())
{
const ChunkHeader chunkHeader (input);
if (chunkHeader.chunkType == chunkName ("desc"))
{
AudioDescriptionChunk audioDescriptionChunk (input);
}
else if (chunkHeader.chunkType == chunkName ("uuid"))
{
UserDefinedChunk userDefinedChunk (input, chunkHeader.chunkSize);
}
else if (chunkHeader.chunkType == chunkName ("data"))
{
// -1 signifies an unknown data size so the data has to be at the
// end of the file so we must have finished the header
if (chunkHeader.chunkSize == -1)
break;
input.skipNextBytes (chunkHeader.chunkSize);
}
else if (chunkHeader.chunkType == chunkName ("midi"))
{
metadataValues.addArray (parseMidiChunk (input, chunkHeader.chunkSize));
}
else if (chunkHeader.chunkType == chunkName ("info"))
{
metadataValues.addArray (parseInformationChunk (input));
}
else
{
// we aren't decoding this chunk yet so just skip over it
input.skipNextBytes (chunkHeader.chunkSize);
}
}
}
input.setPosition (originalPos);
return isCafFile;
}
};
//==============================================================================
class CoreAudioReader : public AudioFormatReader
{
public:
CoreAudioReader (InputStream* const inp)
: AudioFormatReader (inp, coreAudioFormatName),
ok (false), lastReadPosition (0)
{
usesFloatingPointData = true;
bitsPerSample = 32;
if (input != nullptr)
CoreAudioFormatMetatdata::read (*input, metadataValues);
OSStatus status = AudioFileOpenWithCallbacks (this,
&readCallback,
nullptr, // write needs to be null to avoid permisisions errors
&getSizeCallback,
nullptr, // setSize needs to be null to avoid permisisions errors
0, // AudioFileTypeID inFileTypeHint
&audioFileID);
if (status == noErr)
{
status = ExtAudioFileWrapAudioFileID (audioFileID, false, &audioFileRef);
if (status == noErr)
{
AudioStreamBasicDescription sourceAudioFormat;
UInt32 audioStreamBasicDescriptionSize = sizeof (AudioStreamBasicDescription);
ExtAudioFileGetProperty (audioFileRef,
kExtAudioFileProperty_FileDataFormat,
&audioStreamBasicDescriptionSize,
&sourceAudioFormat);
numChannels = sourceAudioFormat.mChannelsPerFrame;
sampleRate = sourceAudioFormat.mSampleRate;
UInt32 sizeOfLengthProperty = sizeof (int64);
ExtAudioFileGetProperty (audioFileRef,
kExtAudioFileProperty_FileLengthFrames,
&sizeOfLengthProperty,
&lengthInSamples);
destinationAudioFormat.mSampleRate = sampleRate;
destinationAudioFormat.mFormatID = kAudioFormatLinearPCM;
destinationAudioFormat.mFormatFlags = kLinearPCMFormatFlagIsFloat | kLinearPCMFormatFlagIsNonInterleaved | kAudioFormatFlagsNativeEndian;
destinationAudioFormat.mBitsPerChannel = sizeof (float) * 8;
destinationAudioFormat.mChannelsPerFrame = numChannels;
destinationAudioFormat.mBytesPerFrame = sizeof (float);
destinationAudioFormat.mFramesPerPacket = 1;
destinationAudioFormat.mBytesPerPacket = destinationAudioFormat.mFramesPerPacket * destinationAudioFormat.mBytesPerFrame;
status = ExtAudioFileSetProperty (audioFileRef,
kExtAudioFileProperty_ClientDataFormat,
sizeof (AudioStreamBasicDescription),
&destinationAudioFormat);
if (status == noErr)
{
bufferList.malloc (1, sizeof (AudioBufferList) + numChannels * sizeof (AudioBuffer));
bufferList->mNumberBuffers = numChannels;
ok = true;
}
}
}
}
~CoreAudioReader()
{
ExtAudioFileDispose (audioFileRef);
AudioFileClose (audioFileID);
}
//==============================================================================
bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
int64 startSampleInFile, int numSamples) override
{
clearSamplesBeyondAvailableLength (destSamples, numDestChannels, startOffsetInDestBuffer,
startSampleInFile, numSamples, lengthInSamples);
if (numSamples <= 0)
return true;
if (lastReadPosition != startSampleInFile)
{
OSStatus status = ExtAudioFileSeek (audioFileRef, startSampleInFile);
if (status != noErr)
return false;
lastReadPosition = startSampleInFile;
}
while (numSamples > 0)
{
const int numThisTime = jmin (8192, numSamples);
const size_t numBytes = sizeof (float) * (size_t) numThisTime;
audioDataBlock.ensureSize (numBytes * numChannels, false);
float* data = static_cast<float*> (audioDataBlock.getData());
for (int j = (int) numChannels; --j >= 0;)
{
bufferList->mBuffers[j].mNumberChannels = 1;
bufferList->mBuffers[j].mDataByteSize = (UInt32) numBytes;
bufferList->mBuffers[j].mData = data;
data += numThisTime;
}
UInt32 numFramesToRead = (UInt32) numThisTime;
OSStatus status = ExtAudioFileRead (audioFileRef, &numFramesToRead, bufferList);
if (status != noErr)
return false;
for (int i = numDestChannels; --i >= 0;)
{
if (destSamples[i] != nullptr)
{
if (i < (int) numChannels)
memcpy (destSamples[i] + startOffsetInDestBuffer, bufferList->mBuffers[i].mData, numBytes);
else
zeromem (destSamples[i] + startOffsetInDestBuffer, numBytes);
}
}
startOffsetInDestBuffer += numThisTime;
numSamples -= numThisTime;
lastReadPosition += numThisTime;
}
return true;
}
bool ok;
private:
AudioFileID audioFileID;
ExtAudioFileRef audioFileRef;
AudioStreamBasicDescription destinationAudioFormat;
MemoryBlock audioDataBlock;
HeapBlock<AudioBufferList> bufferList;
int64 lastReadPosition;
static SInt64 getSizeCallback (void* inClientData)
{
return static_cast<CoreAudioReader*> (inClientData)->input->getTotalLength();
}
static OSStatus readCallback (void* inClientData,
SInt64 inPosition,
UInt32 requestCount,
void* buffer,
UInt32* actualCount)
{
CoreAudioReader* const reader = static_cast<CoreAudioReader*> (inClientData);
reader->input->setPosition (inPosition);
*actualCount = (UInt32) reader->input->read (buffer, (int) requestCount);
return noErr;
}
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (CoreAudioReader)
};
//==============================================================================
CoreAudioFormat::CoreAudioFormat()
: AudioFormat (coreAudioFormatName, findFileExtensionsForCoreAudioCodecs())
{
}
CoreAudioFormat::~CoreAudioFormat() {}
Array<int> CoreAudioFormat::getPossibleSampleRates() { return Array<int>(); }
Array<int> CoreAudioFormat::getPossibleBitDepths() { return Array<int>(); }
bool CoreAudioFormat::canDoStereo() { return true; }
bool CoreAudioFormat::canDoMono() { return true; }
//==============================================================================
AudioFormatReader* CoreAudioFormat::createReaderFor (InputStream* sourceStream,
bool deleteStreamIfOpeningFails)
{
ScopedPointer<CoreAudioReader> r (new CoreAudioReader (sourceStream));
if (r->ok)
return r.release();
if (! deleteStreamIfOpeningFails)
r->input = nullptr;
return nullptr;
}
AudioFormatWriter* CoreAudioFormat::createWriterFor (OutputStream*,
double /*sampleRateToUse*/,
unsigned int /*numberOfChannels*/,
int /*bitsPerSample*/,
const StringPairArray& /*metadataValues*/,
int /*qualityOptionIndex*/)
{
jassertfalse; // not yet implemented!
return nullptr;
}
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_CoreAudioFormat.h
+77
View File
@@ -0,0 +1,77 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
#if JUCE_MAC || JUCE_IOS || DOXYGEN
//==============================================================================
/**
OSX and iOS only - This uses the AudioToolbox framework to read any audio
format that the system has a codec for.
This should be able to understand formats such as mp3, m4a, etc.
@see AudioFormat
*/
class JUCE_API CoreAudioFormat : public AudioFormat
{
public:
//==============================================================================
/** Creates a format object. */
CoreAudioFormat();
/** Destructor. */
~CoreAudioFormat();
//==============================================================================
/** Metadata property name used when reading a caf file with a MIDI chunk. */
static const char* const midiDataBase64;
/** Metadata property name used when reading a caf file with tempo information. */
static const char* const tempo;
/** Metadata property name used when reading a caf file time signature information. */
static const char* const timeSig;
/** Metadata property name used when reading a caf file time signature information. */
static const char* const keySig;
//==============================================================================
Array<int> getPossibleSampleRates() override;
Array<int> getPossibleBitDepths() override;
bool canDoStereo() override;
bool canDoMono() override;
//==============================================================================
AudioFormatReader* createReaderFor (InputStream*,
bool deleteStreamIfOpeningFails) override;
AudioFormatWriter* createWriterFor (OutputStream*,
double sampleRateToUse,
unsigned int numberOfChannels,
int bitsPerSample,
const StringPairArray& metadataValues,
int qualityOptionIndex) override;
private:
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (CoreAudioFormat)
};
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_FlacAudioFormat.cpp
+556
View File
@@ -0,0 +1,556 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
#if JUCE_USE_FLAC
namespace FlacNamespace
{
#if JUCE_INCLUDE_FLAC_CODE || ! defined (JUCE_INCLUDE_FLAC_CODE)
#undef VERSION
#define VERSION "1.2.1"
#define FLAC__NO_DLL 1
#if JUCE_MSVC
#pragma warning (disable: 4267 4127 4244 4996 4100 4701 4702 4013 4133 4206 4312 4505 4365 4005 4334 181 111)
#endif
#if JUCE_MAC
#define FLAC__SYS_DARWIN 1
#endif
#ifndef SIZE_MAX
#define SIZE_MAX 0xffffffff
#endif
#if JUCE_CLANG
#pragma clang diagnostic push
#pragma clang diagnostic ignored "-Wconversion"
#pragma clang diagnostic ignored "-Wshadow"
#pragma clang diagnostic ignored "-Wdeprecated-register"
#endif
#if JUCE_INTEL
#if JUCE_32BIT
#define FLAC__CPU_IA32 1
#endif
#if JUCE_64BIT
#define FLAC__CPU_X86_64 1
#endif
#define FLAC__HAS_X86INTRIN 1
#endif
#undef __STDC_LIMIT_MACROS
#define __STDC_LIMIT_MACROS 1
#define flac_max jmax
#define flac_min jmin
#include "flac/all.h"
#include "flac/libFLAC/bitmath.c"
#include "flac/libFLAC/bitreader.c"
#include "flac/libFLAC/bitwriter.c"
#include "flac/libFLAC/cpu.c"
#include "flac/libFLAC/crc.c"
#include "flac/libFLAC/fixed.c"
#include "flac/libFLAC/float.c"
#include "flac/libFLAC/format.c"
#include "flac/libFLAC/lpc_flac.c"
#include "flac/libFLAC/md5.c"
#include "flac/libFLAC/memory.c"
#include "flac/libFLAC/stream_decoder.c"
#include "flac/libFLAC/stream_encoder.c"
#include "flac/libFLAC/stream_encoder_framing.c"
#include "flac/libFLAC/window_flac.c"
#undef VERSION
#else
#include <FLAC/all.h>
#endif
#if JUCE_CLANG
#pragma clang diagnostic pop
#endif
}
#undef max
#undef min
//==============================================================================
static const char* const flacFormatName = "FLAC file";
//==============================================================================
class FlacReader : public AudioFormatReader
{
public:
FlacReader (InputStream* const in)
: AudioFormatReader (in, flacFormatName),
reservoirStart (0),
samplesInReservoir (0),
scanningForLength (false)
{
using namespace FlacNamespace;
lengthInSamples = 0;
decoder = FLAC__stream_decoder_new();
ok = FLAC__stream_decoder_init_stream (decoder,
readCallback_, seekCallback_, tellCallback_, lengthCallback_,
eofCallback_, writeCallback_, metadataCallback_, errorCallback_,
this) == FLAC__STREAM_DECODER_INIT_STATUS_OK;
if (ok)
{
FLAC__stream_decoder_process_until_end_of_metadata (decoder);
if (lengthInSamples == 0 && sampleRate > 0)
{
// the length hasn't been stored in the metadata, so we'll need to
// work it out the length the hard way, by scanning the whole file..
scanningForLength = true;
FLAC__stream_decoder_process_until_end_of_stream (decoder);
scanningForLength = false;
const int64 tempLength = lengthInSamples;
FLAC__stream_decoder_reset (decoder);
FLAC__stream_decoder_process_until_end_of_metadata (decoder);
lengthInSamples = tempLength;
}
}
}
~FlacReader()
{
FlacNamespace::FLAC__stream_decoder_delete (decoder);
}
void useMetadata (const FlacNamespace::FLAC__StreamMetadata_StreamInfo& info)
{
sampleRate = info.sample_rate;
bitsPerSample = info.bits_per_sample;
lengthInSamples = (unsigned int) info.total_samples;
numChannels = info.channels;
reservoir.setSize ((int) numChannels, 2 * (int) info.max_blocksize, false, false, true);
}
// returns the number of samples read
bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
int64 startSampleInFile, int numSamples) override
{
using namespace FlacNamespace;
if (! ok)
return false;
while (numSamples > 0)
{
if (startSampleInFile >= reservoirStart
&& startSampleInFile < reservoirStart + samplesInReservoir)
{
const int num = (int) jmin ((int64) numSamples,
reservoirStart + samplesInReservoir - startSampleInFile);
jassert (num > 0);
for (int i = jmin (numDestChannels, reservoir.getNumChannels()); --i >= 0;)
if (destSamples[i] != nullptr)
memcpy (destSamples[i] + startOffsetInDestBuffer,
reservoir.getReadPointer (i, (int) (startSampleInFile - reservoirStart)),
sizeof (int) * (size_t) num);
startOffsetInDestBuffer += num;
startSampleInFile += num;
numSamples -= num;
}
else
{
if (startSampleInFile >= (int) lengthInSamples)
{
samplesInReservoir = 0;
}
else if (startSampleInFile < reservoirStart
|| startSampleInFile > reservoirStart + jmax (samplesInReservoir, 511))
{
// had some problems with flac crashing if the read pos is aligned more
// accurately than this. Probably fixed in newer versions of the library, though.
reservoirStart = (int) (startSampleInFile & ~511);
samplesInReservoir = 0;
FLAC__stream_decoder_seek_absolute (decoder, (FLAC__uint64) reservoirStart);
}
else
{
reservoirStart += samplesInReservoir;
samplesInReservoir = 0;
FLAC__stream_decoder_process_single (decoder);
}
if (samplesInReservoir == 0)
break;
}
}
if (numSamples > 0)
{
for (int i = numDestChannels; --i >= 0;)
if (destSamples[i] != nullptr)
zeromem (destSamples[i] + startOffsetInDestBuffer, sizeof (int) * (size_t) numSamples);
}
return true;
}
void useSamples (const FlacNamespace::FLAC__int32* const buffer[], int numSamples)
{
if (scanningForLength)
{
lengthInSamples += numSamples;
}
else
{
if (numSamples > reservoir.getNumSamples())
reservoir.setSize ((int) numChannels, numSamples, false, false, true);
const unsigned int bitsToShift = 32 - bitsPerSample;
for (int i = 0; i < (int) numChannels; ++i)
{
const FlacNamespace::FLAC__int32* src = buffer[i];
int n = i;
while (src == 0 && n > 0)
src = buffer [--n];
if (src != nullptr)
{
int* const dest = reinterpret_cast<int*> (reservoir.getWritePointer(i));
for (int j = 0; j < numSamples; ++j)
dest[j] = src[j] << bitsToShift;
}
}
samplesInReservoir = numSamples;
}
}
//==============================================================================
static FlacNamespace::FLAC__StreamDecoderReadStatus readCallback_ (const FlacNamespace::FLAC__StreamDecoder*, FlacNamespace::FLAC__byte buffer[], size_t* bytes, void* client_data)
{
using namespace FlacNamespace;
*bytes = (size_t) static_cast<const FlacReader*> (client_data)->input->read (buffer, (int) *bytes);
return FLAC__STREAM_DECODER_READ_STATUS_CONTINUE;
}
static FlacNamespace::FLAC__StreamDecoderSeekStatus seekCallback_ (const FlacNamespace::FLAC__StreamDecoder*, FlacNamespace::FLAC__uint64 absolute_byte_offset, void* client_data)
{
using namespace FlacNamespace;
static_cast<const FlacReader*> (client_data)->input->setPosition ((int) absolute_byte_offset);
return FLAC__STREAM_DECODER_SEEK_STATUS_OK;
}
static FlacNamespace::FLAC__StreamDecoderTellStatus tellCallback_ (const FlacNamespace::FLAC__StreamDecoder*, FlacNamespace::FLAC__uint64* absolute_byte_offset, void* client_data)
{
using namespace FlacNamespace;
*absolute_byte_offset = (uint64) static_cast<const FlacReader*> (client_data)->input->getPosition();
return FLAC__STREAM_DECODER_TELL_STATUS_OK;
}
static FlacNamespace::FLAC__StreamDecoderLengthStatus lengthCallback_ (const FlacNamespace::FLAC__StreamDecoder*, FlacNamespace::FLAC__uint64* stream_length, void* client_data)
{
using namespace FlacNamespace;
*stream_length = (uint64) static_cast<const FlacReader*> (client_data)->input->getTotalLength();
return FLAC__STREAM_DECODER_LENGTH_STATUS_OK;
}
static FlacNamespace::FLAC__bool eofCallback_ (const FlacNamespace::FLAC__StreamDecoder*, void* client_data)
{
return static_cast<const FlacReader*> (client_data)->input->isExhausted();
}
static FlacNamespace::FLAC__StreamDecoderWriteStatus writeCallback_ (const FlacNamespace::FLAC__StreamDecoder*,
const FlacNamespace::FLAC__Frame* frame,
const FlacNamespace::FLAC__int32* const buffer[],
void* client_data)
{
using namespace FlacNamespace;
static_cast<FlacReader*> (client_data)->useSamples (buffer, (int) frame->header.blocksize);
return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
static void metadataCallback_ (const FlacNamespace::FLAC__StreamDecoder*,
const FlacNamespace::FLAC__StreamMetadata* metadata,
void* client_data)
{
static_cast<FlacReader*> (client_data)->useMetadata (metadata->data.stream_info);
}
static void errorCallback_ (const FlacNamespace::FLAC__StreamDecoder*, FlacNamespace::FLAC__StreamDecoderErrorStatus, void*)
{
}
private:
FlacNamespace::FLAC__StreamDecoder* decoder;
AudioSampleBuffer reservoir;
int reservoirStart, samplesInReservoir;
bool ok, scanningForLength;
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (FlacReader)
};
//==============================================================================
class FlacWriter : public AudioFormatWriter
{
public:
FlacWriter (OutputStream* const out, double rate, uint32 numChans, uint32 bits, int qualityOptionIndex)
: AudioFormatWriter (out, flacFormatName, rate, numChans, bits)
{
using namespace FlacNamespace;
encoder = FLAC__stream_encoder_new();
if (qualityOptionIndex > 0)
FLAC__stream_encoder_set_compression_level (encoder, (uint32) jmin (8, qualityOptionIndex));
FLAC__stream_encoder_set_do_mid_side_stereo (encoder, numChannels == 2);
FLAC__stream_encoder_set_loose_mid_side_stereo (encoder, numChannels == 2);
FLAC__stream_encoder_set_channels (encoder, numChannels);
FLAC__stream_encoder_set_bits_per_sample (encoder, jmin ((unsigned int) 24, bitsPerSample));
FLAC__stream_encoder_set_sample_rate (encoder, (unsigned int) sampleRate);
FLAC__stream_encoder_set_blocksize (encoder, 0);
FLAC__stream_encoder_set_do_escape_coding (encoder, true);
ok = FLAC__stream_encoder_init_stream (encoder,
encodeWriteCallback, encodeSeekCallback,
encodeTellCallback, encodeMetadataCallback,
this) == FLAC__STREAM_ENCODER_INIT_STATUS_OK;
}
~FlacWriter()
{
if (ok)
{
FlacNamespace::FLAC__stream_encoder_finish (encoder);
output->flush();
}
else
{
output = nullptr; // to stop the base class deleting this, as it needs to be returned
// to the caller of createWriter()
}
FlacNamespace::FLAC__stream_encoder_delete (encoder);
}
//==============================================================================
bool write (const int** samplesToWrite, int numSamples) override
{
using namespace FlacNamespace;
if (! ok)
return false;
HeapBlock<int*> channels;
HeapBlock<int> temp;
const int bitsToShift = 32 - (int) bitsPerSample;
if (bitsToShift > 0)
{
temp.malloc (numChannels * (size_t) numSamples);
channels.calloc (numChannels + 1);
for (unsigned int i = 0; i < numChannels; ++i)
{
if (samplesToWrite[i] == nullptr)
break;
int* const destData = temp.getData() + i * (size_t) numSamples;
channels[i] = destData;
for (int j = 0; j < numSamples; ++j)
destData[j] = (samplesToWrite[i][j] >> bitsToShift);
}
samplesToWrite = const_cast<const int**> (channels.getData());
}
return FLAC__stream_encoder_process (encoder, (const FLAC__int32**) samplesToWrite, (unsigned) numSamples) != 0;
}
bool writeData (const void* const data, const int size) const
{
return output->write (data, (size_t) size);
}
static void packUint32 (FlacNamespace::FLAC__uint32 val, FlacNamespace::FLAC__byte* b, const int bytes)
{
b += bytes;
for (int i = 0; i < bytes; ++i)
{
*(--b) = (FlacNamespace::FLAC__byte) (val & 0xff);
val >>= 8;
}
}
void writeMetaData (const FlacNamespace::FLAC__StreamMetadata* metadata)
{
using namespace FlacNamespace;
const FLAC__StreamMetadata_StreamInfo& info = metadata->data.stream_info;
unsigned char buffer [FLAC__STREAM_METADATA_STREAMINFO_LENGTH];
const unsigned int channelsMinus1 = info.channels - 1;
const unsigned int bitsMinus1 = info.bits_per_sample - 1;
packUint32 (info.min_blocksize, buffer, 2);
packUint32 (info.max_blocksize, buffer + 2, 2);
packUint32 (info.min_framesize, buffer + 4, 3);
packUint32 (info.max_framesize, buffer + 7, 3);
buffer[10] = (uint8) ((info.sample_rate >> 12) & 0xff);
buffer[11] = (uint8) ((info.sample_rate >> 4) & 0xff);
buffer[12] = (uint8) (((info.sample_rate & 0x0f) << 4) | (channelsMinus1 << 1) | (bitsMinus1 >> 4));
buffer[13] = (FLAC__byte) (((bitsMinus1 & 0x0f) << 4) | (unsigned int) ((info.total_samples >> 32) & 0x0f));
packUint32 ((FLAC__uint32) info.total_samples, buffer + 14, 4);
memcpy (buffer + 18, info.md5sum, 16);
const bool seekOk = output->setPosition (4);
(void) seekOk;
// if this fails, you've given it an output stream that can't seek! It needs
// to be able to seek back to write the header
jassert (seekOk);
output->writeIntBigEndian (FLAC__STREAM_METADATA_STREAMINFO_LENGTH);
output->write (buffer, FLAC__STREAM_METADATA_STREAMINFO_LENGTH);
}
//==============================================================================
static FlacNamespace::FLAC__StreamEncoderWriteStatus encodeWriteCallback (const FlacNamespace::FLAC__StreamEncoder*,
const FlacNamespace::FLAC__byte buffer[],
size_t bytes,
unsigned int /*samples*/,
unsigned int /*current_frame*/,
void* client_data)
{
using namespace FlacNamespace;
return static_cast<FlacWriter*> (client_data)->writeData (buffer, (int) bytes)
? FLAC__STREAM_ENCODER_WRITE_STATUS_OK
: FLAC__STREAM_ENCODER_WRITE_STATUS_FATAL_ERROR;
}
static FlacNamespace::FLAC__StreamEncoderSeekStatus encodeSeekCallback (const FlacNamespace::FLAC__StreamEncoder*, FlacNamespace::FLAC__uint64, void*)
{
using namespace FlacNamespace;
return FLAC__STREAM_ENCODER_SEEK_STATUS_UNSUPPORTED;
}
static FlacNamespace::FLAC__StreamEncoderTellStatus encodeTellCallback (const FlacNamespace::FLAC__StreamEncoder*, FlacNamespace::FLAC__uint64* absolute_byte_offset, void* client_data)
{
using namespace FlacNamespace;
if (client_data == nullptr)
return FLAC__STREAM_ENCODER_TELL_STATUS_UNSUPPORTED;
*absolute_byte_offset = (FLAC__uint64) static_cast<FlacWriter*> (client_data)->output->getPosition();
return FLAC__STREAM_ENCODER_TELL_STATUS_OK;
}
static void encodeMetadataCallback (const FlacNamespace::FLAC__StreamEncoder*, const FlacNamespace::FLAC__StreamMetadata* metadata, void* client_data)
{
static_cast<FlacWriter*> (client_data)->writeMetaData (metadata);
}
bool ok;
private:
FlacNamespace::FLAC__StreamEncoder* encoder;
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (FlacWriter)
};
//==============================================================================
FlacAudioFormat::FlacAudioFormat()
: AudioFormat (flacFormatName, ".flac")
{
}
FlacAudioFormat::~FlacAudioFormat()
{
}
Array<int> FlacAudioFormat::getPossibleSampleRates()
{
const int rates[] = { 8000, 11025, 12000, 16000, 22050, 32000, 44100, 48000,
88200, 96000, 176400, 192000, 352800, 384000 };
return Array<int> (rates, numElementsInArray (rates));
}
Array<int> FlacAudioFormat::getPossibleBitDepths()
{
const int depths[] = { 16, 24 };
return Array<int> (depths, numElementsInArray (depths));
}
bool FlacAudioFormat::canDoStereo() { return true; }
bool FlacAudioFormat::canDoMono() { return true; }
bool FlacAudioFormat::isCompressed() { return true; }
AudioFormatReader* FlacAudioFormat::createReaderFor (InputStream* in, const bool deleteStreamIfOpeningFails)
{
ScopedPointer<FlacReader> r (new FlacReader (in));
if (r->sampleRate > 0)
return r.release();
if (! deleteStreamIfOpeningFails)
r->input = nullptr;
return nullptr;
}
AudioFormatWriter* FlacAudioFormat::createWriterFor (OutputStream* out,
double sampleRate,
unsigned int numberOfChannels,
int bitsPerSample,
const StringPairArray& /*metadataValues*/,
int qualityOptionIndex)
{
if (getPossibleBitDepths().contains (bitsPerSample))
{
ScopedPointer<FlacWriter> w (new FlacWriter (out, sampleRate, numberOfChannels,
(uint32) bitsPerSample, qualityOptionIndex));
if (w->ok)
return w.release();
}
return nullptr;
}
StringArray FlacAudioFormat::getQualityOptions()
{
static const char* options[] = { "0 (Fastest)", "1", "2", "3", "4", "5 (Default)","6", "7", "8 (Highest quality)", 0 };
return StringArray (options);
}
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_FlacAudioFormat.h
+65
View File
@@ -0,0 +1,65 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
#if JUCE_USE_FLAC || defined (DOXYGEN)
//==============================================================================
/**
Reads and writes the lossless-compression FLAC audio format.
To compile this, you'll need to set the JUCE_USE_FLAC flag.
@see AudioFormat
*/
class JUCE_API FlacAudioFormat : public AudioFormat
{
public:
//==============================================================================
FlacAudioFormat();
~FlacAudioFormat();
//==============================================================================
Array<int> getPossibleSampleRates() override;
Array<int> getPossibleBitDepths() override;
bool canDoStereo() override;
bool canDoMono() override;
bool isCompressed() override;
StringArray getQualityOptions() override;
//==============================================================================
AudioFormatReader* createReaderFor (InputStream* sourceStream,
bool deleteStreamIfOpeningFails) override;
AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
double sampleRateToUse,
unsigned int numberOfChannels,
int bitsPerSample,
const StringPairArray& metadataValues,
int qualityOptionIndex) override;
private:
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (FlacAudioFormat)
};
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_LAMEEncoderAudioFormat.cpp
+222
View File
@@ -0,0 +1,222 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
#if JUCE_USE_LAME_AUDIO_FORMAT
class LAMEEncoderAudioFormat::Writer : public AudioFormatWriter
{
public:
Writer (OutputStream* destStream, const String& formatName,
const File& appFile, int vbr, int cbr,
double sampleRate, unsigned int numberOfChannels,
unsigned int bitsPerSample, const StringPairArray& metadata)
: AudioFormatWriter (destStream, formatName, sampleRate,
numberOfChannels, bitsPerSample),
vbrLevel (vbr), cbrBitrate (cbr),
tempWav (".wav")
{
WavAudioFormat wavFormat;
if (FileOutputStream* out = tempWav.getFile().createOutputStream())
{
writer = wavFormat.createWriterFor (out, sampleRate, numChannels,
bitsPerSample, metadata, 0);
args.add (appFile.getFullPathName());
args.add ("--quiet");
if (cbrBitrate == 0)
{
args.add ("--vbr-new");
args.add ("-V");
args.add (String (vbrLevel));
}
else
{
args.add ("--cbr");
args.add ("-b");
args.add (String (cbrBitrate));
}
addMetadataArg (metadata, "id3title", "--tt");
addMetadataArg (metadata, "id3artist", "--ta");
addMetadataArg (metadata, "id3album", "--tl");
addMetadataArg (metadata, "id3comment", "--tc");
addMetadataArg (metadata, "id3date", "--ty");
addMetadataArg (metadata, "id3genre", "--tg");
addMetadataArg (metadata, "id3trackNumber", "--tn");
}
}
void addMetadataArg (const StringPairArray& metadata, const char* key, const char* lameFlag)
{
const String value (metadata.getValue (key, String()));
if (value.isNotEmpty())
{
args.add (lameFlag);
args.add (value);
}
}
~Writer()
{
if (writer != nullptr)
{
writer = nullptr;
if (! convertToMP3())
convertToMP3(); // try again
}
}
bool write (const int** samplesToWrite, int numSamples)
{
return writer != nullptr && writer->write (samplesToWrite, numSamples);
}
private:
int vbrLevel, cbrBitrate;
TemporaryFile tempWav;
ScopedPointer<AudioFormatWriter> writer;
StringArray args;
bool runLameChildProcess (const TemporaryFile& tempMP3, const StringArray& processArgs) const
{
ChildProcess cp;
if (cp.start (processArgs))
{
const String childOutput (cp.readAllProcessOutput());
DBG (childOutput); (void) childOutput;
cp.waitForProcessToFinish (10000);
return tempMP3.getFile().getSize() > 0;
}
return false;
}
bool convertToMP3() const
{
TemporaryFile tempMP3 (".mp3");
StringArray args2 (args);
args2.add (tempWav.getFile().getFullPathName());
args2.add (tempMP3.getFile().getFullPathName());
DBG (args2.joinIntoString (" "));
if (runLameChildProcess (tempMP3, args2))
{
FileInputStream fis (tempMP3.getFile());
if (fis.openedOk() && output->writeFromInputStream (fis, -1) > 0)
{
output->flush();
return true;
}
}
return false;
}
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (Writer)
};
//==============================================================================
LAMEEncoderAudioFormat::LAMEEncoderAudioFormat (const File& lameApplication)
: AudioFormat ("MP3 file", ".mp3"),
lameApp (lameApplication)
{
}
LAMEEncoderAudioFormat::~LAMEEncoderAudioFormat()
{
}
bool LAMEEncoderAudioFormat::canHandleFile (const File&)
{
return false;
}
Array<int> LAMEEncoderAudioFormat::getPossibleSampleRates()
{
const int rates[] = { 32000, 44100, 48000, 0 };
return Array<int> (rates);
}
Array<int> LAMEEncoderAudioFormat::getPossibleBitDepths()
{
const int depths[] = { 16, 0 };
return Array<int> (depths);
}
bool LAMEEncoderAudioFormat::canDoStereo() { return true; }
bool LAMEEncoderAudioFormat::canDoMono() { return true; }
bool LAMEEncoderAudioFormat::isCompressed() { return true; }
StringArray LAMEEncoderAudioFormat::getQualityOptions()
{
static const char* vbrOptions[] = { "VBR quality 0 (best)", "VBR quality 1", "VBR quality 2", "VBR quality 3",
"VBR quality 4 (normal)", "VBR quality 5", "VBR quality 6", "VBR quality 7",
"VBR quality 8", "VBR quality 9 (smallest)", nullptr };
StringArray opts (vbrOptions);
const int cbrRates[] = { 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 };
for (int i = 0; i < numElementsInArray (cbrRates); ++i)
opts.add (String (cbrRates[i]) + " Kb/s CBR");
return opts;
}
AudioFormatReader* LAMEEncoderAudioFormat::createReaderFor (InputStream*, const bool)
{
return nullptr;
}
AudioFormatWriter* LAMEEncoderAudioFormat::createWriterFor (OutputStream* streamToWriteTo,
double sampleRateToUse,
unsigned int numberOfChannels,
int bitsPerSample,
const StringPairArray& metadataValues,
int qualityOptionIndex)
{
int vbr = 4;
int cbr = 0;
const String qual (getQualityOptions() [qualityOptionIndex]);
if (qual.contains ("VBR"))
vbr = qual.retainCharacters ("0123456789").getIntValue();
else
cbr = qual.getIntValue();
return new Writer (streamToWriteTo, getFormatName(), lameApp, vbr, cbr,
sampleRateToUse, numberOfChannels, bitsPerSample, metadataValues);
}
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_LAMEEncoderAudioFormat.h
+71
View File
@@ -0,0 +1,71 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
#if JUCE_USE_LAME_AUDIO_FORMAT || defined (DOXYGEN)
//==============================================================================
/**
An AudioFormat class which can use an installed version of the LAME mp3
encoder to encode a file.
This format can't read MP3s, it just writes them. Internally, the
AudioFormatWriter object that is returned writes the incoming audio data
to a temporary WAV file, and then when the writer is deleted, it invokes
the LAME executable to convert the data to an MP3, whose data is then
piped into the original OutputStream that was used when first creating
the writer.
@see AudioFormat
*/
class JUCE_API LAMEEncoderAudioFormat : public AudioFormat
{
public:
/** Creates a LAMEEncoderAudioFormat that expects to find a working LAME
executable at the location given.
*/
LAMEEncoderAudioFormat (const File& lameExecutableToUse);
~LAMEEncoderAudioFormat();
bool canHandleFile (const File&);
Array<int> getPossibleSampleRates();
Array<int> getPossibleBitDepths();
bool canDoStereo();
bool canDoMono();
bool isCompressed();
StringArray getQualityOptions();
AudioFormatReader* createReaderFor (InputStream*, bool deleteStreamIfOpeningFails);
AudioFormatWriter* createWriterFor (OutputStream*, double sampleRateToUse,
unsigned int numberOfChannels, int bitsPerSample,
const StringPairArray& metadataValues, int qualityOptionIndex);
private:
File lameApp;
class Writer;
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (LAMEEncoderAudioFormat)
};
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_MP3AudioFormat.cpp
+3162
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_MP3AudioFormat.h
+64
View File
@@ -0,0 +1,64 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
#if JUCE_USE_MP3AUDIOFORMAT || DOXYGEN
//==============================================================================
/**
Software-based MP3 decoding format (doesn't currently provide an encoder).
IMPORTANT DISCLAIMER: By choosing to enable the JUCE_USE_MP3AUDIOFORMAT flag and
to compile the MP3 code into your software, you do so AT YOUR OWN RISK! By doing so,
you are agreeing that Raw Material Software is in no way responsible for any patent,
copyright, or other legal issues that you may suffer as a result.
The code in juce_MP3AudioFormat.cpp is NOT guaranteed to be free from infringements of 3rd-party
intellectual property. If you wish to use it, please seek your own independent advice about the
legality of doing so. If you are not willing to accept full responsibility for the consequences
of using this code, then do not enable the JUCE_USE_MP3AUDIOFORMAT setting.
*/
class MP3AudioFormat : public AudioFormat
{
public:
//==============================================================================
MP3AudioFormat();
~MP3AudioFormat();
//==============================================================================
Array<int> getPossibleSampleRates() override;
Array<int> getPossibleBitDepths() override;
bool canDoStereo() override;
bool canDoMono() override;
bool isCompressed() override;
StringArray getQualityOptions() override;
//==============================================================================
AudioFormatReader* createReaderFor (InputStream*, bool deleteStreamIfOpeningFails) override;
AudioFormatWriter* createWriterFor (OutputStream*, double sampleRateToUse,
unsigned int numberOfChannels, int bitsPerSample,
const StringPairArray& metadataValues, int qualityOptionIndex) override;
};
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_OggVorbisAudioFormat.cpp
+532
View File
@@ -0,0 +1,532 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
#if JUCE_USE_OGGVORBIS
#if JUCE_MAC && ! defined (__MACOSX__)
#define __MACOSX__ 1
#endif
namespace OggVorbisNamespace
{
#if JUCE_INCLUDE_OGGVORBIS_CODE || ! defined (JUCE_INCLUDE_OGGVORBIS_CODE)
#if JUCE_MSVC
#pragma warning (push)
#pragma warning (disable: 4267 4127 4244 4996 4100 4701 4702 4013 4133 4206 4305 4189 4706 4995 4365)
#endif
#if JUCE_CLANG
#pragma clang diagnostic push
#pragma clang diagnostic ignored "-Wconversion"
#pragma clang diagnostic ignored "-Wshadow"
#pragma clang diagnostic ignored "-Wdeprecated-register"
#endif
#include "oggvorbis/vorbisenc.h"
#include "oggvorbis/codec.h"
#include "oggvorbis/vorbisfile.h"
#include "oggvorbis/bitwise.c"
#include "oggvorbis/framing.c"
#include "oggvorbis/libvorbis-1.3.2/lib/analysis.c"
#include "oggvorbis/libvorbis-1.3.2/lib/bitrate.c"
#include "oggvorbis/libvorbis-1.3.2/lib/block.c"
#include "oggvorbis/libvorbis-1.3.2/lib/codebook.c"
#include "oggvorbis/libvorbis-1.3.2/lib/envelope.c"
#include "oggvorbis/libvorbis-1.3.2/lib/floor0.c"
#include "oggvorbis/libvorbis-1.3.2/lib/floor1.c"
#include "oggvorbis/libvorbis-1.3.2/lib/info.c"
#include "oggvorbis/libvorbis-1.3.2/lib/lpc.c"
#include "oggvorbis/libvorbis-1.3.2/lib/lsp.c"
#include "oggvorbis/libvorbis-1.3.2/lib/mapping0.c"
#include "oggvorbis/libvorbis-1.3.2/lib/mdct.c"
#include "oggvorbis/libvorbis-1.3.2/lib/psy.c"
#include "oggvorbis/libvorbis-1.3.2/lib/registry.c"
#include "oggvorbis/libvorbis-1.3.2/lib/res0.c"
#include "oggvorbis/libvorbis-1.3.2/lib/sharedbook.c"
#include "oggvorbis/libvorbis-1.3.2/lib/smallft.c"
#include "oggvorbis/libvorbis-1.3.2/lib/synthesis.c"
#include "oggvorbis/libvorbis-1.3.2/lib/vorbisenc.c"
#include "oggvorbis/libvorbis-1.3.2/lib/vorbisfile.c"
#include "oggvorbis/libvorbis-1.3.2/lib/window.c"
#if JUCE_MSVC
#pragma warning (pop)
#endif
#if JUCE_CLANG
#pragma clang diagnostic pop
#endif
#else
#include <vorbis/vorbisenc.h>
#include <vorbis/codec.h>
#include <vorbis/vorbisfile.h>
#endif
}
#undef max
#undef min
//==============================================================================
static const char* const oggFormatName = "Ogg-Vorbis file";
const char* const OggVorbisAudioFormat::encoderName = "encoder";
const char* const OggVorbisAudioFormat::id3title = "id3title";
const char* const OggVorbisAudioFormat::id3artist = "id3artist";
const char* const OggVorbisAudioFormat::id3album = "id3album";
const char* const OggVorbisAudioFormat::id3comment = "id3comment";
const char* const OggVorbisAudioFormat::id3date = "id3date";
const char* const OggVorbisAudioFormat::id3genre = "id3genre";
const char* const OggVorbisAudioFormat::id3trackNumber = "id3trackNumber";
//==============================================================================
class OggReader : public AudioFormatReader
{
public:
OggReader (InputStream* const inp)
: AudioFormatReader (inp, oggFormatName),
reservoirStart (0),
samplesInReservoir (0)
{
using namespace OggVorbisNamespace;
sampleRate = 0;
usesFloatingPointData = true;
callbacks.read_func = &oggReadCallback;
callbacks.seek_func = &oggSeekCallback;
callbacks.close_func = &oggCloseCallback;
callbacks.tell_func = &oggTellCallback;
const int err = ov_open_callbacks (input, &ovFile, 0, 0, callbacks);
if (err == 0)
{
vorbis_info* info = ov_info (&ovFile, -1);
vorbis_comment* const comment = ov_comment (&ovFile, -1);
addMetadataItem (comment, "ENCODER", OggVorbisAudioFormat::encoderName);
addMetadataItem (comment, "TITLE", OggVorbisAudioFormat::id3title);
addMetadataItem (comment, "ARTIST", OggVorbisAudioFormat::id3artist);
addMetadataItem (comment, "ALBUM", OggVorbisAudioFormat::id3album);
addMetadataItem (comment, "COMMENT", OggVorbisAudioFormat::id3comment);
addMetadataItem (comment, "DATE", OggVorbisAudioFormat::id3date);
addMetadataItem (comment, "GENRE", OggVorbisAudioFormat::id3genre);
addMetadataItem (comment, "TRACKNUMBER", OggVorbisAudioFormat::id3trackNumber);
lengthInSamples = (uint32) ov_pcm_total (&ovFile, -1);
numChannels = (unsigned int) info->channels;
bitsPerSample = 16;
sampleRate = info->rate;
reservoir.setSize ((int) numChannels, (int) jmin (lengthInSamples, (int64) 4096));
}
}
~OggReader()
{
OggVorbisNamespace::ov_clear (&ovFile);
}
void addMetadataItem (OggVorbisNamespace::vorbis_comment* comment, const char* name, const char* metadataName)
{
if (const char* value = vorbis_comment_query (comment, name, 0))
metadataValues.set (metadataName, value);
}
//==============================================================================
bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
int64 startSampleInFile, int numSamples) override
{
while (numSamples > 0)
{
const int numAvailable = (int) (reservoirStart + samplesInReservoir - startSampleInFile);
if (startSampleInFile >= reservoirStart && numAvailable > 0)
{
// got a few samples overlapping, so use them before seeking..
const int numToUse = jmin (numSamples, numAvailable);
for (int i = jmin (numDestChannels, reservoir.getNumChannels()); --i >= 0;)
if (destSamples[i] != nullptr)
memcpy (destSamples[i] + startOffsetInDestBuffer,
reservoir.getReadPointer (i, (int) (startSampleInFile - reservoirStart)),
sizeof (float) * (size_t) numToUse);
startSampleInFile += numToUse;
numSamples -= numToUse;
startOffsetInDestBuffer += numToUse;
if (numSamples == 0)
break;
}
if (startSampleInFile < reservoirStart
|| startSampleInFile + numSamples > reservoirStart + samplesInReservoir)
{
// buffer miss, so refill the reservoir
int bitStream = 0;
reservoirStart = jmax (0, (int) startSampleInFile);
samplesInReservoir = reservoir.getNumSamples();
if (reservoirStart != (int) OggVorbisNamespace::ov_pcm_tell (&ovFile))
OggVorbisNamespace::ov_pcm_seek (&ovFile, reservoirStart);
int offset = 0;
int numToRead = samplesInReservoir;
while (numToRead > 0)
{
float** dataIn = nullptr;
const long samps = OggVorbisNamespace::ov_read_float (&ovFile, &dataIn, numToRead, &bitStream);
if (samps <= 0)
break;
jassert (samps <= numToRead);
for (int i = jmin ((int) numChannels, reservoir.getNumChannels()); --i >= 0;)
memcpy (reservoir.getWritePointer (i, offset), dataIn[i], sizeof (float) * (size_t) samps);
numToRead -= samps;
offset += samps;
}
if (numToRead > 0)
reservoir.clear (offset, numToRead);
}
}
if (numSamples > 0)
{
for (int i = numDestChannels; --i >= 0;)
if (destSamples[i] != nullptr)
zeromem (destSamples[i] + startOffsetInDestBuffer, sizeof (int) * (size_t) numSamples);
}
return true;
}
//==============================================================================
static size_t oggReadCallback (void* ptr, size_t size, size_t nmemb, void* datasource)
{
return (size_t) (static_cast<InputStream*> (datasource)->read (ptr, (int) (size * nmemb))) / size;
}
static int oggSeekCallback (void* datasource, OggVorbisNamespace::ogg_int64_t offset, int whence)
{
InputStream* const in = static_cast<InputStream*> (datasource);
if (whence == SEEK_CUR)
offset += in->getPosition();
else if (whence == SEEK_END)
offset += in->getTotalLength();
in->setPosition (offset);
return 0;
}
static int oggCloseCallback (void*)
{
return 0;
}
static long oggTellCallback (void* datasource)
{
return (long) static_cast<InputStream*> (datasource)->getPosition();
}
private:
OggVorbisNamespace::OggVorbis_File ovFile;
OggVorbisNamespace::ov_callbacks callbacks;
AudioSampleBuffer reservoir;
int reservoirStart, samplesInReservoir;
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (OggReader)
};
//==============================================================================
class OggWriter : public AudioFormatWriter
{
public:
OggWriter (OutputStream* const out,
const double sampleRate_,
const unsigned int numChannels_,
const unsigned int bitsPerSample_,
const int qualityIndex,
const StringPairArray& metadata)
: AudioFormatWriter (out, oggFormatName, sampleRate_, numChannels_, bitsPerSample_),
ok (false)
{
using namespace OggVorbisNamespace;
vorbis_info_init (&vi);
if (vorbis_encode_init_vbr (&vi, (int) numChannels_, (int) sampleRate_,
jlimit (0.0f, 1.0f, qualityIndex * 0.1f)) == 0)
{
vorbis_comment_init (&vc);
addMetadata (metadata, OggVorbisAudioFormat::encoderName, "ENCODER");
addMetadata (metadata, OggVorbisAudioFormat::id3title, "TITLE");
addMetadata (metadata, OggVorbisAudioFormat::id3artist, "ARTIST");
addMetadata (metadata, OggVorbisAudioFormat::id3album, "ALBUM");
addMetadata (metadata, OggVorbisAudioFormat::id3comment, "COMMENT");
addMetadata (metadata, OggVorbisAudioFormat::id3date, "DATE");
addMetadata (metadata, OggVorbisAudioFormat::id3genre, "GENRE");
addMetadata (metadata, OggVorbisAudioFormat::id3trackNumber, "TRACKNUMBER");
vorbis_analysis_init (&vd, &vi);
vorbis_block_init (&vd, &vb);
ogg_stream_init (&os, Random::getSystemRandom().nextInt());
ogg_packet header;
ogg_packet header_comm;
ogg_packet header_code;
vorbis_analysis_headerout (&vd, &vc, &header, &header_comm, &header_code);
ogg_stream_packetin (&os, &header);
ogg_stream_packetin (&os, &header_comm);
ogg_stream_packetin (&os, &header_code);
for (;;)
{
if (ogg_stream_flush (&os, &og) == 0)
break;
output->write (og.header, (size_t) og.header_len);
output->write (og.body, (size_t) og.body_len);
}
ok = true;
}
}
~OggWriter()
{
using namespace OggVorbisNamespace;
if (ok)
{
// write a zero-length packet to show ogg that we're finished..
writeSamples (0);
ogg_stream_clear (&os);
vorbis_block_clear (&vb);
vorbis_dsp_clear (&vd);
vorbis_comment_clear (&vc);
vorbis_info_clear (&vi);
output->flush();
}
else
{
vorbis_info_clear (&vi);
output = nullptr; // to stop the base class deleting this, as it needs to be returned
// to the caller of createWriter()
}
}
//==============================================================================
bool write (const int** samplesToWrite, int numSamples) override
{
if (ok)
{
using namespace OggVorbisNamespace;
if (numSamples > 0)
{
const double gain = 1.0 / 0x80000000u;
float** const vorbisBuffer = vorbis_analysis_buffer (&vd, numSamples);
for (int i = (int) numChannels; --i >= 0;)
{
float* const dst = vorbisBuffer[i];
const int* const src = samplesToWrite [i];
if (src != nullptr && dst != nullptr)
{
for (int j = 0; j < numSamples; ++j)
dst[j] = (float) (src[j] * gain);
}
}
}
writeSamples (numSamples);
}
return ok;
}
void writeSamples (int numSamples)
{
using namespace OggVorbisNamespace;
vorbis_analysis_wrote (&vd, numSamples);
while (vorbis_analysis_blockout (&vd, &vb) == 1)
{
vorbis_analysis (&vb, 0);
vorbis_bitrate_addblock (&vb);
while (vorbis_bitrate_flushpacket (&vd, &op))
{
ogg_stream_packetin (&os, &op);
for (;;)
{
if (ogg_stream_pageout (&os, &og) == 0)
break;
output->write (og.header, (size_t) og.header_len);
output->write (og.body, (size_t) og.body_len);
if (ogg_page_eos (&og))
break;
}
}
}
}
bool ok;
private:
OggVorbisNamespace::ogg_stream_state os;
OggVorbisNamespace::ogg_page og;
OggVorbisNamespace::ogg_packet op;
OggVorbisNamespace::vorbis_info vi;
OggVorbisNamespace::vorbis_comment vc;
OggVorbisNamespace::vorbis_dsp_state vd;
OggVorbisNamespace::vorbis_block vb;
void addMetadata (const StringPairArray& metadata, const char* name, const char* vorbisName)
{
const String s (metadata [name]);
if (s.isNotEmpty())
vorbis_comment_add_tag (&vc, vorbisName, const_cast <char*> (s.toRawUTF8()));
}
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (OggWriter)
};
//==============================================================================
OggVorbisAudioFormat::OggVorbisAudioFormat() : AudioFormat (oggFormatName, ".ogg")
{
}
OggVorbisAudioFormat::~OggVorbisAudioFormat()
{
}
Array<int> OggVorbisAudioFormat::getPossibleSampleRates()
{
const int rates[] = { 8000, 11025, 12000, 16000, 22050, 32000,
44100, 48000, 88200, 96000, 176400, 192000 };
return Array<int> (rates, numElementsInArray (rates));
}
Array<int> OggVorbisAudioFormat::getPossibleBitDepths()
{
const int depths[] = { 32 };
return Array<int> (depths, numElementsInArray (depths));
}
bool OggVorbisAudioFormat::canDoStereo() { return true; }
bool OggVorbisAudioFormat::canDoMono() { return true; }
bool OggVorbisAudioFormat::isCompressed() { return true; }
AudioFormatReader* OggVorbisAudioFormat::createReaderFor (InputStream* in, const bool deleteStreamIfOpeningFails)
{
ScopedPointer<OggReader> r (new OggReader (in));
if (r->sampleRate > 0)
return r.release();
if (! deleteStreamIfOpeningFails)
r->input = nullptr;
return nullptr;
}
AudioFormatWriter* OggVorbisAudioFormat::createWriterFor (OutputStream* out,
double sampleRate,
unsigned int numChannels,
int bitsPerSample,
const StringPairArray& metadataValues,
int qualityOptionIndex)
{
ScopedPointer <OggWriter> w (new OggWriter (out, sampleRate, numChannels,
(unsigned int) bitsPerSample, qualityOptionIndex, metadataValues));
return w->ok ? w.release() : nullptr;
}
StringArray OggVorbisAudioFormat::getQualityOptions()
{
static const char* options[] = { "64 kbps", "80 kbps", "96 kbps", "112 kbps", "128 kbps", "160 kbps",
"192 kbps", "224 kbps", "256 kbps", "320 kbps", "500 kbps", 0 };
return StringArray (options);
}
int OggVorbisAudioFormat::estimateOggFileQuality (const File& source)
{
if (FileInputStream* const in = source.createInputStream())
{
ScopedPointer<AudioFormatReader> r (createReaderFor (in, true));
if (r != nullptr)
{
const double lengthSecs = r->lengthInSamples / r->sampleRate;
const int approxBitsPerSecond = (int) (source.getSize() * 8 / lengthSecs);
const StringArray qualities (getQualityOptions());
int bestIndex = 0;
int bestDiff = 10000;
for (int i = qualities.size(); --i >= 0;)
{
const int diff = std::abs (qualities[i].getIntValue() - approxBitsPerSecond);
if (diff < bestDiff)
{
bestDiff = diff;
bestIndex = i;
}
}
return bestIndex;
}
}
return 0;
}
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_OggVorbisAudioFormat.h
+93
View File
@@ -0,0 +1,93 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
#if JUCE_USE_OGGVORBIS || defined (DOXYGEN)
//==============================================================================
/**
Reads and writes the Ogg-Vorbis audio format.
To compile this, you'll need to set the JUCE_USE_OGGVORBIS flag.
@see AudioFormat,
*/
class JUCE_API OggVorbisAudioFormat : public AudioFormat
{
public:
//==============================================================================
OggVorbisAudioFormat();
~OggVorbisAudioFormat();
//==============================================================================
Array<int> getPossibleSampleRates() override;
Array<int> getPossibleBitDepths() override;
bool canDoStereo() override;
bool canDoMono() override;
bool isCompressed() override;
StringArray getQualityOptions() override;
//==============================================================================
/** Tries to estimate the quality level of an ogg file based on its size.
If it can't read the file for some reason, this will just return 1 (medium quality),
otherwise it will return the approximate quality setting that would have been used
to create the file.
@see getQualityOptions
*/
int estimateOggFileQuality (const File& source);
//==============================================================================
/** Metadata property name used by the Ogg writer - if you set a string for this
value, it will be written into the ogg file as the name of the encoder app.
@see createWriterFor
*/
static const char* const encoderName;
static const char* const id3title; /**< Metadata key for setting an ID3 title. */
static const char* const id3artist; /**< Metadata key for setting an ID3 artist name. */
static const char* const id3album; /**< Metadata key for setting an ID3 album. */
static const char* const id3comment; /**< Metadata key for setting an ID3 comment. */
static const char* const id3date; /**< Metadata key for setting an ID3 date. */
static const char* const id3genre; /**< Metadata key for setting an ID3 genre. */
static const char* const id3trackNumber; /**< Metadata key for setting an ID3 track number. */
//==============================================================================
AudioFormatReader* createReaderFor (InputStream* sourceStream,
bool deleteStreamIfOpeningFails) override;
AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
double sampleRateToUse,
unsigned int numberOfChannels,
int bitsPerSample,
const StringPairArray& metadataValues,
int qualityOptionIndex) override;
private:
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (OggVorbisAudioFormat)
};
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_QuickTimeAudioFormat.cpp
+391
View File
@@ -0,0 +1,391 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
#if JUCE_QUICKTIME && ! (JUCE_64BIT || JUCE_IOS)
} // (juce namespace)
#if ! JUCE_WINDOWS
#define Point CarbonDummyPointName // (workaround to avoid definition of "Point" by old Carbon headers)
#define Component CarbonDummyCompName
#include <QuickTime/Movies.h>
#include <QuickTime/QTML.h>
#include <QuickTime/QuickTimeComponents.h>
#include <QuickTime/MediaHandlers.h>
#include <QuickTime/ImageCodec.h>
#undef Point
#undef Component
#else
#if JUCE_MSVC
#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 set the JUCE_QUICKTIME flag to 0.
*/
#undef SIZE_MAX
#include <Movies.h>
#include <QTML.h>
#include <QuickTimeComponents.h>
#include <MediaHandlers.h>
#include <ImageCodec.h>
#undef SIZE_MAX
#if JUCE_MSVC
#pragma warning (pop)
#endif
#endif
namespace juce
{
bool juce_OpenQuickTimeMovieFromStream (InputStream* input, Movie& movie, Handle& dataHandle);
static const char* const quickTimeFormatName = "QuickTime file";
//==============================================================================
class QTAudioReader : public AudioFormatReader
{
public:
QTAudioReader (InputStream* const input_, const int trackNum_)
: AudioFormatReader (input_, quickTimeFormatName),
ok (false),
movie (0),
trackNum (trackNum_),
lastSampleRead (0),
lastThreadId (0),
extractor (0),
dataHandle (0)
{
JUCE_AUTORELEASEPOOL
{
bufferList.calloc (256, 1);
#if JUCE_WINDOWS
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);
MovieAudioExtractionBegin (movie, 0, &extractor);
unsigned long output_layout_size;
OSStatus err = MovieAudioExtractionGetPropertyInfo (extractor,
kQTPropertyClass_MovieAudioExtraction_Audio,
kQTMovieAudioExtractionAudioPropertyID_AudioChannelLayout,
0, &output_layout_size, 0);
if (err != noErr)
return;
HeapBlock <AudioChannelLayout> qt_audio_channel_layout;
qt_audio_channel_layout.calloc (output_layout_size, 1);
MovieAudioExtractionGetProperty (extractor,
kQTPropertyClass_MovieAudioExtraction_Audio,
kQTMovieAudioExtractionAudioPropertyID_AudioChannelLayout,
output_layout_size, qt_audio_channel_layout, 0);
qt_audio_channel_layout[0].mChannelLayoutTag = kAudioChannelLayoutTag_Stereo;
MovieAudioExtractionSetProperty (extractor,
kQTPropertyClass_MovieAudioExtraction_Audio,
kQTMovieAudioExtractionAudioPropertyID_AudioChannelLayout,
output_layout_size,
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 ((UInt32) 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 = jmax ((UInt32) 4096, (UInt32) (samplesPerFrame * inputStreamDesc.mBytesPerFrame) + 16);
dataBuffer.malloc (bufferList->mBuffers[0].mDataByteSize);
bufferList->mBuffers[0].mData = dataBuffer;
sampleRate = inputStreamDesc.mSampleRate;
bitsPerSample = 16;
numChannels = inputStreamDesc.mChannelsPerFrame;
detachThread();
ok = true;
}
}
~QTAudioReader()
{
JUCE_AUTORELEASEPOOL
{
checkThreadIsAttached();
if (dataHandle != nullptr)
DisposeHandle (dataHandle);
if (extractor != nullptr)
{
MovieAudioExtractionEnd (extractor);
extractor = nullptr;
}
DisposeMovie (movie);
#if JUCE_MAC
ExitMoviesOnThread ();
#endif
}
}
bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
int64 startSampleInFile, int numSamples)
{
JUCE_AUTORELEASEPOOL
{
checkThreadIsAttached();
bool readOk = true;
while (numSamples > 0)
{
if (lastSampleRead != startSampleInFile)
{
TimeRecord time;
time.scale = (TimeScale) inputStreamDesc.mSampleRate;
time.base = 0;
time.value.hi = 0;
time.value.lo = (UInt32) startSampleInFile;
OSStatus err = MovieAudioExtractionSetProperty (extractor,
kQTPropertyClass_MovieAudioExtraction_Movie,
kQTMovieAudioExtractionMoviePropertyID_CurrentTime,
sizeof (time), &time);
if (err != noErr)
{
readOk = false;
break;
}
}
int framesToDo = jmin (numSamples, (int) (bufferList->mBuffers[0].mDataByteSize / inputStreamDesc.mBytesPerFrame));
bufferList->mBuffers[0].mDataByteSize = inputStreamDesc.mBytesPerFrame * framesToDo;
UInt32 outFlags = 0;
UInt32 actualNumFrames = framesToDo;
OSStatus err = MovieAudioExtractionFillBuffer (extractor, &actualNumFrames, bufferList, &outFlags);
if (err != noErr)
{
readOk = false;
break;
}
lastSampleRead = startSampleInFile + actualNumFrames;
const int samplesReceived = actualNumFrames;
for (int j = numDestChannels; --j >= 0;)
{
if (destSamples[j] != nullptr)
{
const short* src = ((const short*) bufferList->mBuffers[0].mData) + j;
for (int i = 0; i < samplesReceived; ++i)
{
destSamples[j][startOffsetInDestBuffer + i] = (*src << 16);
src += numChannels;
}
}
}
startOffsetInDestBuffer += samplesReceived;
startSampleInFile += samplesReceived;
numSamples -= samplesReceived;
if (((outFlags & kQTMovieAudioExtractionComplete) != 0 || samplesReceived == 0) && numSamples > 0)
{
for (int j = numDestChannels; --j >= 0;)
if (destSamples[j] != nullptr)
zeromem (destSamples[j] + startOffsetInDestBuffer, sizeof (int) * numSamples);
break;
}
}
detachThread();
return readOk;
}
}
bool ok;
private:
Movie movie;
Media media;
Track track;
const int trackNum;
double trackUnitsPerFrame;
int samplesPerFrame;
int64 lastSampleRead;
Thread::ThreadID lastThreadId;
MovieAudioExtractionRef extractor;
AudioStreamBasicDescription inputStreamDesc;
HeapBlock <AudioBufferList> bufferList;
HeapBlock <char> dataBuffer;
Handle dataHandle;
//==============================================================================
void checkThreadIsAttached()
{
#if JUCE_MAC
if (Thread::getCurrentThreadId() != lastThreadId)
EnterMoviesOnThread (0);
AttachMovieToCurrentThread (movie);
#endif
}
void detachThread()
{
#if JUCE_MAC
DetachMovieFromCurrentThread (movie);
#endif
}
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (QTAudioReader)
};
//==============================================================================
QuickTimeAudioFormat::QuickTimeAudioFormat() : AudioFormat (quickTimeFormatName, ".mov .mp3 .mp4 .m4a")
{
}
QuickTimeAudioFormat::~QuickTimeAudioFormat()
{
}
Array<int> QuickTimeAudioFormat::getPossibleSampleRates() { return Array<int>(); }
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)
{
ScopedPointer<QTAudioReader> r (new QTAudioReader (sourceStream, 0));
if (r->ok)
return r.release();
if (! deleteStreamIfOpeningFails)
r->input = 0;
return nullptr;
}
AudioFormatWriter* QuickTimeAudioFormat::createWriterFor (OutputStream* /*streamToWriteTo*/,
double /*sampleRateToUse*/,
unsigned int /*numberOfChannels*/,
int /*bitsPerSample*/,
const StringPairArray& /*metadataValues*/,
int /*qualityOptionIndex*/)
{
jassertfalse; // not yet implemented!
return nullptr;
}
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_QuickTimeAudioFormat.h
+69
View File
@@ -0,0 +1,69 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
#if JUCE_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();
//==============================================================================
Array<int> getPossibleSampleRates();
Array<int> getPossibleBitDepths();
bool canDoStereo();
bool canDoMono();
//==============================================================================
AudioFormatReader* createReaderFor (InputStream* sourceStream,
bool deleteStreamIfOpeningFails);
AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
double sampleRateToUse,
unsigned int numberOfChannels,
int bitsPerSample,
const StringPairArray& metadataValues,
int qualityOptionIndex);
private:
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (QuickTimeAudioFormat)
};
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_WavAudioFormat.cpp
+1471
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_WavAudioFormat.h
+171
View File
@@ -0,0 +1,171 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
//==============================================================================
/**
Reads and Writes 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 char* 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 char* 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 char* 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 char* 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 char* 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 char* const bwavTimeReference;
/** Metadata property name used by wav readers and writers for adding
a BWAV chunk to the file.
@see AudioFormatReader::metadataValues, createWriterFor
*/
static const char* 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 StringPairArray createBWAVMetadata (const String& description,
const String& originator,
const String& originatorRef,
const Time dateAndTime,
const int64 timeReferenceSamples,
const String& codingHistory);
//==============================================================================
/** Metadata property name used when reading a WAV file with an acid chunk. */
static const char* const acidOneShot;
/** Metadata property name used when reading a WAV file with an acid chunk. */
static const char* const acidRootSet;
/** Metadata property name used when reading a WAV file with an acid chunk. */
static const char* const acidStretch;
/** Metadata property name used when reading a WAV file with an acid chunk. */
static const char* const acidDiskBased;
/** Metadata property name used when reading a WAV file with an acid chunk. */
static const char* const acidizerFlag;
/** Metadata property name used when reading a WAV file with an acid chunk. */
static const char* const acidRootNote;
/** Metadata property name used when reading a WAV file with an acid chunk. */
static const char* const acidBeats;
/** Metadata property name used when reading a WAV file with an acid chunk. */
static const char* const acidDenominator;
/** Metadata property name used when reading a WAV file with an acid chunk. */
static const char* const acidNumerator;
/** Metadata property name used when reading a WAV file with an acid chunk. */
static const char* const acidTempo;
//==============================================================================
/** Metadata property name used when reading an ISRC code from an AXML chunk. */
static const char* const ISRC;
/** Metadata property name used when reading a WAV file with a Tracktion chunk. */
static const char* const tracktionLoopInfo;
//==============================================================================
Array<int> getPossibleSampleRates() override;
Array<int> getPossibleBitDepths() override;
bool canDoStereo() override;
bool canDoMono() override;
//==============================================================================
AudioFormatReader* createReaderFor (InputStream* sourceStream,
bool deleteStreamIfOpeningFails) override;
MemoryMappedAudioFormatReader* createMemoryMappedReader (const File& file) override;
AudioFormatWriter* createWriterFor (OutputStream* streamToWriteTo,
double sampleRateToUse,
unsigned int numberOfChannels,
int bitsPerSample,
const StringPairArray& metadataValues,
int qualityOptionIndex) override;
//==============================================================================
/** Utility function to replace the metadata in a wav file with a new set of values.
If possible, this cheats by overwriting just the metadata region of the file, rather
than by copying the whole file again.
*/
bool replaceMetadataInFile (const File& wavFile, const StringPairArray& newMetadata);
private:
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (WavAudioFormat)
};
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_WindowsMediaAudioFormat.cpp
+350
View File
@@ -0,0 +1,350 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
namespace WindowsMediaCodec
{
class JuceIStream : public ComBaseClassHelper <IStream>
{
public:
JuceIStream (InputStream& in) noexcept
: ComBaseClassHelper <IStream> (0), source (in)
{
}
JUCE_COMRESULT Commit (DWORD) { return S_OK; }
JUCE_COMRESULT Write (const void*, ULONG, ULONG*) { return E_NOTIMPL; }
JUCE_COMRESULT Clone (IStream**) { return E_NOTIMPL; }
JUCE_COMRESULT SetSize (ULARGE_INTEGER) { return E_NOTIMPL; }
JUCE_COMRESULT Revert() { return E_NOTIMPL; }
JUCE_COMRESULT LockRegion (ULARGE_INTEGER, ULARGE_INTEGER, DWORD) { return E_NOTIMPL; }
JUCE_COMRESULT UnlockRegion (ULARGE_INTEGER, ULARGE_INTEGER, DWORD) { return E_NOTIMPL; }
JUCE_COMRESULT Read (void* dest, ULONG numBytes, ULONG* bytesRead)
{
const int numRead = source.read (dest, numBytes);
if (bytesRead != nullptr)
*bytesRead = numRead;
return (numRead == (int) numBytes) ? S_OK : S_FALSE;
}
JUCE_COMRESULT Seek (LARGE_INTEGER position, DWORD origin, ULARGE_INTEGER* resultPosition)
{
int64 newPos = (int64) position.QuadPart;
if (origin == STREAM_SEEK_CUR)
{
newPos += source.getPosition();
}
else if (origin == STREAM_SEEK_END)
{
const int64 len = source.getTotalLength();
if (len < 0)
return E_NOTIMPL;
newPos += len;
}
if (resultPosition != nullptr)
resultPosition->QuadPart = newPos;
return source.setPosition (newPos) ? S_OK : E_NOTIMPL;
}
JUCE_COMRESULT CopyTo (IStream* destStream, ULARGE_INTEGER numBytesToDo,
ULARGE_INTEGER* bytesRead, ULARGE_INTEGER* bytesWritten)
{
uint64 totalCopied = 0;
int64 numBytes = numBytesToDo.QuadPart;
while (numBytes > 0 && ! source.isExhausted())
{
char buffer [1024];
const int numToCopy = (int) jmin ((int64) sizeof (buffer), (int64) numBytes);
const int numRead = source.read (buffer, numToCopy);
if (numRead <= 0)
break;
destStream->Write (buffer, numRead, nullptr);
totalCopied += numRead;
}
if (bytesRead != nullptr) bytesRead->QuadPart = totalCopied;
if (bytesWritten != nullptr) bytesWritten->QuadPart = totalCopied;
return S_OK;
}
JUCE_COMRESULT Stat (STATSTG* stat, DWORD)
{
if (stat == nullptr)
return STG_E_INVALIDPOINTER;
zerostruct (*stat);
stat->type = STGTY_STREAM;
stat->cbSize.QuadPart = jmax ((int64) 0, source.getTotalLength());
return S_OK;
}
private:
InputStream& source;
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (JuceIStream)
};
//==============================================================================
static const char* wmFormatName = "Windows Media";
static const char* const extensions[] = { ".mp3", ".wmv", ".asf", ".wm", ".wma", 0 };
//==============================================================================
class WMAudioReader : public AudioFormatReader
{
public:
WMAudioReader (InputStream* const input_)
: AudioFormatReader (input_, TRANS (wmFormatName)),
wmvCoreLib ("Wmvcore.dll")
{
JUCE_LOAD_WINAPI_FUNCTION (wmvCoreLib, WMCreateSyncReader, wmCreateSyncReader,
HRESULT, (IUnknown*, DWORD, IWMSyncReader**))
if (wmCreateSyncReader != nullptr)
{
checkCoInitialiseCalled();
HRESULT hr = wmCreateSyncReader (nullptr, WMT_RIGHT_PLAYBACK, wmSyncReader.resetAndGetPointerAddress());
if (SUCCEEDED (hr))
hr = wmSyncReader->OpenStream (new JuceIStream (*input));
if (SUCCEEDED (hr))
{
WORD streamNum = 1;
hr = wmSyncReader->GetStreamNumberForOutput (0, &streamNum);
hr = wmSyncReader->SetReadStreamSamples (streamNum, false);
scanFileForDetails();
}
}
}
~WMAudioReader()
{
if (wmSyncReader != nullptr)
wmSyncReader->Close();
}
bool readSamples (int** destSamples, int numDestChannels, int startOffsetInDestBuffer,
int64 startSampleInFile, int numSamples) override
{
if (sampleRate <= 0)
return false;
checkCoInitialiseCalled();
const int stride = numChannels * sizeof (int16);
while (numSamples > 0)
{
if (! bufferedRange.contains (startSampleInFile))
{
const bool hasJumped = (startSampleInFile != bufferedRange.getEnd());
if (hasJumped)
wmSyncReader->SetRange ((QWORD) (startSampleInFile * 10000000 / (int64) sampleRate), 0);
ComSmartPtr<INSSBuffer> sampleBuffer;
QWORD sampleTime, duration;
DWORD flags, outputNum;
WORD streamNum;
HRESULT hr = wmSyncReader->GetNextSample (1, sampleBuffer.resetAndGetPointerAddress(),
&sampleTime, &duration, &flags, &outputNum, &streamNum);
if (sampleBuffer != nullptr)
{
BYTE* rawData = nullptr;
DWORD dataLength = 0;
hr = sampleBuffer->GetBufferAndLength (&rawData, &dataLength);
if (dataLength == 0)
return false;
if (hasJumped)
bufferedRange.setStart ((int64) ((sampleTime * (int64) sampleRate) / 10000000));
else
bufferedRange.setStart (bufferedRange.getEnd()); // (because the positions returned often aren't continguous)
bufferedRange.setLength ((int64) (dataLength / stride));
buffer.ensureSize ((int) dataLength);
memcpy (buffer.getData(), rawData, (size_t) dataLength);
}
else if (hr == NS_E_NO_MORE_SAMPLES)
{
bufferedRange.setStart (startSampleInFile);
bufferedRange.setLength (256);
buffer.ensureSize (256 * stride);
buffer.fillWith (0);
}
else
{
return false;
}
}
const int offsetInBuffer = (int) (startSampleInFile - bufferedRange.getStart());
const int16* const rawData = static_cast<const int16*> (addBytesToPointer (buffer.getData(), offsetInBuffer * stride));
const int numToDo = jmin (numSamples, (int) (bufferedRange.getLength() - offsetInBuffer));
for (int i = 0; i < numDestChannels; ++i)
{
jassert (destSamples[i] != nullptr);
const int srcChan = jmin (i, (int) numChannels - 1);
const int16* src = rawData + srcChan;
int* const dst = destSamples[i] + startOffsetInDestBuffer;
for (int j = 0; j < numToDo; ++j)
{
dst[j] = ((uint32) *src) << 16;
src += numChannels;
}
}
startSampleInFile += numToDo;
startOffsetInDestBuffer += numToDo;
numSamples -= numToDo;
}
return true;
}
private:
DynamicLibrary wmvCoreLib;
ComSmartPtr<IWMSyncReader> wmSyncReader;
MemoryBlock buffer;
Range<int64> bufferedRange;
void checkCoInitialiseCalled()
{
CoInitialize (0);
}
void scanFileForDetails()
{
ComSmartPtr<IWMHeaderInfo> wmHeaderInfo;
HRESULT hr = wmSyncReader.QueryInterface (wmHeaderInfo);
if (SUCCEEDED (hr))
{
QWORD lengthInNanoseconds = 0;
WORD lengthOfLength = sizeof (lengthInNanoseconds);
WORD streamNum = 0;
WMT_ATTR_DATATYPE wmAttrDataType;
hr = wmHeaderInfo->GetAttributeByName (&streamNum, L"Duration", &wmAttrDataType,
(BYTE*) &lengthInNanoseconds, &lengthOfLength);
ComSmartPtr<IWMProfile> wmProfile;
hr = wmSyncReader.QueryInterface (wmProfile);
if (SUCCEEDED (hr))
{
ComSmartPtr<IWMStreamConfig> wmStreamConfig;
hr = wmProfile->GetStream (0, wmStreamConfig.resetAndGetPointerAddress());
if (SUCCEEDED (hr))
{
ComSmartPtr<IWMMediaProps> wmMediaProperties;
hr = wmStreamConfig.QueryInterface (wmMediaProperties);
if (SUCCEEDED (hr))
{
DWORD sizeMediaType;
hr = wmMediaProperties->GetMediaType (0, &sizeMediaType);
HeapBlock<WM_MEDIA_TYPE> mediaType;
mediaType.malloc (sizeMediaType, 1);
hr = wmMediaProperties->GetMediaType (mediaType, &sizeMediaType);
if (mediaType->majortype == WMMEDIATYPE_Audio)
{
const WAVEFORMATEX* const inputFormat = reinterpret_cast<WAVEFORMATEX*> (mediaType->pbFormat);
sampleRate = inputFormat->nSamplesPerSec;
numChannels = inputFormat->nChannels;
bitsPerSample = inputFormat->wBitsPerSample;
lengthInSamples = (lengthInNanoseconds * (int) sampleRate) / 10000000;
}
}
}
}
}
}
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (WMAudioReader)
};
}
//==============================================================================
WindowsMediaAudioFormat::WindowsMediaAudioFormat()
: AudioFormat (TRANS (WindowsMediaCodec::wmFormatName),
StringArray (WindowsMediaCodec::extensions))
{
}
WindowsMediaAudioFormat::~WindowsMediaAudioFormat() {}
Array<int> WindowsMediaAudioFormat::getPossibleSampleRates() { return Array<int>(); }
Array<int> WindowsMediaAudioFormat::getPossibleBitDepths() { return Array<int>(); }
bool WindowsMediaAudioFormat::canDoStereo() { return true; }
bool WindowsMediaAudioFormat::canDoMono() { return true; }
bool WindowsMediaAudioFormat::isCompressed() { return true; }
//==============================================================================
AudioFormatReader* WindowsMediaAudioFormat::createReaderFor (InputStream* sourceStream, bool deleteStreamIfOpeningFails)
{
ScopedPointer<WindowsMediaCodec::WMAudioReader> r (new WindowsMediaCodec::WMAudioReader (sourceStream));
if (r->sampleRate > 0)
return r.release();
if (! deleteStreamIfOpeningFails)
r->input = nullptr;
return nullptr;
}
AudioFormatWriter* WindowsMediaAudioFormat::createWriterFor (OutputStream* /*streamToWriteTo*/, double /*sampleRateToUse*/,
unsigned int /*numberOfChannels*/, int /*bitsPerSample*/,
const StringPairArray& /*metadataValues*/, int /*qualityOptionIndex*/)
{
jassertfalse; // not yet implemented!
return nullptr;
}
JuceLibraryCode/modules/juce_audio_formats/codecs/juce_WindowsMediaAudioFormat.h
+53
View File
@@ -0,0 +1,53 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2013 - Raw Material Software Ltd.
Permission is granted to use this software under the terms of either:
a) the GPL v2 (or any later version)
b) the Affero GPL v3
Details of these licenses can be found at: www.gnu.org/licenses
JUCE is distributed in the hope that it will be useful, but WITHOUT ANY
WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR
A PARTICULAR PURPOSE. See the GNU General Public License for more details.
------------------------------------------------------------------------------
To release a closed-source product which uses JUCE, commercial licenses are
available: visit www.juce.com for more information.
==============================================================================
*/
#if JUCE_WINDOWS || DOXYGEN
//==============================================================================
/**
Audio format which uses the Windows Media codecs (Windows only).
*/
class WindowsMediaAudioFormat : public AudioFormat
{
public:
//==============================================================================
WindowsMediaAudioFormat();
~WindowsMediaAudioFormat();
//==============================================================================
Array<int> getPossibleSampleRates() override;
Array<int> getPossibleBitDepths() override;
bool canDoStereo() override;
bool canDoMono() override;
bool isCompressed() override;
//==============================================================================
AudioFormatReader* createReaderFor (InputStream*, bool deleteStreamIfOpeningFails) override;
AudioFormatWriter* createWriterFor (OutputStream*, double sampleRateToUse,
unsigned int numberOfChannels, int bitsPerSample,
const StringPairArray& metadataValues, int qualityOptionIndex) override;
};
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/Ogg Vorbis Licence.txt
+47
View File
@@ -0,0 +1,47 @@
=====================================================================
I've incorporated Ogg-Vorbis directly into the Juce codebase because it makes
things much easier than having to make all your builds link correctly to
the appropriate libraries on every different platform.
I've made minimal changes to the Ogg-Vorbis code - just tweaked a few include
paths to make it build smoothly, and added some headers to allow you to exclude
it from the build.
=====================================================================
The following license is the BSD-style license that comes with the
Ogg-Vorbis distribution, and which applies just to the header files I've
included in this directory. For more info, and to get the rest of the
distribution, visit the Ogg-Vorbis homepage: www.vorbis.com
=====================================================================
Copyright (c) 2002-2004 Xiph.org Foundation
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
- Neither the name of the Xiph.org Foundation nor the names of its
contributors may be used to endorse or promote products derived from
this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION
OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/bitwise.c
+784
View File
@@ -0,0 +1,784 @@
/********************************************************************
* *
* 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 <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
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/codec.h
+242
View File
@@ -0,0 +1,242 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2001 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
********************************************************************
function: libvorbis codec headers
last mod: $Id: codec.h 17021 2010-03-24 09:29:41Z xiphmont $
********************************************************************/
#ifndef _vorbis_codec_h_
#define _vorbis_codec_h_
#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */
#include "ogg.h"
typedef struct vorbis_info{
int version;
int channels;
long rate;
/* The below bitrate declarations are *hints*.
Combinations of the three values carry the following implications:
all three set to the same value:
implies a fixed rate bitstream
only nominal set:
implies a VBR stream that averages the nominal bitrate. No hard
upper/lower limit
upper and or lower set:
implies a VBR bitstream that obeys the bitrate limits. nominal
may also be set to give a nominal rate.
none set:
the coder does not care to speculate.
*/
long bitrate_upper;
long bitrate_nominal;
long bitrate_lower;
long bitrate_window;
void *codec_setup;
} vorbis_info;
/* vorbis_dsp_state buffers the current vorbis audio
analysis/synthesis state. The DSP state belongs to a specific
logical bitstream ****************************************************/
typedef struct vorbis_dsp_state{
int analysisp;
vorbis_info *vi;
float **pcm;
float **pcmret;
int pcm_storage;
int pcm_current;
int pcm_returned;
int preextrapolate;
int eofflag;
long lW;
long W;
long nW;
long centerW;
ogg_int64_t granulepos;
ogg_int64_t sequence;
ogg_int64_t glue_bits;
ogg_int64_t time_bits;
ogg_int64_t floor_bits;
ogg_int64_t res_bits;
void *backend_state;
} vorbis_dsp_state;
typedef struct vorbis_block{
/* necessary stream state for linking to the framing abstraction */
float **pcm; /* this is a pointer into local storage */
oggpack_buffer opb;
long lW;
long W;
long nW;
int pcmend;
int mode;
int eofflag;
ogg_int64_t granulepos;
ogg_int64_t sequence;
vorbis_dsp_state *vd; /* For read-only access of configuration */
/* local storage to avoid remallocing; it's up to the mapping to
structure it */
void *localstore;
long localtop;
long localalloc;
long totaluse;
struct alloc_chain *reap;
/* bitmetrics for the frame */
long glue_bits;
long time_bits;
long floor_bits;
long res_bits;
void *internal;
} vorbis_block;
/* vorbis_block is a single block of data to be processed as part of
the analysis/synthesis stream; it belongs to a specific logical
bitstream, but is independent from other vorbis_blocks belonging to
that logical bitstream. *************************************************/
struct alloc_chain{
void *ptr;
struct alloc_chain *next;
};
/* vorbis_info contains all the setup information specific to the
specific compression/decompression mode in progress (eg,
psychoacoustic settings, channel setup, options, codebook
etc). vorbis_info and substructures are in backends.h.
*********************************************************************/
/* the comments are not part of vorbis_info so that vorbis_info can be
static storage */
typedef struct vorbis_comment{
/* unlimited user comment fields. libvorbis writes 'libvorbis'
whatever vendor is set to in encode */
char **user_comments;
int *comment_lengths;
int comments;
char *vendor;
} vorbis_comment;
/* libvorbis encodes in two abstraction layers; first we perform DSP
and produce a packet (see docs/analysis.txt). The packet is then
coded into a framed OggSquish bitstream by the second layer (see
docs/framing.txt). Decode is the reverse process; we sync/frame
the bitstream and extract individual packets, then decode the
packet back into PCM audio.
The extra framing/packetizing is used in streaming formats, such as
files. Over the net (such as with UDP), the framing and
packetization aren't necessary as they're provided by the transport
and the streaming layer is not used */
/* Vorbis PRIMITIVES: general ***************************************/
extern void vorbis_info_init(vorbis_info *vi);
extern void vorbis_info_clear(vorbis_info *vi);
extern int vorbis_info_blocksize(vorbis_info *vi,int zo);
extern void vorbis_comment_init(vorbis_comment *vc);
extern void vorbis_comment_add(vorbis_comment *vc, const char *comment);
extern void vorbis_comment_add_tag(vorbis_comment *vc,
const char *tag, const char *contents);
extern char *vorbis_comment_query(vorbis_comment *vc, const char *tag, int count);
extern int vorbis_comment_query_count(vorbis_comment *vc, const char *tag);
extern void vorbis_comment_clear(vorbis_comment *vc);
extern int vorbis_block_init(vorbis_dsp_state *v, vorbis_block *vb);
extern int vorbis_block_clear(vorbis_block *vb);
extern void vorbis_dsp_clear(vorbis_dsp_state *v);
extern double vorbis_granule_time(vorbis_dsp_state *v,
ogg_int64_t granulepos);
extern const char *vorbis_version_string(void);
/* Vorbis PRIMITIVES: analysis/DSP layer ****************************/
extern int vorbis_analysis_init(vorbis_dsp_state *v,vorbis_info *vi);
extern int vorbis_commentheader_out(vorbis_comment *vc, ogg_packet *op);
extern int vorbis_analysis_headerout(vorbis_dsp_state *v,
vorbis_comment *vc,
ogg_packet *op,
ogg_packet *op_comm,
ogg_packet *op_code);
extern float **vorbis_analysis_buffer(vorbis_dsp_state *v,int vals);
extern int vorbis_analysis_wrote(vorbis_dsp_state *v,int vals);
extern int vorbis_analysis_blockout(vorbis_dsp_state *v,vorbis_block *vb);
extern int vorbis_analysis(vorbis_block *vb,ogg_packet *op);
extern int vorbis_bitrate_addblock(vorbis_block *vb);
extern int vorbis_bitrate_flushpacket(vorbis_dsp_state *vd,
ogg_packet *op);
/* Vorbis PRIMITIVES: synthesis layer *******************************/
extern int vorbis_synthesis_idheader(ogg_packet *op);
extern int vorbis_synthesis_headerin(vorbis_info *vi,vorbis_comment *vc,
ogg_packet *op);
extern int vorbis_synthesis_init(vorbis_dsp_state *v,vorbis_info *vi);
extern int vorbis_synthesis_restart(vorbis_dsp_state *v);
extern int vorbis_synthesis(vorbis_block *vb,ogg_packet *op);
extern int vorbis_synthesis_trackonly(vorbis_block *vb,ogg_packet *op);
extern int vorbis_synthesis_blockin(vorbis_dsp_state *v,vorbis_block *vb);
extern int vorbis_synthesis_pcmout(vorbis_dsp_state *v,float ***pcm);
extern int vorbis_synthesis_lapout(vorbis_dsp_state *v,float ***pcm);
extern int vorbis_synthesis_read(vorbis_dsp_state *v,int samples);
extern long vorbis_packet_blocksize(vorbis_info *vi,ogg_packet *op);
extern int vorbis_synthesis_halfrate(vorbis_info *v,int flag);
extern int vorbis_synthesis_halfrate_p(vorbis_info *v);
/* Vorbis ERRORS and return codes ***********************************/
#define OV_FALSE -1
#define OV_EOF -2
#define OV_HOLE -3
#define OV_EREAD -128
#define OV_EFAULT -129
#define OV_EIMPL -130
#define OV_EINVAL -131
#define OV_ENOTVORBIS -132
#define OV_EBADHEADER -133
#define OV_EVERSION -134
#define OV_ENOTAUDIO -135
#define OV_EBADPACKET -136
#define OV_EBADLINK -137
#define OV_ENOSEEK -138
#ifdef __cplusplus
}
#endif /* __cplusplus */
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/config_types.h
+10
View File
@@ -0,0 +1,10 @@
#ifndef __CONFIG_TYPES_H__
#define __CONFIG_TYPES_H__
typedef int16_t ogg_int16_t;
typedef unsigned short ogg_uint16_t;
typedef int32_t ogg_int32_t;
typedef unsigned int ogg_uint32_t;
typedef int64_t ogg_int64_t;
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/framing.c
+1796
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/AUTHORS
+3
View File
@@ -0,0 +1,3 @@
Monty <monty@xiph.org>
and the rest of the Xiph.org Foundation.
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/CHANGES
+126
View File
@@ -0,0 +1,126 @@
libvorbis 1.3.2 (2010-11-01) -- "Xiph.Org libVorbis I 20101101 (Schaufenugget)"
* vorbis: additional proofing against invalid/malicious
streams in floor, residue, and bos/eos packet trimming
code (see SVN for details).
* vorbis: Added programming documentation tree for the
low-level calls
* vorbisfile: Correct handling of serial numbers array
element [0] on non-seekable streams
* vorbisenc: Back out an [old] AoTuV HF weighting that was
first enabled in 1.3.0; there are a few samples where I
really don't like the effect it causes.
* vorbis: return correct timestamp for granule positions
with high bit set.
* vorbisfile: the [undocumented] half-rate decode api made no
attempt to keep the pcm offset tracking consistent in seeks.
Fix and add a testing mode to seeking_example.c to torture
test seeking in halfrate mode. Also remove requirement that
halfrate mode only work with seekable files.
* vorbisfile: Fix a chaining bug in raw_seeks where seeking
out of the current link would fail due to not
reinitializing the decode machinery.
* vorbisfile: improve seeking strategy. Reduces the
necessary number of seek callbacks in an open or seek
operation by well over 2/3.
libvorbis 1.3.1 (2010-02-26) -- "Xiph.Org libVorbis I 20100325 (Everywhere)"
* tweak + minor arithmetic fix in floor1 fit
* revert noise norm to conservative 1.2.3 behavior pending
more listening testing
libvorbis 1.3.0 (2010-02-25) -- unreleased staging snapshot
* Optimized surround support for 5.1 encoding at 44.1/48kHz
* Added encoder control call to disable channel coupling
* Correct an overflow bug in very low-bitrate encoding on 32 bit
machines that caused inflated bitrates
* Numerous API hardening, leak and build fixes
* Correct bug in 22kHz compand setup that could cause a crash
* Correct bug in 16kHz codebooks that could cause unstable pure
tones at high bitrates
libvorbis 1.2.3 (2009-07-09) -- "Xiph.Org libVorbis I 20090709"
* correct a vorbisfile bug that prevented proper playback of
Vorbis files where all audio in a logical stream is in a
single page
* Additional decode setup hardening against malicious streams
* Add 'OV_EXCLUDE_STATIC_CALLBACKS' define for developers who
wish to avoid unused symbol warnings from the static callbacks
defined in vorbisfile.h
libvorbis 1.2.2 (2009-06-24) -- "Xiph.Org libVorbis I 20090624"
* define VENDOR and ENCODER strings
* seek correctly in files bigger than 2 GB (Windows)
* fix regression from CVE-2008-1420; 1.0b1 files work again
* mark all tables as constant to reduce memory occupation
* additional decoder hardening against malicious streams
* substantially reduce amount of seeking performed by Vorbisfile
* Multichannel decode bugfix
* build system updates
* minor specification clarifications/fixes
libvorbis 1.2.1 (unreleased) -- "Xiph.Org libVorbis I 20080501"
* Improved robustness with corrupt streams.
* New ov_read_filter() vorbisfile call allows filtering decoded
audio as floats before converting to integer samples.
* Fix an encoder bug with multichannel streams.
* Replaced RTP payload format draft with RFC 5215.
* Bare bones self test under 'make check'.
* Fix a problem encoding some streams between 14 and 28 kHz.
* Fix a numerical instability in the edge extrapolation filter.
* Build system improvements.
* Specification correction.
libvorbis 1.2.0 (2007-07-25) -- "Xiph.Org libVorbis I 20070622"
* new ov_fopen() convenience call that avoids the common
stdio conflicts with ov_open() and MSVC runtimes.
* libvorbisfile now handles multiplexed streams
* improve robustness to corrupt input streams
* fix a minor encoder bug
* updated RTP draft
* build system updates
* minor corrections to the specification
libvorbis 1.1.2 (2005-11-27) -- "Xiph.Org libVorbis I 20050304"
* fix a serious encoder bug with gcc 4 optimized builds
* documentation and spec fixes
* updated VS2003 and XCode builds
* new draft RTP encapsulation spec
libvorbis 1.1.1 (2005-06-27) -- "Xiph.Org libVorbis I 20050304"
* bug fix to the bitrate management encoder interface
* bug fix to properly set packetno field in the encoder
* new draft RTP encapsulation spec
* library API documentation improvements
libvorbis 1.1.0 (2004-09-22) -- "Xiph.Org libVorbis I 20040629"
* merges tuning improvements from Aoyumi's aoTuV with fixups
* new managed bitrate (CBR) mode support
* new vorbis_encoder_ctl() interface
* extensive documentation updates
* application/ogg mimetype is now official
* autotools cleanup from Thomas Vander Stichele
* SymbianOS build support from Colin Ward at CSIRO
* various bugfixes
* various packaging improvements
libvorbis 1.0.1 (2003-11-17) -- "Xiph.Org libVorbis I 20030909"
* numerous bug fixes
* specification corrections
* new crosslap and halfrate APIs for game use
* packaging and build updates
libvorbis 1.0.0 (2002-07-19) -- "Xiph.Org libVorbis I 20020717"
* first stable release
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/COPYING
+28
View File
@@ -0,0 +1,28 @@
Copyright (c) 2002-2008 Xiph.org Foundation
Redistribution and use in source and binary forms, with or without
modification, are permitted provided that the following conditions
are met:
- Redistributions of source code must retain the above copyright
notice, this list of conditions and the following disclaimer.
- Redistributions in binary form must reproduce the above copyright
notice, this list of conditions and the following disclaimer in the
documentation and/or other materials provided with the distribution.
- Neither the name of the Xiph.org Foundation nor the names of its
contributors may be used to endorse or promote products derived from
this software without specific prior written permission.
THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION
OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/README
+134
View File
@@ -0,0 +1,134 @@
********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 *
* by the Xiph.org Foundation, http://www.xiph.org/ *
* *
********************************************************************
Vorbis is a general purpose audio and music encoding format
contemporary to MPEG-4's AAC and TwinVQ, the next generation beyond
MPEG audio layer 3. Unlike the MPEG sponsored formats (and other
proprietary formats such as RealAudio G2 and Windows' flavor of the
month), the Vorbis CODEC specification belongs to the public domain.
All the technical details are published and documented, and any
software entity may make full use of the format without license
fee, royalty or patent concerns.
This package contains:
* libvorbis, a BSD-style license software implementation of
the Vorbis specification by the Xiph.Org Foundation
(http://www.xiph.org/)
* libvorbisfile, a BSD-style license convenience library
built on Vorbis designed to simplify common uses
* libvorbisenc, a BSD-style license library that provides a simple,
programmatic encoding setup interface
* example code making use of libogg, libvorbis, libvorbisfile and
libvorbisenc
WHAT'S HERE:
This source distribution includes libvorbis and an example
encoder/player to demonstrate use of libvorbis as well as
documentation on the Ogg Vorbis audio coding format.
You'll need libogg (distributed separately) to compile this library.
A more comprehensive set of utilities is available in the vorbis-tools
package.
Directory:
./lib The source for the libraries, a BSD-license implementation
of the public domain Ogg Vorbis audio encoding format.
./include Library API headers
./debian Rules/spec files for building Debian .deb packages
./doc Vorbis documentation
./examples Example code illustrating programmatic use of libvorbis,
libvorbisfile and libvorbisenc
./mac Codewarrior project files and build tweaks for MacOS.
./macosx Project files for MacOS X.
./win32 Win32 projects files and build automation
./vq Internal utilities for training/building new LSP/residue
and auxiliary codebooks.
CONTACT:
The Ogg homepage is located at 'http://www.xiph.org/ogg/'.
Vorbis's homepage is located at 'http://www.xiph.org/vorbis/'.
Up to date technical documents, contact information, source code and
pre-built utilities may be found there.
The user website for Ogg Vorbis software and audio is http://vorbis.com/
BUILDING FROM TRUNK:
Development source is under subversion revision control at
https://svn.xiph.org/trunk/vorbis/. You will also need the
newest versions of autoconf, automake, libtool and pkg-config in
order to compile Vorbis from development source. A configure script
is provided for you in the source tarball distributions.
[update or checkout latest source]
./autogen.sh
make
and as root if desired:
make install
This will install the Vorbis libraries (static and shared) into
/usr/local/lib, includes into /usr/local/include and API manpages
(once we write some) into /usr/local/man.
Documentation building requires xsltproc and pdfxmltex.
BUILDING FROM TARBALL DISTRIBUTIONS:
./configure
make
and optionally (as root):
make install
BUILDING RPMS:
after normal configuring:
make dist
rpm -ta libvorbis-<version>.tar.gz
BUILDING ON MACOS 9:
Vorbis on MacOS 9 is built using Metroworks CodeWarrior. To build it,
first verify that the Ogg libraries are already built following the
instructions in the Ogg module README. Open vorbis/mac/libvorbis.mcp,
switch to the "Targets" pane, select everything, and make the project.
Do the same thing to build libvorbisenc.mcp, and libvorbisfile.mcp (in
that order). In vorbis/mac/Output you will now have both debug and final
versions of Vorbis shared libraries to link your projects against.
To build a project using Ogg Vorbis, add access paths to your
CodeWarrior project for the ogg/include, ogg/mac/Output,
vorbis/include, and vorbis/mac/Output folders. Be sure that
"interpret DOS and Unix paths" is turned on in your project; it can
be found in the "access paths" pane in your project settings. Now
simply add the shared libraries you need to your project (OggLib and
VorbisLib at least) and #include "ogg/ogg.h" and "vorbis/codec.h"
wherever you need to access Ogg and Vorbis functionality.
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/analysis.c
+109
View File
@@ -0,0 +1,109 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: single-block PCM analysis mode dispatch
last mod: $Id: analysis.c 16226 2009-07-08 06:43:49Z xiphmont $
********************************************************************/
#include <stdio.h>
#include <string.h>
#include <math.h>
#include "../../ogg.h"
#include "../../codec.h"
#include "codec_internal.h"
#include "registry.h"
#include "scales.h"
#include "os.h"
#include "misc.h"
/* decides between modes, dispatches to the appropriate mapping. */
int vorbis_analysis(vorbis_block *vb, ogg_packet *op){
int ret,i;
vorbis_block_internal *vbi=(vorbis_block_internal *)vb->internal;
vb->glue_bits=0;
vb->time_bits=0;
vb->floor_bits=0;
vb->res_bits=0;
/* first things first. Make sure encode is ready */
for(i=0;i<PACKETBLOBS;i++)
oggpack_reset(vbi->packetblob[i]);
/* we only have one mapping type (0), and we let the mapping code
itself figure out what soft mode to use. This allows easier
bitrate management */
if((ret=_mapping_P[0]->forward(vb)))
return(ret);
if(op){
if(vorbis_bitrate_managed(vb))
/* The app is using a bitmanaged mode... but not using the
bitrate management interface. */
return(OV_EINVAL);
op->packet=oggpack_get_buffer(&vb->opb);
op->bytes=oggpack_bytes(&vb->opb);
op->b_o_s=0;
op->e_o_s=vb->eofflag;
op->granulepos=vb->granulepos;
op->packetno=vb->sequence; /* for sake of completeness */
}
return(0);
}
#ifdef ANALYSIS
int analysis_noisy=1;
/* there was no great place to put this.... */
void _analysis_output_always(char *base,int i,float *v,int n,int bark,int dB,ogg_int64_t off){
int j;
FILE *of;
char buffer[80];
sprintf(buffer,"%s_%d.m",base,i);
of=fopen(buffer,"w");
if(!of)perror("failed to open data dump file");
for(j=0;j<n;j++){
if(bark){
float b=toBARK((4000.f*j/n)+.25);
fprintf(of,"%f ",b);
}else
if(off!=0)
fprintf(of,"%f ",(double)(j+off)/8000.);
else
fprintf(of,"%f ",(double)j);
if(dB){
float val;
if(v[j]==0.)
val=-140.;
else
val=todB(v+j);
fprintf(of,"%f\n",val);
}else{
fprintf(of,"%f\n",v[j]);
}
}
fclose(of);
}
void _analysis_output(char *base,int i,float *v,int n,int bark,int dB,
ogg_int64_t off){
if(analysis_noisy)_analysis_output_always(base,i,v,n,bark,dB,off);
}
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/backends.h
+144
View File
@@ -0,0 +1,144 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: libvorbis backend and mapping structures; needed for
static mode headers
last mod: $Id: backends.h 16962 2010-03-11 07:30:34Z xiphmont $
********************************************************************/
/* this is exposed up here because we need it for static modes.
Lookups for each backend aren't exposed because there's no reason
to do so */
#ifndef _vorbis_backend_h_
#define _vorbis_backend_h_
#include "codec_internal.h"
/* this would all be simpler/shorter with templates, but.... */
/* Floor backend generic *****************************************/
typedef struct{
void (*pack) (vorbis_info_floor *,oggpack_buffer *);
vorbis_info_floor *(*unpack)(vorbis_info *,oggpack_buffer *);
vorbis_look_floor *(*look) (vorbis_dsp_state *,vorbis_info_floor *);
void (*free_info) (vorbis_info_floor *);
void (*free_look) (vorbis_look_floor *);
void *(*inverse1) (struct vorbis_block *,vorbis_look_floor *);
int (*inverse2) (struct vorbis_block *,vorbis_look_floor *,
void *buffer,float *);
} vorbis_func_floor;
typedef struct{
int order;
long rate;
long barkmap;
int ampbits;
int ampdB;
int numbooks; /* <= 16 */
int books[16];
float lessthan; /* encode-only config setting hacks for libvorbis */
float greaterthan; /* encode-only config setting hacks for libvorbis */
} vorbis_info_floor0;
#define VIF_POSIT 63
#define VIF_CLASS 16
#define VIF_PARTS 31
typedef struct{
int partitions; /* 0 to 31 */
int partitionclass[VIF_PARTS]; /* 0 to 15 */
int class_dim[VIF_CLASS]; /* 1 to 8 */
int class_subs[VIF_CLASS]; /* 0,1,2,3 (bits: 1<<n poss) */
int class_book[VIF_CLASS]; /* subs ^ dim entries */
int class_subbook[VIF_CLASS][8]; /* [VIF_CLASS][subs] */
int mult; /* 1 2 3 or 4 */
int postlist[VIF_POSIT+2]; /* first two implicit */
/* encode side analysis parameters */
float maxover;
float maxunder;
float maxerr;
float twofitweight;
float twofitatten;
int n;
} vorbis_info_floor1;
/* Residue backend generic *****************************************/
typedef struct{
void (*pack) (vorbis_info_residue *,oggpack_buffer *);
vorbis_info_residue *(*unpack)(vorbis_info *,oggpack_buffer *);
vorbis_look_residue *(*look) (vorbis_dsp_state *,
vorbis_info_residue *);
void (*free_info) (vorbis_info_residue *);
void (*free_look) (vorbis_look_residue *);
long **(*classx) (struct vorbis_block *,vorbis_look_residue *,
int **,int *,int);
int (*forward) (oggpack_buffer *,struct vorbis_block *,
vorbis_look_residue *,
int **,int *,int,long **,int);
int (*inverse) (struct vorbis_block *,vorbis_look_residue *,
float **,int *,int);
} vorbis_func_residue;
typedef struct vorbis_info_residue0{
/* block-partitioned VQ coded straight residue */
long begin;
long end;
/* first stage (lossless partitioning) */
int grouping; /* group n vectors per partition */
int partitions; /* possible codebooks for a partition */
int partvals; /* partitions ^ groupbook dim */
int groupbook; /* huffbook for partitioning */
int secondstages[64]; /* expanded out to pointers in lookup */
int booklist[512]; /* list of second stage books */
/*const*/ int classmetric1[64];
/*const*/ int classmetric2[64];
} vorbis_info_residue0;
/* Mapping backend generic *****************************************/
typedef struct{
void (*pack) (vorbis_info *,vorbis_info_mapping *,
oggpack_buffer *);
vorbis_info_mapping *(*unpack)(vorbis_info *,oggpack_buffer *);
void (*free_info) (vorbis_info_mapping *);
int (*forward) (struct vorbis_block *vb);
int (*inverse) (struct vorbis_block *vb,vorbis_info_mapping *);
} vorbis_func_mapping;
typedef struct vorbis_info_mapping0{
int submaps; /* <= 16 */
int chmuxlist[256]; /* up to 256 channels in a Vorbis stream */
int floorsubmap[16]; /* [mux] submap to floors */
int residuesubmap[16]; /* [mux] submap to residue */
int coupling_steps;
int coupling_mag[256];
int coupling_ang[256];
} vorbis_info_mapping0;
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/bitrate.c
+253
View File
@@ -0,0 +1,253 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: bitrate tracking and management
last mod: $Id: bitrate.c 16227 2009-07-08 06:58:46Z xiphmont $
********************************************************************/
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include "../../ogg.h"
#include "../../codec.h"
#include "codec_internal.h"
#include "os.h"
#include "misc.h"
#include "bitrate.h"
/* compute bitrate tracking setup */
void vorbis_bitrate_init(vorbis_info *vi,bitrate_manager_state *bm){
codec_setup_info *ci=(codec_setup_info *)vi->codec_setup;
bitrate_manager_info *bi=&ci->bi;
memset(bm,0,sizeof(*bm));
if(bi && (bi->reservoir_bits>0)){
long ratesamples=vi->rate;
int halfsamples=ci->blocksizes[0]>>1;
bm->short_per_long=ci->blocksizes[1]/ci->blocksizes[0];
bm->managed=1;
bm->avg_bitsper= (int) rint(1.*bi->avg_rate*halfsamples/ratesamples);
bm->min_bitsper= (int) rint(1.*bi->min_rate*halfsamples/ratesamples);
bm->max_bitsper= (int) rint(1.*bi->max_rate*halfsamples/ratesamples);
bm->avgfloat=PACKETBLOBS/2;
/* not a necessary fix, but one that leads to a more balanced
typical initialization */
{
long desired_fill = (long) (bi->reservoir_bits*bi->reservoir_bias);
bm->minmax_reservoir=desired_fill;
bm->avg_reservoir=desired_fill;
}
}
}
void vorbis_bitrate_clear(bitrate_manager_state *bm){
memset(bm,0,sizeof(*bm));
return;
}
int vorbis_bitrate_managed(vorbis_block *vb){
vorbis_dsp_state *vd=vb->vd;
private_state *b=(private_state*)vd->backend_state;
bitrate_manager_state *bm=&b->bms;
if(bm && bm->managed)return(1);
return(0);
}
/* finish taking in the block we just processed */
int vorbis_bitrate_addblock(vorbis_block *vb){
vorbis_block_internal *vbi=(vorbis_block_internal*)vb->internal;
vorbis_dsp_state *vd=vb->vd;
private_state *b=(private_state*)vd->backend_state;
bitrate_manager_state *bm=&b->bms;
vorbis_info *vi=vd->vi;
codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
bitrate_manager_info *bi=&ci->bi;
int choice = (int) rint(bm->avgfloat);
long this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
long min_target_bits=(vb->W?bm->min_bitsper*bm->short_per_long:bm->min_bitsper);
long max_target_bits=(vb->W?bm->max_bitsper*bm->short_per_long:bm->max_bitsper);
int samples=ci->blocksizes[vb->W]>>1;
long desired_fill = (long) (bi->reservoir_bits*bi->reservoir_bias);
if(!bm->managed){
/* not a bitrate managed stream, but for API simplicity, we'll
buffer the packet to keep the code path clean */
if(bm->vb)return(-1); /* one has been submitted without
being claimed */
bm->vb=vb;
return(0);
}
bm->vb=vb;
/* look ahead for avg floater */
if(bm->avg_bitsper>0){
double slew=0.;
long avg_target_bits=(vb->W?bm->avg_bitsper*bm->short_per_long:bm->avg_bitsper);
double slewlimit= 15./bi->slew_damp;
/* choosing a new floater:
if we're over target, we slew down
if we're under target, we slew up
choose slew as follows: look through packetblobs of this frame
and set slew as the first in the appropriate direction that
gives us the slew we want. This may mean no slew if delta is
already favorable.
Then limit slew to slew max */
if(bm->avg_reservoir+(this_bits-avg_target_bits)>desired_fill){
while(choice>0 && this_bits>avg_target_bits &&
bm->avg_reservoir+(this_bits-avg_target_bits)>desired_fill){
choice--;
this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
}
}else if(bm->avg_reservoir+(this_bits-avg_target_bits)<desired_fill){
while(choice+1<PACKETBLOBS && this_bits<avg_target_bits &&
bm->avg_reservoir+(this_bits-avg_target_bits)<desired_fill){
choice++;
this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
}
}
slew=rint(choice-bm->avgfloat)/samples*vi->rate;
if(slew<-slewlimit)slew=-slewlimit;
if(slew>slewlimit)slew=slewlimit;
choice = (int) rint(bm->avgfloat+= slew/vi->rate*samples);
this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
}
/* enforce min(if used) on the current floater (if used) */
if(bm->min_bitsper>0){
/* do we need to force the bitrate up? */
if(this_bits<min_target_bits){
while(bm->minmax_reservoir-(min_target_bits-this_bits)<0){
choice++;
if(choice>=PACKETBLOBS)break;
this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
}
}
}
/* enforce max (if used) on the current floater (if used) */
if(bm->max_bitsper>0){
/* do we need to force the bitrate down? */
if(this_bits>max_target_bits){
while(bm->minmax_reservoir+(this_bits-max_target_bits)>bi->reservoir_bits){
choice--;
if(choice<0)break;
this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
}
}
}
/* Choice of packetblobs now made based on floater, and min/max
requirements. Now boundary check extreme choices */
if(choice<0){
/* choosing a smaller packetblob is insufficient to trim bitrate.
frame will need to be truncated */
long maxsize=(max_target_bits+(bi->reservoir_bits-bm->minmax_reservoir))/8;
bm->choice=choice=0;
if(oggpack_bytes(vbi->packetblob[choice])>maxsize){
oggpack_writetrunc(vbi->packetblob[choice],maxsize*8);
this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
}
}else{
long minsize=(min_target_bits-bm->minmax_reservoir+7)/8;
if(choice>=PACKETBLOBS)
choice=PACKETBLOBS-1;
bm->choice=choice;
/* prop up bitrate according to demand. pad this frame out with zeroes */
minsize-=oggpack_bytes(vbi->packetblob[choice]);
while(minsize-->0)oggpack_write(vbi->packetblob[choice],0,8);
this_bits=oggpack_bytes(vbi->packetblob[choice])*8;
}
/* now we have the final packet and the final packet size. Update statistics */
/* min and max reservoir */
if(bm->min_bitsper>0 || bm->max_bitsper>0){
if(max_target_bits>0 && this_bits>max_target_bits){
bm->minmax_reservoir+=(this_bits-max_target_bits);
}else if(min_target_bits>0 && this_bits<min_target_bits){
bm->minmax_reservoir+=(this_bits-min_target_bits);
}else{
/* inbetween; we want to take reservoir toward but not past desired_fill */
if(bm->minmax_reservoir>desired_fill){
if(max_target_bits>0){ /* logical bulletproofing against initialization state */
bm->minmax_reservoir+=(this_bits-max_target_bits);
if(bm->minmax_reservoir<desired_fill)bm->minmax_reservoir=desired_fill;
}else{
bm->minmax_reservoir=desired_fill;
}
}else{
if(min_target_bits>0){ /* logical bulletproofing against initialization state */
bm->minmax_reservoir+=(this_bits-min_target_bits);
if(bm->minmax_reservoir>desired_fill)bm->minmax_reservoir=desired_fill;
}else{
bm->minmax_reservoir=desired_fill;
}
}
}
}
/* avg reservoir */
if(bm->avg_bitsper>0){
long avg_target_bits=(vb->W?bm->avg_bitsper*bm->short_per_long:bm->avg_bitsper);
bm->avg_reservoir+=this_bits-avg_target_bits;
}
return(0);
}
int vorbis_bitrate_flushpacket(vorbis_dsp_state *vd,ogg_packet *op){
private_state *b=(private_state*)vd->backend_state;
bitrate_manager_state *bm=&b->bms;
vorbis_block *vb=bm->vb;
int choice=PACKETBLOBS/2;
if(!vb)return 0;
if(op){
vorbis_block_internal *vbi=(vorbis_block_internal*)vb->internal;
if(vorbis_bitrate_managed(vb))
choice=bm->choice;
op->packet=oggpack_get_buffer(vbi->packetblob[choice]);
op->bytes=oggpack_bytes(vbi->packetblob[choice]);
op->b_o_s=0;
op->e_o_s=vb->eofflag;
op->granulepos=vb->granulepos;
op->packetno=vb->sequence; /* for sake of completeness */
}
bm->vb=0;
return(1);
}
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/bitrate.h
+59
View File
@@ -0,0 +1,59 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: bitrate tracking and management
last mod: $Id: bitrate.h 13293 2007-07-24 00:09:47Z xiphmont $
********************************************************************/
#ifndef _V_BITRATE_H_
#define _V_BITRATE_H_
#include "../../codec.h"
#include "codec_internal.h"
#include "os.h"
/* encode side bitrate tracking */
typedef struct bitrate_manager_state {
int managed;
long avg_reservoir;
long minmax_reservoir;
long avg_bitsper;
long min_bitsper;
long max_bitsper;
long short_per_long;
double avgfloat;
vorbis_block *vb;
int choice;
} bitrate_manager_state;
typedef struct bitrate_manager_info{
long avg_rate;
long min_rate;
long max_rate;
long reservoir_bits;
double reservoir_bias;
double slew_damp;
} bitrate_manager_info;
extern void vorbis_bitrate_init(vorbis_info *vi,bitrate_manager_state *bs);
extern void vorbis_bitrate_clear(bitrate_manager_state *bs);
extern int vorbis_bitrate_managed(vorbis_block *vb);
extern int vorbis_bitrate_addblock(vorbis_block *vb);
extern int vorbis_bitrate_flushpacket(vorbis_dsp_state *vd, ogg_packet *op);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/block.c
+1033
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/books/coupled/res_books_51.h
+12256
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/books/coupled/res_books_stereo.h
+15782
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/books/floor/floor_books.h
+1546
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/books/uncoupled/res_books_uncoupled.h
+7757
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/codebook.c
+479
View File
@@ -0,0 +1,479 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: basic codebook pack/unpack/code/decode operations
last mod: $Id: codebook.c 17553 2010-10-21 17:54:26Z tterribe $
********************************************************************/
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include "../../ogg.h"
#include "../../codec.h"
#include "codebook.h"
#include "scales.h"
#include "misc.h"
#include "os.h"
/* packs the given codebook into the bitstream **************************/
int vorbis_staticbook_pack(const static_codebook *c,oggpack_buffer *opb){
long i,j;
int ordered=0;
/* first the basic parameters */
oggpack_write(opb,0x564342,24);
oggpack_write(opb,c->dim,16);
oggpack_write(opb,c->entries,24);
/* pack the codewords. There are two packings; length ordered and
length random. Decide between the two now. */
for(i=1;i<c->entries;i++)
if(c->lengthlist[i-1]==0 || c->lengthlist[i]<c->lengthlist[i-1])break;
if(i==c->entries)ordered=1;
if(ordered){
/* length ordered. We only need to say how many codewords of
each length. The actual codewords are generated
deterministically */
long count=0;
oggpack_write(opb,1,1); /* ordered */
oggpack_write(opb,c->lengthlist[0]-1,5); /* 1 to 32 */
for(i=1;i<c->entries;i++){
long thisx=c->lengthlist[i];
long last=c->lengthlist[i-1];
if(thisx>last){
for(j=last;j<thisx;j++){
oggpack_write(opb,i-count,_ilog(c->entries-count));
count=i;
}
}
}
oggpack_write(opb,i-count,_ilog(c->entries-count));
}else{
/* length random. Again, we don't code the codeword itself, just
the length. This time, though, we have to encode each length */
oggpack_write(opb,0,1); /* unordered */
/* algortihmic mapping has use for 'unused entries', which we tag
here. The algorithmic mapping happens as usual, but the unused
entry has no codeword. */
for(i=0;i<c->entries;i++)
if(c->lengthlist[i]==0)break;
if(i==c->entries){
oggpack_write(opb,0,1); /* no unused entries */
for(i=0;i<c->entries;i++)
oggpack_write(opb,c->lengthlist[i]-1,5);
}else{
oggpack_write(opb,1,1); /* we have unused entries; thus we tag */
for(i=0;i<c->entries;i++){
if(c->lengthlist[i]==0){
oggpack_write(opb,0,1);
}else{
oggpack_write(opb,1,1);
oggpack_write(opb,c->lengthlist[i]-1,5);
}
}
}
}
/* is the entry number the desired return value, or do we have a
mapping? If we have a mapping, what type? */
oggpack_write(opb,c->maptype,4);
switch(c->maptype){
case 0:
/* no mapping */
break;
case 1:case 2:
/* implicitly populated value mapping */
/* explicitly populated value mapping */
if(!c->quantlist){
/* no quantlist? error */
return(-1);
}
/* values that define the dequantization */
oggpack_write(opb,c->q_min,32);
oggpack_write(opb,c->q_delta,32);
oggpack_write(opb,c->q_quant-1,4);
oggpack_write(opb,c->q_sequencep,1);
{
int quantvals;
switch(c->maptype){
case 1:
/* a single column of (c->entries/c->dim) quantized values for
building a full value list algorithmically (square lattice) */
quantvals=_book_maptype1_quantvals(c);
break;
case 2:
/* every value (c->entries*c->dim total) specified explicitly */
quantvals=c->entries*c->dim;
break;
default: /* NOT_REACHABLE */
quantvals=-1;
}
/* quantized values */
for(i=0;i<quantvals;i++)
oggpack_write(opb,labs(c->quantlist[i]),c->q_quant);
}
break;
default:
/* error case; we don't have any other map types now */
return(-1);
}
return(0);
}
/* unpacks a codebook from the packet buffer into the codebook struct,
readies the codebook auxiliary structures for decode *************/
static_codebook *vorbis_staticbook_unpack(oggpack_buffer *opb){
long i,j;
static_codebook *s=(static_codebook*)_ogg_calloc(1,sizeof(*s));
s->allocedp=1;
/* make sure alignment is correct */
if(oggpack_read(opb,24)!=0x564342)goto _eofout;
/* first the basic parameters */
s->dim=oggpack_read(opb,16);
s->entries=oggpack_read(opb,24);
if(s->entries==-1)goto _eofout;
if(_ilog(s->dim)+_ilog(s->entries)>24)goto _eofout;
/* codeword ordering.... length ordered or unordered? */
switch((int)oggpack_read(opb,1)){
case 0:{
long unused;
/* allocated but unused entries? */
unused=oggpack_read(opb,1);
if((s->entries*(unused?1:5)+7)>>3>opb->storage-oggpack_bytes(opb))
goto _eofout;
/* unordered */
s->lengthlist=(long*)_ogg_malloc(sizeof(*s->lengthlist)*s->entries);
/* allocated but unused entries? */
if(unused){
/* yes, unused entries */
for(i=0;i<s->entries;i++){
if(oggpack_read(opb,1)){
long num=oggpack_read(opb,5);
if(num==-1)goto _eofout;
s->lengthlist[i]=num+1;
}else
s->lengthlist[i]=0;
}
}else{
/* all entries used; no tagging */
for(i=0;i<s->entries;i++){
long num=oggpack_read(opb,5);
if(num==-1)goto _eofout;
s->lengthlist[i]=num+1;
}
}
break;
}
case 1:
/* ordered */
{
long length=oggpack_read(opb,5)+1;
if(length==0)goto _eofout;
s->lengthlist=(long*)_ogg_malloc(sizeof(*s->lengthlist)*s->entries);
for(i=0;i<s->entries;){
long num=oggpack_read(opb,_ilog(s->entries-i));
if(num==-1)goto _eofout;
if(length>32 || num>s->entries-i ||
(num>0 && (num-1)>>(length-1)>1)){
goto _errout;
}
if(length>32)goto _errout;
for(j=0;j<num;j++,i++)
s->lengthlist[i]=length;
length++;
}
}
break;
default:
/* EOF */
goto _eofout;
}
/* Do we have a mapping to unpack? */
switch((s->maptype=oggpack_read(opb,4))){
case 0:
/* no mapping */
break;
case 1: case 2:
/* implicitly populated value mapping */
/* explicitly populated value mapping */
s->q_min=oggpack_read(opb,32);
s->q_delta=oggpack_read(opb,32);
s->q_quant=oggpack_read(opb,4)+1;
s->q_sequencep=oggpack_read(opb,1);
if(s->q_sequencep==-1)goto _eofout;
{
int quantvals=0;
switch(s->maptype){
case 1:
quantvals=(s->dim==0?0:_book_maptype1_quantvals(s));
break;
case 2:
quantvals=s->entries*s->dim;
break;
}
/* quantized values */
if(((quantvals * s->q_quant + 7) >> 3) > opb->storage-oggpack_bytes(opb))
goto _eofout;
s->quantlist=(long*)_ogg_malloc(sizeof(*s->quantlist)*quantvals);
for(i=0;i<quantvals;i++)
s->quantlist[i]=oggpack_read(opb,s->q_quant);
if(quantvals&&s->quantlist[quantvals-1]==-1)goto _eofout;
}
break;
default:
goto _errout;
}
/* all set */
return(s);
_errout:
_eofout:
vorbis_staticbook_destroy(s);
return(NULL);
}
/* returns the number of bits ************************************************/
int vorbis_book_encode(codebook *book, int a, oggpack_buffer *b){
if(a<0 || a>=book->c->entries)return(0);
oggpack_write(b,book->codelist[a],book->c->lengthlist[a]);
return(book->c->lengthlist[a]);
}
/* the 'eliminate the decode tree' optimization actually requires the
codewords to be MSb first, not LSb. This is an annoying inelegancy
(and one of the first places where carefully thought out design
turned out to be wrong; Vorbis II and future Ogg codecs should go
to an MSb bitpacker), but not actually the huge hit it appears to
be. The first-stage decode table catches most words so that
bitreverse is not in the main execution path. */
static ogg_uint32_t bitreverse(ogg_uint32_t x){
x= ((x>>16)&0x0000ffff) | ((x<<16)&0xffff0000);
x= ((x>> 8)&0x00ff00ff) | ((x<< 8)&0xff00ff00);
x= ((x>> 4)&0x0f0f0f0f) | ((x<< 4)&0xf0f0f0f0);
x= ((x>> 2)&0x33333333) | ((x<< 2)&0xcccccccc);
return((x>> 1)&0x55555555) | ((x<< 1)&0xaaaaaaaa);
}
STIN long decode_packed_entry_number(codebook *book, oggpack_buffer *b){
int read=book->dec_maxlength;
long lo,hi;
long lok = oggpack_look(b,book->dec_firsttablen);
if (lok >= 0) {
long entry = book->dec_firsttable[lok];
if(entry&0x80000000UL){
lo=(entry>>15)&0x7fff;
hi=book->used_entries-(entry&0x7fff);
}else{
oggpack_adv(b, book->dec_codelengths[entry-1]);
return(entry-1);
}
}else{
lo=0;
hi=book->used_entries;
}
lok = oggpack_look(b, read);
while(lok<0 && read>1)
lok = oggpack_look(b, --read);
if(lok<0)return -1;
/* bisect search for the codeword in the ordered list */
{
ogg_uint32_t testword=bitreverse((ogg_uint32_t)lok);
while(hi-lo>1){
long p=(hi-lo)>>1;
long test=book->codelist[lo+p]>testword;
lo+=p&(test-1);
hi-=p&(-test);
}
if(book->dec_codelengths[lo]<=read){
oggpack_adv(b, book->dec_codelengths[lo]);
return(lo);
}
}
oggpack_adv(b, read);
return(-1);
}
/* Decode side is specced and easier, because we don't need to find
matches using different criteria; we simply read and map. There are
two things we need to do 'depending':
We may need to support interleave. We don't really, but it's
convenient to do it here rather than rebuild the vector later.
Cascades may be additive or multiplicitive; this is not inherent in
the codebook, but set in the code using the codebook. Like
interleaving, it's easiest to do it here.
addmul==0 -> declarative (set the value)
addmul==1 -> additive
addmul==2 -> multiplicitive */
/* returns the [original, not compacted] entry number or -1 on eof *********/
long vorbis_book_decode(codebook *book, oggpack_buffer *b){
if(book->used_entries>0){
long packed_entry=decode_packed_entry_number(book,b);
if(packed_entry>=0)
return(book->dec_index[packed_entry]);
}
/* if there's no dec_index, the codebook unpacking isn't collapsed */
return(-1);
}
/* returns 0 on OK or -1 on eof *************************************/
long vorbis_book_decodevs_add(codebook *book,float *a,oggpack_buffer *b,int n){
if(book->used_entries>0){
int step=n/book->dim;
long *entry = (long*)alloca(sizeof(*entry)*step);
float **t = (float**)alloca(sizeof(*t)*step);
int i,j,o;
for (i = 0; i < step; i++) {
entry[i]=decode_packed_entry_number(book,b);
if(entry[i]==-1)return(-1);
t[i] = book->valuelist+entry[i]*book->dim;
}
for(i=0,o=0;i<book->dim;i++,o+=step)
for (j=0;j<step;j++)
a[o+j]+=t[j][i];
}
return(0);
}
long vorbis_book_decodev_add(codebook *book,float *a,oggpack_buffer *b,int n){
if(book->used_entries>0){
int i,j,entry;
float *t;
if(book->dim>8){
for(i=0;i<n;){
entry = decode_packed_entry_number(book,b);
if(entry==-1)return(-1);
t = book->valuelist+entry*book->dim;
for (j=0;j<book->dim;)
a[i++]+=t[j++];
}
}else{
for(i=0;i<n;){
entry = decode_packed_entry_number(book,b);
if(entry==-1)return(-1);
t = book->valuelist+entry*book->dim;
j=0;
switch((int)book->dim){
case 8:
a[i++]+=t[j++];
case 7:
a[i++]+=t[j++];
case 6:
a[i++]+=t[j++];
case 5:
a[i++]+=t[j++];
case 4:
a[i++]+=t[j++];
case 3:
a[i++]+=t[j++];
case 2:
a[i++]+=t[j++];
case 1:
a[i++]+=t[j++];
case 0:
break;
}
}
}
}
return(0);
}
long vorbis_book_decodev_set(codebook *book,float *a,oggpack_buffer *b,int n){
if(book->used_entries>0){
int i,j,entry;
float *t;
for(i=0;i<n;){
entry = decode_packed_entry_number(book,b);
if(entry==-1)return(-1);
t = book->valuelist+entry*book->dim;
for (j=0;j<book->dim;)
a[i++]=t[j++];
}
}else{
int i,j;
for(i=0;i<n;){
for (j=0;j<book->dim;)
a[i++]=0.f;
}
}
return(0);
}
long vorbis_book_decodevv_add(codebook *book,float **a,long offset,int ch,
oggpack_buffer *b,int n){
long i,j,entry;
int chptr=0;
if(book->used_entries>0){
for(i=offset/ch;i<(offset+n)/ch;){
entry = decode_packed_entry_number(book,b);
if(entry==-1)return(-1);
{
const float *t = book->valuelist+entry*book->dim;
for (j=0;j<book->dim;j++){
a[chptr++][i]+=t[j];
if(chptr==ch){
chptr=0;
i++;
}
}
}
}
}
return(0);
}
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/codebook.h
+119
View File
@@ -0,0 +1,119 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: basic shared codebook operations
last mod: $Id: codebook.h 17030 2010-03-25 06:52:55Z xiphmont $
********************************************************************/
#ifndef _V_CODEBOOK_H_
#define _V_CODEBOOK_H_
#include "../../ogg.h"
/* This structure encapsulates huffman and VQ style encoding books; it
doesn't do anything specific to either.
valuelist/quantlist are nonNULL (and q_* significant) only if
there's entry->value mapping to be done.
If encode-side mapping must be done (and thus the entry needs to be
hunted), the auxiliary encode pointer will point to a decision
tree. This is true of both VQ and huffman, but is mostly useful
with VQ.
*/
typedef struct static_codebook{
long dim; /* codebook dimensions (elements per vector) */
long entries; /* codebook entries */
long *lengthlist; /* codeword lengths in bits */
/* mapping ***************************************************************/
int maptype; /* 0=none
1=implicitly populated values from map column
2=listed arbitrary values */
/* The below does a linear, single monotonic sequence mapping. */
long q_min; /* packed 32 bit float; quant value 0 maps to minval */
long q_delta; /* packed 32 bit float; val 1 - val 0 == delta */
int q_quant; /* bits: 0 < quant <= 16 */
int q_sequencep; /* bitflag */
long *quantlist; /* map == 1: (int)(entries^(1/dim)) element column map
map == 2: list of dim*entries quantized entry vals
*/
int allocedp;
} static_codebook;
typedef struct codebook{
long dim; /* codebook dimensions (elements per vector) */
long entries; /* codebook entries */
long used_entries; /* populated codebook entries */
const static_codebook *c;
/* for encode, the below are entry-ordered, fully populated */
/* for decode, the below are ordered by bitreversed codeword and only
used entries are populated */
float *valuelist; /* list of dim*entries actual entry values */
ogg_uint32_t *codelist; /* list of bitstream codewords for each entry */
int *dec_index; /* only used if sparseness collapsed */
char *dec_codelengths;
ogg_uint32_t *dec_firsttable;
int dec_firsttablen;
int dec_maxlength;
/* The current encoder uses only centered, integer-only lattice books. */
int quantvals;
int minval;
int delta;
} codebook;
extern void vorbis_staticbook_destroy(static_codebook *b);
extern int vorbis_book_init_encode(codebook *dest,const static_codebook *source);
extern int vorbis_book_init_decode(codebook *dest,const static_codebook *source);
extern void vorbis_book_clear(codebook *b);
extern float *_book_unquantize(const static_codebook *b,int n,int *map);
extern float *_book_logdist(const static_codebook *b,float *vals);
extern float _float32_unpack(long val);
extern long _float32_pack(float val);
extern int _best(codebook *book, float *a, int step);
extern int _ilog(unsigned int v);
extern long _book_maptype1_quantvals(const static_codebook *b);
extern int vorbis_book_besterror(codebook *book,float *a,int step,int addmul);
extern long vorbis_book_codeword(codebook *book,int entry);
extern long vorbis_book_codelen(codebook *book,int entry);
extern int vorbis_staticbook_pack(const static_codebook *c,oggpack_buffer *b);
extern static_codebook *vorbis_staticbook_unpack(oggpack_buffer *b);
extern int vorbis_book_encode(codebook *book, int a, oggpack_buffer *b);
extern long vorbis_book_decode(codebook *book, oggpack_buffer *b);
extern long vorbis_book_decodevs_add(codebook *book, float *a,
oggpack_buffer *b,int n);
extern long vorbis_book_decodev_set(codebook *book, float *a,
oggpack_buffer *b,int n);
extern long vorbis_book_decodev_add(codebook *book, float *a,
oggpack_buffer *b,int n);
extern long vorbis_book_decodevv_add(codebook *book, float **a,
long off,int ch,
oggpack_buffer *b,int n);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/codec_internal.h
+187
View File
@@ -0,0 +1,187 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: libvorbis codec headers
last mod: $Id: codec_internal.h 16227 2009-07-08 06:58:46Z xiphmont $
********************************************************************/
#ifndef _V_CODECI_H_
#define _V_CODECI_H_
#include "envelope.h"
#include "codebook.h"
#define BLOCKTYPE_IMPULSE 0
#define BLOCKTYPE_PADDING 1
#define BLOCKTYPE_TRANSITION 0
#define BLOCKTYPE_LONG 1
#define PACKETBLOBS 15
typedef struct vorbis_block_internal{
float **pcmdelay; /* this is a pointer into local storage */
float ampmax;
int blocktype;
oggpack_buffer *packetblob[PACKETBLOBS]; /* initialized, must be freed;
blob [PACKETBLOBS/2] points to
the oggpack_buffer in the
main vorbis_block */
} vorbis_block_internal;
typedef void vorbis_look_floor;
typedef void vorbis_look_residue;
typedef void vorbis_look_transform;
/* mode ************************************************************/
typedef struct {
int blockflag;
int windowtype;
int transformtype;
int mapping;
} vorbis_info_mode;
typedef void vorbis_info_floor;
typedef void vorbis_info_residue;
typedef void vorbis_info_mapping;
#include "psy.h"
#include "bitrate.h"
static int ilog(unsigned int v){
int ret=0;
while(v){
ret++;
v>>=1;
}
return(ret);
}
static int ilog2(unsigned int v){
int ret=0;
if(v)--v;
while(v){
ret++;
v>>=1;
}
return(ret);
}
typedef struct private_state {
/* local lookup storage */
envelope_lookup *ve; /* envelope lookup */
int window[2];
vorbis_look_transform **transform[2]; /* block, type */
drft_lookup fft_look[2];
int modebits;
vorbis_look_floor **flr;
vorbis_look_residue **residue;
vorbis_look_psy *psy;
vorbis_look_psy_global *psy_g_look;
/* local storage, only used on the encoding side. This way the
application does not need to worry about freeing some packets'
memory and not others'; packet storage is always tracked.
Cleared next call to a _dsp_ function */
unsigned char *header;
unsigned char *header1;
unsigned char *header2;
bitrate_manager_state bms;
ogg_int64_t sample_count;
} private_state;
/* codec_setup_info contains all the setup information specific to the
specific compression/decompression mode in progress (eg,
psychoacoustic settings, channel setup, options, codebook
etc).
*********************************************************************/
#include "highlevel.h"
typedef struct codec_setup_info {
/* Vorbis supports only short and long blocks, but allows the
encoder to choose the sizes */
long blocksizes[2];
/* modes are the primary means of supporting on-the-fly different
blocksizes, different channel mappings (LR or M/A),
different residue backends, etc. Each mode consists of a
blocksize flag and a mapping (along with the mapping setup */
int modes;
int maps;
int floors;
int residues;
int books;
int psys; /* encode only */
vorbis_info_mode *mode_param[64];
int map_type[64];
vorbis_info_mapping *map_param[64];
int floor_type[64];
vorbis_info_floor *floor_param[64];
int residue_type[64];
vorbis_info_residue *residue_param[64];
static_codebook *book_param[256];
codebook *fullbooks;
vorbis_info_psy *psy_param[4]; /* encode only */
vorbis_info_psy_global psy_g_param;
bitrate_manager_info bi;
highlevel_encode_setup hi; /* used only by vorbisenc.c. It's a
highly redundant structure, but
improves clarity of program flow. */
int halfrate_flag; /* painless downsample for decode */
} codec_setup_info;
extern vorbis_look_psy_global *_vp_global_look(vorbis_info *vi);
extern void _vp_global_free(vorbis_look_psy_global *look);
typedef struct {
int sorted_index[VIF_POSIT+2];
int forward_index[VIF_POSIT+2];
int reverse_index[VIF_POSIT+2];
int hineighbor[VIF_POSIT];
int loneighbor[VIF_POSIT];
int posts;
int n;
int quant_q;
vorbis_info_floor1 *vi;
long phrasebits;
long postbits;
long frames;
} vorbis_look_floor1;
extern int *floor1_fit(vorbis_block *vb,vorbis_look_floor1 *look,
const float *logmdct, /* in */
const float *logmask);
extern int *floor1_interpolate_fit(vorbis_block *vb,vorbis_look_floor1 *look,
int *A,int *B,
int del);
extern int floor1_encode(oggpack_buffer *opb,vorbis_block *vb,
vorbis_look_floor1 *look,
int *post,int *ilogmask);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/envelope.c
+375
View File
@@ -0,0 +1,375 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: PCM data envelope analysis
last mod: $Id: envelope.c 16227 2009-07-08 06:58:46Z xiphmont $
********************************************************************/
#include <stdlib.h>
#include <string.h>
#include <stdio.h>
#include <math.h>
#include "../../ogg.h"
#include "../../codec.h"
#include "codec_internal.h"
#include "os.h"
#include "scales.h"
#include "envelope.h"
#include "mdct.h"
#include "misc.h"
void _ve_envelope_init(envelope_lookup *e,vorbis_info *vi){
codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
vorbis_info_psy_global *gi=&ci->psy_g_param;
int ch=vi->channels;
int i,j;
int n=e->winlength=128;
e->searchstep=64; /* not random */
e->minenergy=gi->preecho_minenergy;
e->ch=ch;
e->storage=128;
e->cursor=ci->blocksizes[1]/2;
e->mdct_win=(float*)_ogg_calloc(n,sizeof(*e->mdct_win));
mdct_init(&e->mdct,n);
for(i=0;i<n;i++){
e->mdct_win[i]=sin(i/(n-1.)*M_PI);
e->mdct_win[i]*=e->mdct_win[i];
}
/* magic follows */
e->band[0].begin=2; e->band[0].end=4;
e->band[1].begin=4; e->band[1].end=5;
e->band[2].begin=6; e->band[2].end=6;
e->band[3].begin=9; e->band[3].end=8;
e->band[4].begin=13; e->band[4].end=8;
e->band[5].begin=17; e->band[5].end=8;
e->band[6].begin=22; e->band[6].end=8;
for(j=0;j<VE_BANDS;j++){
n=e->band[j].end;
e->band[j].window=(float*)_ogg_malloc(n*sizeof(*e->band[0].window));
for(i=0;i<n;i++){
e->band[j].window[i]=sin((i+.5)/n*M_PI);
e->band[j].total+=e->band[j].window[i];
}
e->band[j].total=1./e->band[j].total;
}
e->filter=(envelope_filter_state*)_ogg_calloc(VE_BANDS*ch,sizeof(*e->filter));
e->mark=(int*)_ogg_calloc(e->storage,sizeof(*e->mark));
}
void _ve_envelope_clear(envelope_lookup *e){
int i;
mdct_clear(&e->mdct);
for(i=0;i<VE_BANDS;i++)
_ogg_free(e->band[i].window);
_ogg_free(e->mdct_win);
_ogg_free(e->filter);
_ogg_free(e->mark);
memset(e,0,sizeof(*e));
}
/* fairly straight threshhold-by-band based until we find something
that works better and isn't patented. */
static int _ve_amp(envelope_lookup *ve,
vorbis_info_psy_global *gi,
float *data,
envelope_band *bands,
envelope_filter_state *filters){
long n=ve->winlength;
int ret=0;
long i,j;
float decay;
/* we want to have a 'minimum bar' for energy, else we're just
basing blocks on quantization noise that outweighs the signal
itself (for low power signals) */
float minV=ve->minenergy;
float *vec=(float*) alloca(n*sizeof(*vec));
/* stretch is used to gradually lengthen the number of windows
considered prevoius-to-potential-trigger */
int stretch=max(VE_MINSTRETCH,ve->stretch/2);
float penalty=gi->stretch_penalty-(ve->stretch/2-VE_MINSTRETCH);
if(penalty<0.f)penalty=0.f;
if(penalty>gi->stretch_penalty)penalty=gi->stretch_penalty;
/*_analysis_output_always("lpcm",seq2,data,n,0,0,
totalshift+pos*ve->searchstep);*/
/* window and transform */
for(i=0;i<n;i++)
vec[i]=data[i]*ve->mdct_win[i];
mdct_forward(&ve->mdct,vec,vec);
/*_analysis_output_always("mdct",seq2,vec,n/2,0,1,0); */
/* near-DC spreading function; this has nothing to do with
psychoacoustics, just sidelobe leakage and window size */
{
float temp=vec[0]*vec[0]+.7*vec[1]*vec[1]+.2*vec[2]*vec[2];
int ptr=filters->nearptr;
/* the accumulation is regularly refreshed from scratch to avoid
floating point creep */
if(ptr==0){
decay=filters->nearDC_acc=filters->nearDC_partialacc+temp;
filters->nearDC_partialacc=temp;
}else{
decay=filters->nearDC_acc+=temp;
filters->nearDC_partialacc+=temp;
}
filters->nearDC_acc-=filters->nearDC[ptr];
filters->nearDC[ptr]=temp;
decay*=(1./(VE_NEARDC+1));
filters->nearptr++;
if(filters->nearptr>=VE_NEARDC)filters->nearptr=0;
decay=todB(&decay)*.5-15.f;
}
/* perform spreading and limiting, also smooth the spectrum. yes,
the MDCT results in all real coefficients, but it still *behaves*
like real/imaginary pairs */
for(i=0;i<n/2;i+=2){
float val=vec[i]*vec[i]+vec[i+1]*vec[i+1];
val=todB(&val)*.5f;
if(val<decay)val=decay;
if(val<minV)val=minV;
vec[i>>1]=val;
decay-=8.;
}
/*_analysis_output_always("spread",seq2++,vec,n/4,0,0,0);*/
/* perform preecho/postecho triggering by band */
for(j=0;j<VE_BANDS;j++){
float acc=0.;
float valmax,valmin;
/* accumulate amplitude */
for(i=0;i<bands[j].end;i++)
acc+=vec[i+bands[j].begin]*bands[j].window[i];
acc*=bands[j].total;
/* convert amplitude to delta */
{
int p,thisx=filters[j].ampptr;
float postmax,postmin,premax=-99999.f,premin=99999.f;
p=thisx;
p--;
if(p<0)p+=VE_AMP;
postmax=max(acc,filters[j].ampbuf[p]);
postmin=min(acc,filters[j].ampbuf[p]);
for(i=0;i<stretch;i++){
p--;
if(p<0)p+=VE_AMP;
premax=max(premax,filters[j].ampbuf[p]);
premin=min(premin,filters[j].ampbuf[p]);
}
valmin=postmin-premin;
valmax=postmax-premax;
/*filters[j].markers[pos]=valmax;*/
filters[j].ampbuf[thisx]=acc;
filters[j].ampptr++;
if(filters[j].ampptr>=VE_AMP)filters[j].ampptr=0;
}
/* look at min/max, decide trigger */
if(valmax>gi->preecho_thresh[j]+penalty){
ret|=1;
ret|=4;
}
if(valmin<gi->postecho_thresh[j]-penalty)ret|=2;
}
return(ret);
}
#if 0
static int seq=0;
static ogg_int64_t totalshift=-1024;
#endif
long _ve_envelope_search(vorbis_dsp_state *v){
vorbis_info *vi=v->vi;
codec_setup_info *ci=(codec_setup_info *)vi->codec_setup;
vorbis_info_psy_global *gi=&ci->psy_g_param;
envelope_lookup *ve=((private_state *)(v->backend_state))->ve;
long i,j;
int first=ve->current/ve->searchstep;
int last=v->pcm_current/ve->searchstep-VE_WIN;
if(first<0)first=0;
/* make sure we have enough storage to match the PCM */
if(last+VE_WIN+VE_POST>ve->storage){
ve->storage=last+VE_WIN+VE_POST; /* be sure */
ve->mark=(int*)_ogg_realloc(ve->mark,ve->storage*sizeof(*ve->mark));
}
for(j=first;j<last;j++){
int ret=0;
ve->stretch++;
if(ve->stretch>VE_MAXSTRETCH*2)
ve->stretch=VE_MAXSTRETCH*2;
for(i=0;i<ve->ch;i++){
float *pcm=v->pcm[i]+ve->searchstep*(j);
ret|=_ve_amp(ve,gi,pcm,ve->band,ve->filter+i*VE_BANDS);
}
ve->mark[j+VE_POST]=0;
if(ret&1){
ve->mark[j]=1;
ve->mark[j+1]=1;
}
if(ret&2){
ve->mark[j]=1;
if(j>0)ve->mark[j-1]=1;
}
if(ret&4)ve->stretch=-1;
}
ve->current=last*ve->searchstep;
{
long centerW=v->centerW;
long testW=
centerW+
ci->blocksizes[v->W]/4+
ci->blocksizes[1]/2+
ci->blocksizes[0]/4;
j=ve->cursor;
while(j<ve->current-(ve->searchstep)){/* account for postecho
working back one window */
if(j>=testW)return(1);
ve->cursor=j;
if(ve->mark[j/ve->searchstep]){
if(j>centerW){
#if 0
if(j>ve->curmark){
float *marker=(float*)alloca(v->pcm_current*sizeof(*marker));
int l,m;
memset(marker,0,sizeof(*marker)*v->pcm_current);
fprintf(stderr,"mark! seq=%d, cursor:%fs time:%fs\n",
seq,
(totalshift+ve->cursor)/44100.,
(totalshift+j)/44100.);
_analysis_output_always("pcmL",seq,v->pcm[0],v->pcm_current,0,0,totalshift);
_analysis_output_always("pcmR",seq,v->pcm[1],v->pcm_current,0,0,totalshift);
_analysis_output_always("markL",seq,v->pcm[0],j,0,0,totalshift);
_analysis_output_always("markR",seq,v->pcm[1],j,0,0,totalshift);
for(m=0;m<VE_BANDS;m++){
char buf[80];
sprintf(buf,"delL%d",m);
for(l=0;l<last;l++)marker[l*ve->searchstep]=ve->filter[m].markers[l]*.1;
_analysis_output_always(buf,seq,marker,v->pcm_current,0,0,totalshift);
}
for(m=0;m<VE_BANDS;m++){
char buf[80];
sprintf(buf,"delR%d",m);
for(l=0;l<last;l++)marker[l*ve->searchstep]=ve->filter[m+VE_BANDS].markers[l]*.1;
_analysis_output_always(buf,seq,marker,v->pcm_current,0,0,totalshift);
}
for(l=0;l<last;l++)marker[l*ve->searchstep]=ve->mark[l]*.4;
_analysis_output_always("mark",seq,marker,v->pcm_current,0,0,totalshift);
seq++;
}
#endif
ve->curmark=j;
if(j>=testW)return(1);
return(0);
}
}
j+=ve->searchstep;
}
}
return(-1);
}
int _ve_envelope_mark(vorbis_dsp_state *v){
envelope_lookup *ve=((private_state *)(v->backend_state))->ve;
vorbis_info *vi=v->vi;
codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
long centerW=v->centerW;
long beginW=centerW-ci->blocksizes[v->W]/4;
long endW=centerW+ci->blocksizes[v->W]/4;
if(v->W){
beginW-=ci->blocksizes[v->lW]/4;
endW+=ci->blocksizes[v->nW]/4;
}else{
beginW-=ci->blocksizes[0]/4;
endW+=ci->blocksizes[0]/4;
}
if(ve->curmark>=beginW && ve->curmark<endW)return(1);
{
long first=beginW/ve->searchstep;
long last=endW/ve->searchstep;
long i;
for(i=first;i<last;i++)
if(ve->mark[i])return(1);
}
return(0);
}
void _ve_envelope_shift(envelope_lookup *e,long shift){
int smallsize=e->current/e->searchstep+VE_POST; /* adjust for placing marks
ahead of ve->current */
int smallshift=shift/e->searchstep;
memmove(e->mark,e->mark+smallshift,(smallsize-smallshift)*sizeof(*e->mark));
#if 0
for(i=0;i<VE_BANDS*e->ch;i++)
memmove(e->filter[i].markers,
e->filter[i].markers+smallshift,
(1024-smallshift)*sizeof(*(*e->filter).markers));
totalshift+=shift;
#endif
e->current-=shift;
if(e->curmark>=0)
e->curmark-=shift;
e->cursor-=shift;
}
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/envelope.h
+80
View File
@@ -0,0 +1,80 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: PCM data envelope analysis and manipulation
last mod: $Id: envelope.h 16227 2009-07-08 06:58:46Z xiphmont $
********************************************************************/
#ifndef _V_ENVELOPE_
#define _V_ENVELOPE_
#include "mdct.h"
#define VE_PRE 16
#define VE_WIN 4
#define VE_POST 2
#define VE_AMP (VE_PRE+VE_POST-1)
#define VE_BANDS 7
#define VE_NEARDC 15
#define VE_MINSTRETCH 2 /* a bit less than short block */
#define VE_MAXSTRETCH 12 /* one-third full block */
typedef struct {
float ampbuf[VE_AMP];
int ampptr;
float nearDC[VE_NEARDC];
float nearDC_acc;
float nearDC_partialacc;
int nearptr;
} envelope_filter_state;
typedef struct {
int begin;
int end;
float *window;
float total;
} envelope_band;
typedef struct {
int ch;
int winlength;
int searchstep;
float minenergy;
mdct_lookup mdct;
float *mdct_win;
envelope_band band[VE_BANDS];
envelope_filter_state *filter;
int stretch;
int *mark;
long storage;
long current;
long curmark;
long cursor;
} envelope_lookup;
extern void _ve_envelope_init(envelope_lookup *e,vorbis_info *vi);
extern void _ve_envelope_clear(envelope_lookup *e);
extern long _ve_envelope_search(vorbis_dsp_state *v);
extern void _ve_envelope_shift(envelope_lookup *e,long shift);
extern int _ve_envelope_mark(vorbis_dsp_state *v);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/floor0.c
+223
View File
@@ -0,0 +1,223 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: floor backend 0 implementation
last mod: $Id: floor0.c 17558 2010-10-22 00:24:41Z tterribe $
********************************************************************/
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include "../../ogg.h"
#include "../../codec.h"
#include "codec_internal.h"
#include "registry.h"
#include "lpc.h"
#include "lsp.h"
#include "codebook.h"
#include "scales.h"
#include "misc.h"
#include "os.h"
#include "misc.h"
#include <stdio.h>
typedef struct {
int ln;
int m;
int **linearmap;
int n[2];
vorbis_info_floor0 *vi;
long bits;
long frames;
} vorbis_look_floor0;
/***********************************************/
static void floor0_free_info(vorbis_info_floor *i){
vorbis_info_floor0 *info=(vorbis_info_floor0 *)i;
if(info){
memset(info,0,sizeof(*info));
_ogg_free(info);
}
}
static void floor0_free_look(vorbis_look_floor *i){
vorbis_look_floor0 *look=(vorbis_look_floor0 *)i;
if(look){
if(look->linearmap){
if(look->linearmap[0])_ogg_free(look->linearmap[0]);
if(look->linearmap[1])_ogg_free(look->linearmap[1]);
_ogg_free(look->linearmap);
}
memset(look,0,sizeof(*look));
_ogg_free(look);
}
}
static vorbis_info_floor *floor0_unpack (vorbis_info *vi,oggpack_buffer *opb){
codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
int j;
vorbis_info_floor0 *info=(vorbis_info_floor0*)_ogg_malloc(sizeof(*info));
info->order=oggpack_read(opb,8);
info->rate=oggpack_read(opb,16);
info->barkmap=oggpack_read(opb,16);
info->ampbits=oggpack_read(opb,6);
info->ampdB=oggpack_read(opb,8);
info->numbooks=oggpack_read(opb,4)+1;
if(info->order<1)goto err_out;
if(info->rate<1)goto err_out;
if(info->barkmap<1)goto err_out;
if(info->numbooks<1)goto err_out;
for(j=0;j<info->numbooks;j++){
info->books[j]=oggpack_read(opb,8);
if(info->books[j]<0 || info->books[j]>=ci->books)goto err_out;
if(ci->book_param[info->books[j]]->maptype==0)goto err_out;
if(ci->book_param[info->books[j]]->dim<1)goto err_out;
}
return(info);
err_out:
floor0_free_info(info);
return(NULL);
}
/* initialize Bark scale and normalization lookups. We could do this
with static tables, but Vorbis allows a number of possible
combinations, so it's best to do it computationally.
The below is authoritative in terms of defining scale mapping.
Note that the scale depends on the sampling rate as well as the
linear block and mapping sizes */
static void floor0_map_lazy_init(vorbis_block *vb,
vorbis_info_floor *infoX,
vorbis_look_floor0 *look){
if(!look->linearmap[vb->W]){
vorbis_dsp_state *vd=vb->vd;
vorbis_info *vi=vd->vi;
codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
vorbis_info_floor0 *info=(vorbis_info_floor0 *)infoX;
int W=vb->W;
int n=ci->blocksizes[W]/2,j;
/* we choose a scaling constant so that:
floor(bark(rate/2-1)*C)=mapped-1
floor(bark(rate/2)*C)=mapped */
float scale=look->ln/toBARK(info->rate/2.f);
/* the mapping from a linear scale to a smaller bark scale is
straightforward. We do *not* make sure that the linear mapping
does not skip bark-scale bins; the decoder simply skips them and
the encoder may do what it wishes in filling them. They're
necessary in some mapping combinations to keep the scale spacing
accurate */
look->linearmap[W]=(int*)_ogg_malloc((n+1)*sizeof(**look->linearmap));
for(j=0;j<n;j++){
int val=floor( toBARK((info->rate/2.f)/n*j)
*scale); /* bark numbers represent band edges */
if(val>=look->ln)val=look->ln-1; /* guard against the approximation */
look->linearmap[W][j]=val;
}
look->linearmap[W][j]=-1;
look->n[W]=n;
}
}
static vorbis_look_floor *floor0_look(vorbis_dsp_state* /* vd */,
vorbis_info_floor *i){
vorbis_info_floor0 *info=(vorbis_info_floor0 *)i;
vorbis_look_floor0 *look=(vorbis_look_floor0*)_ogg_calloc(1,sizeof(*look));
look->m=info->order;
look->ln=info->barkmap;
look->vi=info;
look->linearmap=(int**)_ogg_calloc(2,sizeof(*look->linearmap));
return look;
}
static void *floor0_inverse1(vorbis_block *vb,vorbis_look_floor *i){
vorbis_look_floor0 *look=(vorbis_look_floor0 *)i;
vorbis_info_floor0 *info=look->vi;
int j,k;
int ampraw=oggpack_read(&vb->opb,info->ampbits);
if(ampraw>0){ /* also handles the -1 out of data case */
long maxval=(1<<info->ampbits)-1;
float amp=(float)ampraw/maxval*info->ampdB;
int booknum=oggpack_read(&vb->opb,_ilog(info->numbooks));
if(booknum!=-1 && booknum<info->numbooks){ /* be paranoid */
codec_setup_info *ci=(codec_setup_info *)vb->vd->vi->codec_setup;
codebook *b=ci->fullbooks+info->books[booknum];
float last=0.f;
/* the additional b->dim is a guard against any possible stack
smash; b->dim is provably more than we can overflow the
vector */
float *lsp=(float*)_vorbis_block_alloc(vb,sizeof(*lsp)*(look->m+b->dim+1));
for(j=0;j<look->m;j+=b->dim)
if(vorbis_book_decodev_set(b,lsp+j,&vb->opb,b->dim)==-1)goto eop;
for(j=0;j<look->m;){
for(k=0;k<b->dim;k++,j++)lsp[j]+=last;
last=lsp[j-1];
}
lsp[look->m]=amp;
return(lsp);
}
}
eop:
return(NULL);
}
static int floor0_inverse2(vorbis_block *vb,vorbis_look_floor *i,
void *memo,float *out){
vorbis_look_floor0 *look=(vorbis_look_floor0 *)i;
vorbis_info_floor0 *info=look->vi;
floor0_map_lazy_init(vb,info,look);
if(memo){
float *lsp=(float *)memo;
float amp=lsp[look->m];
/* take the coefficients back to a spectral envelope curve */
vorbis_lsp_to_curve(out,
look->linearmap[vb->W],
look->n[vb->W],
look->ln,
lsp,look->m,amp,(float)info->ampdB);
return(1);
}
memset(out,0,sizeof(*out)*look->n[vb->W]);
return(0);
}
/* export hooks */
const vorbis_func_floor floor0_exportbundle={
NULL,&floor0_unpack,&floor0_look,&floor0_free_info,
&floor0_free_look,&floor0_inverse1,&floor0_inverse2
};
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/floor1.c
+1080
View File
File diff suppressed because it is too large Load Diff
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/highlevel.h
+58
View File
@@ -0,0 +1,58 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: highlevel encoder setup struct separated out for vorbisenc clarity
last mod: $Id: highlevel.h 17195 2010-05-05 21:49:51Z giles $
********************************************************************/
typedef struct highlevel_byblocktype {
double tone_mask_setting;
double tone_peaklimit_setting;
double noise_bias_setting;
double noise_compand_setting;
} highlevel_byblocktype;
typedef struct highlevel_encode_setup {
int set_in_stone;
const void *setup;
double base_setting;
double impulse_noisetune;
/* bitrate management below all settable */
float req;
int managed;
long bitrate_min;
long bitrate_av;
double bitrate_av_damp;
long bitrate_max;
long bitrate_reservoir;
double bitrate_reservoir_bias;
int impulse_block_p;
int noise_normalize_p;
int coupling_p;
double stereo_point_setting;
double lowpass_kHz;
int lowpass_altered;
double ath_floating_dB;
double ath_absolute_dB;
double amplitude_track_dBpersec;
double trigger_setting;
highlevel_byblocktype block[4]; /* padding, impulse, transition, long */
} highlevel_encode_setup;
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/info.c
+660
View File
@@ -0,0 +1,660 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2010 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: maintain the info structure, info <-> header packets
last mod: $Id: info.c 17584 2010-11-01 19:26:16Z xiphmont $
********************************************************************/
/* general handling of the header and the vorbis_info structure (and
substructures) */
#include <stdlib.h>
#include <string.h>
#include <ctype.h>
#include "../../ogg.h"
#include "../../codec.h"
#include "codec_internal.h"
#include "codebook.h"
#include "registry.h"
#include "window.h"
#include "psy.h"
#include "misc.h"
#include "os.h"
#define GENERAL_VENDOR_STRING "Xiph.Org libVorbis 1.3.2"
#define ENCODE_VENDOR_STRING "Xiph.Org libVorbis I 20101101 (Schaufenugget)"
/* helpers */
static void _v_writestring(oggpack_buffer *o,const char *s, int bytes){
while(bytes--){
oggpack_write(o,*s++,8);
}
}
static void _v_readstring(oggpack_buffer *o,char *buf,int bytes){
while(bytes--){
*buf++=oggpack_read(o,8);
}
}
void vorbis_comment_init(vorbis_comment *vc){
memset(vc,0,sizeof(*vc));
}
void vorbis_comment_add(vorbis_comment *vc,const char *comment){
vc->user_comments=(char**)_ogg_realloc(vc->user_comments,
(vc->comments+2)*sizeof(*vc->user_comments));
vc->comment_lengths=(int*)_ogg_realloc(vc->comment_lengths,
(vc->comments+2)*sizeof(*vc->comment_lengths));
vc->comment_lengths[vc->comments]=strlen(comment);
vc->user_comments[vc->comments]=(char*)_ogg_malloc(vc->comment_lengths[vc->comments]+1);
strcpy(vc->user_comments[vc->comments], comment);
vc->comments++;
vc->user_comments[vc->comments]=NULL;
}
void vorbis_comment_add_tag(vorbis_comment *vc, const char *tag, const char *contents){
char *comment=(char*)alloca(strlen(tag)+strlen(contents)+2); /* +2 for = and \0 */
strcpy(comment, tag);
strcat(comment, "=");
strcat(comment, contents);
vorbis_comment_add(vc, comment);
}
/* This is more or less the same as strncasecmp - but that doesn't exist
* everywhere, and this is a fairly trivial function, so we include it */
static int tagcompare(const char *s1, const char *s2, int n){
int c=0;
while(c < n){
if(toupper(s1[c]) != toupper(s2[c]))
return !0;
c++;
}
return 0;
}
char *vorbis_comment_query(vorbis_comment *vc, const char *tag, int count){
long i;
int found = 0;
int taglen = strlen(tag)+1; /* +1 for the = we append */
char *fulltag = (char*)alloca(taglen+ 1);
strcpy(fulltag, tag);
strcat(fulltag, "=");
for(i=0;i<vc->comments;i++){
if(!tagcompare(vc->user_comments[i], fulltag, taglen)){
if(count == found)
/* We return a pointer to the data, not a copy */
return vc->user_comments[i] + taglen;
else
found++;
}
}
return NULL; /* didn't find anything */
}
int vorbis_comment_query_count(vorbis_comment *vc, const char *tag){
int i,count=0;
int taglen = strlen(tag)+1; /* +1 for the = we append */
char *fulltag = (char*)alloca(taglen+1);
strcpy(fulltag,tag);
strcat(fulltag, "=");
for(i=0;i<vc->comments;i++){
if(!tagcompare(vc->user_comments[i], fulltag, taglen))
count++;
}
return count;
}
void vorbis_comment_clear(vorbis_comment *vc){
if(vc){
long i;
if(vc->user_comments){
for(i=0;i<vc->comments;i++)
if(vc->user_comments[i])_ogg_free(vc->user_comments[i]);
_ogg_free(vc->user_comments);
}
if(vc->comment_lengths)_ogg_free(vc->comment_lengths);
if(vc->vendor)_ogg_free(vc->vendor);
memset(vc,0,sizeof(*vc));
}
}
/* blocksize 0 is guaranteed to be short, 1 is guaranteed to be long.
They may be equal, but short will never ge greater than long */
int vorbis_info_blocksize(vorbis_info *vi,int zo){
codec_setup_info *ci = (codec_setup_info*)vi->codec_setup;
return ci ? ci->blocksizes[zo] : -1;
}
/* used by synthesis, which has a full, alloced vi */
void vorbis_info_init(vorbis_info *vi){
memset(vi,0,sizeof(*vi));
vi->codec_setup=_ogg_calloc(1,sizeof(codec_setup_info));
}
void vorbis_info_clear(vorbis_info *vi){
codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
int i;
if(ci){
for(i=0;i<ci->modes;i++)
if(ci->mode_param[i])_ogg_free(ci->mode_param[i]);
for(i=0;i<ci->maps;i++) /* unpack does the range checking */
if(ci->map_param[i]) /* this may be cleaning up an aborted
unpack, in which case the below type
cannot be trusted */
_mapping_P[ci->map_type[i]]->free_info(ci->map_param[i]);
for(i=0;i<ci->floors;i++) /* unpack does the range checking */
if(ci->floor_param[i]) /* this may be cleaning up an aborted
unpack, in which case the below type
cannot be trusted */
_floor_P[ci->floor_type[i]]->free_info(ci->floor_param[i]);
for(i=0;i<ci->residues;i++) /* unpack does the range checking */
if(ci->residue_param[i]) /* this may be cleaning up an aborted
unpack, in which case the below type
cannot be trusted */
_residue_P[ci->residue_type[i]]->free_info(ci->residue_param[i]);
for(i=0;i<ci->books;i++){
if(ci->book_param[i]){
/* knows if the book was not alloced */
vorbis_staticbook_destroy(ci->book_param[i]);
}
if(ci->fullbooks)
vorbis_book_clear(ci->fullbooks+i);
}
if(ci->fullbooks)
_ogg_free(ci->fullbooks);
for(i=0;i<ci->psys;i++)
_vi_psy_free(ci->psy_param[i]);
_ogg_free(ci);
}
memset(vi,0,sizeof(*vi));
}
/* Header packing/unpacking ********************************************/
static int _vorbis_unpack_info(vorbis_info *vi,oggpack_buffer *opb){
codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
if(!ci)return(OV_EFAULT);
vi->version=oggpack_read(opb,32);
if(vi->version!=0)return(OV_EVERSION);
vi->channels=oggpack_read(opb,8);
vi->rate=oggpack_read(opb,32);
vi->bitrate_upper=oggpack_read(opb,32);
vi->bitrate_nominal=oggpack_read(opb,32);
vi->bitrate_lower=oggpack_read(opb,32);
ci->blocksizes[0]=1<<oggpack_read(opb,4);
ci->blocksizes[1]=1<<oggpack_read(opb,4);
if(vi->rate<1)goto err_out;
if(vi->channels<1)goto err_out;
if(ci->blocksizes[0]<64)goto err_out;
if(ci->blocksizes[1]<ci->blocksizes[0])goto err_out;
if(ci->blocksizes[1]>8192)goto err_out;
if(oggpack_read(opb,1)!=1)goto err_out; /* EOP check */
return(0);
err_out:
vorbis_info_clear(vi);
return(OV_EBADHEADER);
}
static int _vorbis_unpack_comment(vorbis_comment *vc,oggpack_buffer *opb){
int i;
int vendorlen=oggpack_read(opb,32);
if(vendorlen<0)goto err_out;
if(vendorlen>opb->storage-8)goto err_out;
vc->vendor=(char*)_ogg_calloc(vendorlen+1,1);
_v_readstring(opb,vc->vendor,vendorlen);
i=oggpack_read(opb,32);
if(i<0)goto err_out;
if(i>((opb->storage-oggpack_bytes(opb))>>2))goto err_out;
vc->comments=i;
vc->user_comments=(char**)_ogg_calloc(vc->comments+1,sizeof(*vc->user_comments));
vc->comment_lengths=(int*)_ogg_calloc(vc->comments+1, sizeof(*vc->comment_lengths));
for(i=0;i<vc->comments;i++){
int len=oggpack_read(opb,32);
if(len<0)goto err_out;
if(len>opb->storage-oggpack_bytes(opb))goto err_out;
vc->comment_lengths[i]=len;
vc->user_comments[i]=(char*)_ogg_calloc(len+1,1);
_v_readstring(opb,vc->user_comments[i],len);
}
if(oggpack_read(opb,1)!=1)goto err_out; /* EOP check */
return(0);
err_out:
vorbis_comment_clear(vc);
return(OV_EBADHEADER);
}
/* all of the real encoding details are here. The modes, books,
everything */
static int _vorbis_unpack_books(vorbis_info *vi,oggpack_buffer *opb){
codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
int i;
if(!ci)return(OV_EFAULT);
/* codebooks */
ci->books=oggpack_read(opb,8)+1;
if(ci->books<=0)goto err_out;
for(i=0;i<ci->books;i++){
ci->book_param[i]=vorbis_staticbook_unpack(opb);
if(!ci->book_param[i])goto err_out;
}
/* time backend settings; hooks are unused */
{
int times=oggpack_read(opb,6)+1;
if(times<=0)goto err_out;
for(i=0;i<times;i++){
int test=oggpack_read(opb,16);
if(test<0 || test>=VI_TIMEB)goto err_out;
}
}
/* floor backend settings */
ci->floors=oggpack_read(opb,6)+1;
if(ci->floors<=0)goto err_out;
for(i=0;i<ci->floors;i++){
ci->floor_type[i]=oggpack_read(opb,16);
if(ci->floor_type[i]<0 || ci->floor_type[i]>=VI_FLOORB)goto err_out;
ci->floor_param[i]=_floor_P[ci->floor_type[i]]->unpack(vi,opb);
if(!ci->floor_param[i])goto err_out;
}
/* residue backend settings */
ci->residues=oggpack_read(opb,6)+1;
if(ci->residues<=0)goto err_out;
for(i=0;i<ci->residues;i++){
ci->residue_type[i]=oggpack_read(opb,16);
if(ci->residue_type[i]<0 || ci->residue_type[i]>=VI_RESB)goto err_out;
ci->residue_param[i]=_residue_P[ci->residue_type[i]]->unpack(vi,opb);
if(!ci->residue_param[i])goto err_out;
}
/* map backend settings */
ci->maps=oggpack_read(opb,6)+1;
if(ci->maps<=0)goto err_out;
for(i=0;i<ci->maps;i++){
ci->map_type[i]=oggpack_read(opb,16);
if(ci->map_type[i]<0 || ci->map_type[i]>=VI_MAPB)goto err_out;
ci->map_param[i]=_mapping_P[ci->map_type[i]]->unpack(vi,opb);
if(!ci->map_param[i])goto err_out;
}
/* mode settings */
ci->modes=oggpack_read(opb,6)+1;
if(ci->modes<=0)goto err_out;
for(i=0;i<ci->modes;i++){
ci->mode_param[i]=(vorbis_info_mode*)_ogg_calloc(1,sizeof(*ci->mode_param[i]));
ci->mode_param[i]->blockflag=oggpack_read(opb,1);
ci->mode_param[i]->windowtype=oggpack_read(opb,16);
ci->mode_param[i]->transformtype=oggpack_read(opb,16);
ci->mode_param[i]->mapping=oggpack_read(opb,8);
if(ci->mode_param[i]->windowtype>=VI_WINDOWB)goto err_out;
if(ci->mode_param[i]->transformtype>=VI_WINDOWB)goto err_out;
if(ci->mode_param[i]->mapping>=ci->maps)goto err_out;
if(ci->mode_param[i]->mapping<0)goto err_out;
}
if(oggpack_read(opb,1)!=1)goto err_out; /* top level EOP check */
return(0);
err_out:
vorbis_info_clear(vi);
return(OV_EBADHEADER);
}
/* Is this packet a vorbis ID header? */
int vorbis_synthesis_idheader(ogg_packet *op){
oggpack_buffer opb;
char buffer[6];
if(op){
oggpack_readinit(&opb,op->packet,op->bytes);
if(!op->b_o_s)
return(0); /* Not the initial packet */
if(oggpack_read(&opb,8) != 1)
return 0; /* not an ID header */
memset(buffer,0,6);
_v_readstring(&opb,buffer,6);
if(memcmp(buffer,"vorbis",6))
return 0; /* not vorbis */
return 1;
}
return 0;
}
/* The Vorbis header is in three packets; the initial small packet in
the first page that identifies basic parameters, a second packet
with bitstream comments and a third packet that holds the
codebook. */
int vorbis_synthesis_headerin(vorbis_info *vi,vorbis_comment *vc,ogg_packet *op){
oggpack_buffer opb;
if(op){
oggpack_readinit(&opb,op->packet,op->bytes);
/* Which of the three types of header is this? */
/* Also verify header-ness, vorbis */
{
char buffer[6];
int packtype=oggpack_read(&opb,8);
memset(buffer,0,6);
_v_readstring(&opb,buffer,6);
if(memcmp(buffer,"vorbis",6)){
/* not a vorbis header */
return(OV_ENOTVORBIS);
}
switch(packtype){
case 0x01: /* least significant *bit* is read first */
if(!op->b_o_s){
/* Not the initial packet */
return(OV_EBADHEADER);
}
if(vi->rate!=0){
/* previously initialized info header */
return(OV_EBADHEADER);
}
return(_vorbis_unpack_info(vi,&opb));
case 0x03: /* least significant *bit* is read first */
if(vi->rate==0){
/* um... we didn't get the initial header */
return(OV_EBADHEADER);
}
return(_vorbis_unpack_comment(vc,&opb));
case 0x05: /* least significant *bit* is read first */
if(vi->rate==0 || vc->vendor==NULL){
/* um... we didn;t get the initial header or comments yet */
return(OV_EBADHEADER);
}
return(_vorbis_unpack_books(vi,&opb));
default:
/* Not a valid vorbis header type */
return(OV_EBADHEADER);
break;
}
}
}
return(OV_EBADHEADER);
}
/* pack side **********************************************************/
static int _vorbis_pack_info(oggpack_buffer *opb,vorbis_info *vi){
codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
if(!ci)return(OV_EFAULT);
/* preamble */
oggpack_write(opb,0x01,8);
_v_writestring(opb,"vorbis", 6);
/* basic information about the stream */
oggpack_write(opb,0x00,32);
oggpack_write(opb,vi->channels,8);
oggpack_write(opb,vi->rate,32);
oggpack_write(opb,vi->bitrate_upper,32);
oggpack_write(opb,vi->bitrate_nominal,32);
oggpack_write(opb,vi->bitrate_lower,32);
oggpack_write(opb,ilog2(ci->blocksizes[0]),4);
oggpack_write(opb,ilog2(ci->blocksizes[1]),4);
oggpack_write(opb,1,1);
return(0);
}
static int _vorbis_pack_comment(oggpack_buffer *opb,vorbis_comment *vc){
int bytes = strlen(ENCODE_VENDOR_STRING);
/* preamble */
oggpack_write(opb,0x03,8);
_v_writestring(opb,"vorbis", 6);
/* vendor */
oggpack_write(opb,bytes,32);
_v_writestring(opb,ENCODE_VENDOR_STRING, bytes);
/* comments */
oggpack_write(opb,vc->comments,32);
if(vc->comments){
int i;
for(i=0;i<vc->comments;i++){
if(vc->user_comments[i]){
oggpack_write(opb,vc->comment_lengths[i],32);
_v_writestring(opb,vc->user_comments[i], vc->comment_lengths[i]);
}else{
oggpack_write(opb,0,32);
}
}
}
oggpack_write(opb,1,1);
return(0);
}
static int _vorbis_pack_books(oggpack_buffer *opb,vorbis_info *vi){
codec_setup_info *ci=(codec_setup_info*)vi->codec_setup;
int i;
if(!ci)return(OV_EFAULT);
oggpack_write(opb,0x05,8);
_v_writestring(opb,"vorbis", 6);
/* books */
oggpack_write(opb,ci->books-1,8);
for(i=0;i<ci->books;i++)
if(vorbis_staticbook_pack(ci->book_param[i],opb))goto err_out;
/* times; hook placeholders */
oggpack_write(opb,0,6);
oggpack_write(opb,0,16);
/* floors */
oggpack_write(opb,ci->floors-1,6);
for(i=0;i<ci->floors;i++){
oggpack_write(opb,ci->floor_type[i],16);
if(_floor_P[ci->floor_type[i]]->pack)
_floor_P[ci->floor_type[i]]->pack(ci->floor_param[i],opb);
else
goto err_out;
}
/* residues */
oggpack_write(opb,ci->residues-1,6);
for(i=0;i<ci->residues;i++){
oggpack_write(opb,ci->residue_type[i],16);
_residue_P[ci->residue_type[i]]->pack(ci->residue_param[i],opb);
}
/* maps */
oggpack_write(opb,ci->maps-1,6);
for(i=0;i<ci->maps;i++){
oggpack_write(opb,ci->map_type[i],16);
_mapping_P[ci->map_type[i]]->pack(vi,ci->map_param[i],opb);
}
/* modes */
oggpack_write(opb,ci->modes-1,6);
for(i=0;i<ci->modes;i++){
oggpack_write(opb,ci->mode_param[i]->blockflag,1);
oggpack_write(opb,ci->mode_param[i]->windowtype,16);
oggpack_write(opb,ci->mode_param[i]->transformtype,16);
oggpack_write(opb,ci->mode_param[i]->mapping,8);
}
oggpack_write(opb,1,1);
return(0);
err_out:
return(-1);
}
int vorbis_commentheader_out(vorbis_comment *vc,
ogg_packet *op){
oggpack_buffer opb;
oggpack_writeinit(&opb);
if(_vorbis_pack_comment(&opb,vc)) return OV_EIMPL;
op->packet = (unsigned char*) _ogg_malloc(oggpack_bytes(&opb));
memcpy(op->packet, opb.buffer, oggpack_bytes(&opb));
op->bytes=oggpack_bytes(&opb);
op->b_o_s=0;
op->e_o_s=0;
op->granulepos=0;
op->packetno=1;
return 0;
}
int vorbis_analysis_headerout(vorbis_dsp_state *v,
vorbis_comment *vc,
ogg_packet *op,
ogg_packet *op_comm,
ogg_packet *op_code){
int ret=OV_EIMPL;
vorbis_info *vi=v->vi;
oggpack_buffer opb;
private_state *b=(private_state*)v->backend_state;
if(!b){
ret=OV_EFAULT;
goto err_out;
}
/* first header packet **********************************************/
oggpack_writeinit(&opb);
if(_vorbis_pack_info(&opb,vi))goto err_out;
/* build the packet */
if(b->header)_ogg_free(b->header);
b->header=(unsigned char*) _ogg_malloc(oggpack_bytes(&opb));
memcpy(b->header,opb.buffer,oggpack_bytes(&opb));
op->packet=b->header;
op->bytes=oggpack_bytes(&opb);
op->b_o_s=1;
op->e_o_s=0;
op->granulepos=0;
op->packetno=0;
/* second header packet (comments) **********************************/
oggpack_reset(&opb);
if(_vorbis_pack_comment(&opb,vc))goto err_out;
if(b->header1)_ogg_free(b->header1);
b->header1=(unsigned char*) _ogg_malloc(oggpack_bytes(&opb));
memcpy(b->header1,opb.buffer,oggpack_bytes(&opb));
op_comm->packet=b->header1;
op_comm->bytes=oggpack_bytes(&opb);
op_comm->b_o_s=0;
op_comm->e_o_s=0;
op_comm->granulepos=0;
op_comm->packetno=1;
/* third header packet (modes/codebooks) ****************************/
oggpack_reset(&opb);
if(_vorbis_pack_books(&opb,vi))goto err_out;
if(b->header2)_ogg_free(b->header2);
b->header2=(unsigned char*) _ogg_malloc(oggpack_bytes(&opb));
memcpy(b->header2,opb.buffer,oggpack_bytes(&opb));
op_code->packet=b->header2;
op_code->bytes=oggpack_bytes(&opb);
op_code->b_o_s=0;
op_code->e_o_s=0;
op_code->granulepos=0;
op_code->packetno=2;
oggpack_writeclear(&opb);
return(0);
err_out:
memset(op,0,sizeof(*op));
memset(op_comm,0,sizeof(*op_comm));
memset(op_code,0,sizeof(*op_code));
if(b){
oggpack_writeclear(&opb);
if(b->header)_ogg_free(b->header);
if(b->header1)_ogg_free(b->header1);
if(b->header2)_ogg_free(b->header2);
b->header=NULL;
b->header1=NULL;
b->header2=NULL;
}
return(ret);
}
double vorbis_granule_time(vorbis_dsp_state *v,ogg_int64_t granulepos){
if(granulepos == -1) return -1;
/* We're not guaranteed a 64 bit unsigned type everywhere, so we
have to put the unsigned granpo in a signed type. */
if(granulepos>=0){
return((double)granulepos/v->vi->rate);
}else{
ogg_int64_t granuleoff=0xffffffff;
granuleoff<<=31;
#ifdef __GNUC__
granuleoff |= 0x7ffffffffLL;
#else
granuleoff |= 0x7ffffffff;
#endif
return(((double)granulepos+2+granuleoff+granuleoff)/v->vi->rate);
}
}
const char *vorbis_version_string(void){
return GENERAL_VENDOR_STRING;
}
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/lookup.c
+94
View File
@@ -0,0 +1,94 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: lookup based functions
last mod: $Id: lookup.c 16227 2009-07-08 06:58:46Z xiphmont $
********************************************************************/
#include <math.h>
#include "lookup.h"
#include "lookup_data.h"
#include "os.h"
#include "misc.h"
#ifdef FLOAT_LOOKUP
/* interpolated lookup based cos function, domain 0 to PI only */
float vorbis_coslook(float a){
double d=a*(.31830989*(float)COS_LOOKUP_SZ);
int i=vorbis_ftoi(d-.5);
return COS_LOOKUP[i]+ (d-i)*(COS_LOOKUP[i+1]-COS_LOOKUP[i]);
}
/* interpolated 1./sqrt(p) where .5 <= p < 1. */
float vorbis_invsqlook(float a){
double d=a*(2.f*(float)INVSQ_LOOKUP_SZ)-(float)INVSQ_LOOKUP_SZ;
int i=vorbis_ftoi(d-.5f);
return INVSQ_LOOKUP[i]+ (d-i)*(INVSQ_LOOKUP[i+1]-INVSQ_LOOKUP[i]);
}
/* interpolated 1./sqrt(p) where .5 <= p < 1. */
float vorbis_invsq2explook(int a){
return INVSQ2EXP_LOOKUP[a-INVSQ2EXP_LOOKUP_MIN];
}
#include <stdio.h>
/* interpolated lookup based fromdB function, domain -140dB to 0dB only */
float vorbis_fromdBlook(float a){
int i=vorbis_ftoi(a*((float)(-(1<<FROMdB2_SHIFT)))-.5f);
return (i<0)?1.f:
((i>=(FROMdB_LOOKUP_SZ<<FROMdB_SHIFT))?0.f:
FROMdB_LOOKUP[i>>FROMdB_SHIFT]*FROMdB2_LOOKUP[i&FROMdB2_MASK]);
}
#endif
#ifdef INT_LOOKUP
/* interpolated 1./sqrt(p) where .5 <= a < 1. (.100000... to .111111...) in
16.16 format
returns in m.8 format */
long vorbis_invsqlook_i(long a,long e){
long i=(a&0x7fff)>>(INVSQ_LOOKUP_I_SHIFT-1);
long d=(a&INVSQ_LOOKUP_I_MASK)<<(16-INVSQ_LOOKUP_I_SHIFT); /* 0.16 */
long val=INVSQ_LOOKUP_I[i]- /* 1.16 */
(((INVSQ_LOOKUP_I[i]-INVSQ_LOOKUP_I[i+1])* /* 0.16 */
d)>>16); /* result 1.16 */
e+=32;
if(e&1)val=(val*5792)>>13; /* multiply val by 1/sqrt(2) */
e=(e>>1)-8;
return(val>>e);
}
/* interpolated lookup based fromdB function, domain -140dB to 0dB only */
/* a is in n.12 format */
float vorbis_fromdBlook_i(long a){
int i=(-a)>>(12-FROMdB2_SHIFT);
return (i<0)?1.f:
((i>=(FROMdB_LOOKUP_SZ<<FROMdB_SHIFT))?0.f:
FROMdB_LOOKUP[i>>FROMdB_SHIFT]*FROMdB2_LOOKUP[i&FROMdB2_MASK]);
}
/* interpolated lookup based cos function, domain 0 to PI only */
/* a is in 0.16 format, where 0==0, 2^^16-1==PI, return 0.14 */
long vorbis_coslook_i(long a){
int i=a>>COS_LOOKUP_I_SHIFT;
int d=a&COS_LOOKUP_I_MASK;
return COS_LOOKUP_I[i]- ((d*(COS_LOOKUP_I[i]-COS_LOOKUP_I[i+1]))>>
COS_LOOKUP_I_SHIFT);
}
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/lookup.h
+32
View File
@@ -0,0 +1,32 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: lookup based functions
last mod: $Id: lookup.h 16227 2009-07-08 06:58:46Z xiphmont $
********************************************************************/
#ifndef _V_LOOKUP_H_
#ifdef FLOAT_LOOKUP
extern float vorbis_coslook(float a);
extern float vorbis_invsqlook(float a);
extern float vorbis_invsq2explook(int a);
extern float vorbis_fromdBlook(float a);
#endif
#ifdef INT_LOOKUP
extern long vorbis_invsqlook_i(long a,long e);
extern long vorbis_coslook_i(long a);
extern float vorbis_fromdBlook_i(long a);
#endif
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/lookup_data.h
+192
View File
@@ -0,0 +1,192 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: lookup data; generated by lookups.pl; edit there
last mod: $Id: lookup_data.h 16037 2009-05-26 21:10:58Z xiphmont $
********************************************************************/
#ifndef _V_LOOKUP_DATA_H_
#ifdef FLOAT_LOOKUP
#define COS_LOOKUP_SZ 128
static const float COS_LOOKUP[COS_LOOKUP_SZ+1]={
+1.0000000000000f,+0.9996988186962f,+0.9987954562052f,+0.9972904566787f,
+0.9951847266722f,+0.9924795345987f,+0.9891765099648f,+0.9852776423889f,
+0.9807852804032f,+0.9757021300385f,+0.9700312531945f,+0.9637760657954f,
+0.9569403357322f,+0.9495281805930f,+0.9415440651830f,+0.9329927988347f,
+0.9238795325113f,+0.9142097557035f,+0.9039892931234f,+0.8932243011955f,
+0.8819212643484f,+0.8700869911087f,+0.8577286100003f,+0.8448535652497f,
+0.8314696123025f,+0.8175848131516f,+0.8032075314806f,+0.7883464276266f,
+0.7730104533627f,+0.7572088465065f,+0.7409511253550f,+0.7242470829515f,
+0.7071067811865f,+0.6895405447371f,+0.6715589548470f,+0.6531728429538f,
+0.6343932841636f,+0.6152315905806f,+0.5956993044924f,+0.5758081914178f,
+0.5555702330196f,+0.5349976198871f,+0.5141027441932f,+0.4928981922298f,
+0.4713967368260f,+0.4496113296546f,+0.4275550934303f,+0.4052413140050f,
+0.3826834323651f,+0.3598950365350f,+0.3368898533922f,+0.3136817403989f,
+0.2902846772545f,+0.2667127574749f,+0.2429801799033f,+0.2191012401569f,
+0.1950903220161f,+0.1709618887603f,+0.1467304744554f,+0.1224106751992f,
+0.0980171403296f,+0.0735645635997f,+0.0490676743274f,+0.0245412285229f,
+0.0000000000000f,-0.0245412285229f,-0.0490676743274f,-0.0735645635997f,
-0.0980171403296f,-0.1224106751992f,-0.1467304744554f,-0.1709618887603f,
-0.1950903220161f,-0.2191012401569f,-0.2429801799033f,-0.2667127574749f,
-0.2902846772545f,-0.3136817403989f,-0.3368898533922f,-0.3598950365350f,
-0.3826834323651f,-0.4052413140050f,-0.4275550934303f,-0.4496113296546f,
-0.4713967368260f,-0.4928981922298f,-0.5141027441932f,-0.5349976198871f,
-0.5555702330196f,-0.5758081914178f,-0.5956993044924f,-0.6152315905806f,
-0.6343932841636f,-0.6531728429538f,-0.6715589548470f,-0.6895405447371f,
-0.7071067811865f,-0.7242470829515f,-0.7409511253550f,-0.7572088465065f,
-0.7730104533627f,-0.7883464276266f,-0.8032075314806f,-0.8175848131516f,
-0.8314696123025f,-0.8448535652497f,-0.8577286100003f,-0.8700869911087f,
-0.8819212643484f,-0.8932243011955f,-0.9039892931234f,-0.9142097557035f,
-0.9238795325113f,-0.9329927988347f,-0.9415440651830f,-0.9495281805930f,
-0.9569403357322f,-0.9637760657954f,-0.9700312531945f,-0.9757021300385f,
-0.9807852804032f,-0.9852776423889f,-0.9891765099648f,-0.9924795345987f,
-0.9951847266722f,-0.9972904566787f,-0.9987954562052f,-0.9996988186962f,
-1.0000000000000f,
};
#define INVSQ_LOOKUP_SZ 32
static const float INVSQ_LOOKUP[INVSQ_LOOKUP_SZ+1]={
1.414213562373f,1.392621247646f,1.371988681140f,1.352246807566f,
1.333333333333f,1.315191898443f,1.297771369046f,1.281025230441f,
1.264911064067f,1.249390095109f,1.234426799697f,1.219988562661f,
1.206045378311f,1.192569588000f,1.179535649239f,1.166919931983f,
1.154700538379f,1.142857142857f,1.131370849898f,1.120224067222f,
1.109400392450f,1.098884511590f,1.088662107904f,1.078719779941f,
1.069044967650f,1.059625885652f,1.050451462878f,1.041511287847f,
1.032795558989f,1.024295039463f,1.016001016002f,1.007905261358f,
1.000000000000f,
};
#define INVSQ2EXP_LOOKUP_MIN (-32)
#define INVSQ2EXP_LOOKUP_MAX 32
static const float INVSQ2EXP_LOOKUP[INVSQ2EXP_LOOKUP_MAX-\
INVSQ2EXP_LOOKUP_MIN+1]={
65536.f, 46340.95001f, 32768.f, 23170.47501f,
16384.f, 11585.2375f, 8192.f, 5792.618751f,
4096.f, 2896.309376f, 2048.f, 1448.154688f,
1024.f, 724.0773439f, 512.f, 362.038672f,
256.f, 181.019336f, 128.f, 90.50966799f,
64.f, 45.254834f, 32.f, 22.627417f,
16.f, 11.3137085f, 8.f, 5.656854249f,
4.f, 2.828427125f, 2.f, 1.414213562f,
1.f, 0.7071067812f, 0.5f, 0.3535533906f,
0.25f, 0.1767766953f, 0.125f, 0.08838834765f,
0.0625f, 0.04419417382f, 0.03125f, 0.02209708691f,
0.015625f, 0.01104854346f, 0.0078125f, 0.005524271728f,
0.00390625f, 0.002762135864f, 0.001953125f, 0.001381067932f,
0.0009765625f, 0.000690533966f, 0.00048828125f, 0.000345266983f,
0.000244140625f,0.0001726334915f,0.0001220703125f,8.631674575e-05f,
6.103515625e-05f,4.315837288e-05f,3.051757812e-05f,2.157918644e-05f,
1.525878906e-05f,
};
#endif
#define FROMdB_LOOKUP_SZ 35
#define FROMdB2_LOOKUP_SZ 32
#define FROMdB_SHIFT 5
#define FROMdB2_SHIFT 3
#define FROMdB2_MASK 31
#ifdef FLOAT_LOOKUP
static const float FROMdB_LOOKUP[FROMdB_LOOKUP_SZ]={
1.f, 0.6309573445f, 0.3981071706f, 0.2511886432f,
0.1584893192f, 0.1f, 0.06309573445f, 0.03981071706f,
0.02511886432f, 0.01584893192f, 0.01f, 0.006309573445f,
0.003981071706f, 0.002511886432f, 0.001584893192f, 0.001f,
0.0006309573445f,0.0003981071706f,0.0002511886432f,0.0001584893192f,
0.0001f,6.309573445e-05f,3.981071706e-05f,2.511886432e-05f,
1.584893192e-05f, 1e-05f,6.309573445e-06f,3.981071706e-06f,
2.511886432e-06f,1.584893192e-06f, 1e-06f,6.309573445e-07f,
3.981071706e-07f,2.511886432e-07f,1.584893192e-07f,
};
static const float FROMdB2_LOOKUP[FROMdB2_LOOKUP_SZ]={
0.9928302478f, 0.9786445908f, 0.9646616199f, 0.9508784391f,
0.9372921937f, 0.92390007f, 0.9106992942f, 0.8976871324f,
0.8848608897f, 0.8722179097f, 0.8597555737f, 0.8474713009f,
0.835362547f, 0.8234268041f, 0.8116616003f, 0.8000644989f,
0.7886330981f, 0.7773650302f, 0.7662579617f, 0.755309592f,
0.7445176537f, 0.7338799116f, 0.7233941627f, 0.7130582353f,
0.7028699885f, 0.6928273125f, 0.6829281272f, 0.6731703824f,
0.6635520573f, 0.6540711597f, 0.6447257262f, 0.6355138211f,
};
#endif
#ifdef INT_LOOKUP
#define INVSQ_LOOKUP_I_SHIFT 10
#define INVSQ_LOOKUP_I_MASK 1023
static const long INVSQ_LOOKUP_I[64+1]={
92682l, 91966l, 91267l, 90583l,
89915l, 89261l, 88621l, 87995l,
87381l, 86781l, 86192l, 85616l,
85051l, 84497l, 83953l, 83420l,
82897l, 82384l, 81880l, 81385l,
80899l, 80422l, 79953l, 79492l,
79039l, 78594l, 78156l, 77726l,
77302l, 76885l, 76475l, 76072l,
75674l, 75283l, 74898l, 74519l,
74146l, 73778l, 73415l, 73058l,
72706l, 72359l, 72016l, 71679l,
71347l, 71019l, 70695l, 70376l,
70061l, 69750l, 69444l, 69141l,
68842l, 68548l, 68256l, 67969l,
67685l, 67405l, 67128l, 66855l,
66585l, 66318l, 66054l, 65794l,
65536l,
};
#define COS_LOOKUP_I_SHIFT 9
#define COS_LOOKUP_I_MASK 511
#define COS_LOOKUP_I_SZ 128
static const long COS_LOOKUP_I[COS_LOOKUP_I_SZ+1]={
16384l, 16379l, 16364l, 16340l,
16305l, 16261l, 16207l, 16143l,
16069l, 15986l, 15893l, 15791l,
15679l, 15557l, 15426l, 15286l,
15137l, 14978l, 14811l, 14635l,
14449l, 14256l, 14053l, 13842l,
13623l, 13395l, 13160l, 12916l,
12665l, 12406l, 12140l, 11866l,
11585l, 11297l, 11003l, 10702l,
10394l, 10080l, 9760l, 9434l,
9102l, 8765l, 8423l, 8076l,
7723l, 7366l, 7005l, 6639l,
6270l, 5897l, 5520l, 5139l,
4756l, 4370l, 3981l, 3590l,
3196l, 2801l, 2404l, 2006l,
1606l, 1205l, 804l, 402l,
0l, -401l, -803l, -1204l,
-1605l, -2005l, -2403l, -2800l,
-3195l, -3589l, -3980l, -4369l,
-4755l, -5138l, -5519l, -5896l,
-6269l, -6638l, -7004l, -7365l,
-7722l, -8075l, -8422l, -8764l,
-9101l, -9433l, -9759l, -10079l,
-10393l, -10701l, -11002l, -11296l,
-11584l, -11865l, -12139l, -12405l,
-12664l, -12915l, -13159l, -13394l,
-13622l, -13841l, -14052l, -14255l,
-14448l, -14634l, -14810l, -14977l,
-15136l, -15285l, -15425l, -15556l,
-15678l, -15790l, -15892l, -15985l,
-16068l, -16142l, -16206l, -16260l,
-16304l, -16339l, -16363l, -16378l,
-16383l,
};
#endif
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/lpc.c
+160
View File
@@ -0,0 +1,160 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: LPC low level routines
last mod: $Id: lpc.c 16227 2009-07-08 06:58:46Z xiphmont $
********************************************************************/
/* Some of these routines (autocorrelator, LPC coefficient estimator)
are derived from code written by Jutta Degener and Carsten Bormann;
thus we include their copyright below. The entirety of this file
is freely redistributable on the condition that both of these
copyright notices are preserved without modification. */
/* Preserved Copyright: *********************************************/
/* Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
Technische Universita"t Berlin
Any use of this software is permitted provided that this notice is not
removed and that neither the authors nor the Technische Universita"t
Berlin are deemed to have made any representations as to the
suitability of this software for any purpose nor are held responsible
for any defects of this software. THERE IS ABSOLUTELY NO WARRANTY FOR
THIS SOFTWARE.
As a matter of courtesy, the authors request to be informed about uses
this software has found, about bugs in this software, and about any
improvements that may be of general interest.
Berlin, 28.11.1994
Jutta Degener
Carsten Bormann
*********************************************************************/
#include <stdlib.h>
#include <string.h>
#include <math.h>
#include "os.h"
#include "smallft.h"
#include "lpc.h"
#include "scales.h"
#include "misc.h"
/* Autocorrelation LPC coeff generation algorithm invented by
N. Levinson in 1947, modified by J. Durbin in 1959. */
/* Input : n elements of time doamin data
Output: m lpc coefficients, excitation energy */
float vorbis_lpc_from_data(float *data,float *lpci,int n,int m){
double *aut=(double*)alloca(sizeof(*aut)*(m+1));
double *lpc=(double*)alloca(sizeof(*lpc)*(m));
double error;
double epsilon;
int i,j;
/* autocorrelation, p+1 lag coefficients */
j=m+1;
while(j--){
double d=0; /* double needed for accumulator depth */
for(i=j;i<n;i++)d+=(double)data[i]*data[i-j];
aut[j]=d;
}
/* Generate lpc coefficients from autocorr values */
/* set our noise floor to about -100dB */
error=aut[0] * (1. + 1e-10);
epsilon=1e-9*aut[0]+1e-10;
for(i=0;i<m;i++){
double r= -aut[i+1];
if(error<epsilon){
memset(lpc+i,0,(m-i)*sizeof(*lpc));
goto done;
}
/* Sum up this iteration's reflection coefficient; note that in
Vorbis we don't save it. If anyone wants to recycle this code
and needs reflection coefficients, save the results of 'r' from
each iteration. */
for(j=0;j<i;j++)r-=lpc[j]*aut[i-j];
r/=error;
/* Update LPC coefficients and total error */
lpc[i]=r;
for(j=0;j<i/2;j++){
double tmp=lpc[j];
lpc[j]+=r*lpc[i-1-j];
lpc[i-1-j]+=r*tmp;
}
if(i&1)lpc[j]+=lpc[j]*r;
error*=1.-r*r;
}
done:
/* slightly damp the filter */
{
double g = .99;
double damp = g;
for(j=0;j<m;j++){
lpc[j]*=damp;
damp*=g;
}
}
for(j=0;j<m;j++)lpci[j]=(float)lpc[j];
/* we need the error value to know how big an impulse to hit the
filter with later */
return error;
}
void vorbis_lpc_predict(float *coeff,float *prime,int m,
float *data,long n){
/* in: coeff[0...m-1] LPC coefficients
prime[0...m-1] initial values (allocated size of n+m-1)
out: data[0...n-1] data samples */
long i,j,o,p;
float y;
float *work=(float*)alloca(sizeof(*work)*(m+n));
if(!prime)
for(i=0;i<m;i++)
work[i]=0.f;
else
for(i=0;i<m;i++)
work[i]=prime[i];
for(i=0;i<n;i++){
y=0;
o=i;
p=m;
for(j=0;j<m;j++)
y-=work[o++]*coeff[--p];
data[i]=work[o]=y;
}
}
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/lpc.h
+29
View File
@@ -0,0 +1,29 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2007 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: LPC low level routines
last mod: $Id: lpc.h 16037 2009-05-26 21:10:58Z xiphmont $
********************************************************************/
#ifndef _V_LPC_H_
#define _V_LPC_H_
#include "../../codec.h"
/* simple linear scale LPC code */
extern float vorbis_lpc_from_data(float *data,float *lpc,int n,int m);
extern void vorbis_lpc_predict(float *coeff,float *prime,int m,
float *data,long n);
#endif
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/lsp.c
+454
View File
@@ -0,0 +1,454 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: LSP (also called LSF) conversion routines
last mod: $Id: lsp.c 17538 2010-10-15 02:52:29Z tterribe $
The LSP generation code is taken (with minimal modification and a
few bugfixes) from "On the Computation of the LSP Frequencies" by
Joseph Rothweiler (see http://www.rothweiler.us for contact info).
The paper is available at:
http://www.myown1.com/joe/lsf
********************************************************************/
/* Note that the lpc-lsp conversion finds the roots of polynomial with
an iterative root polisher (CACM algorithm 283). It *is* possible
to confuse this algorithm into not converging; that should only
happen with absurdly closely spaced roots (very sharp peaks in the
LPC f response) which in turn should be impossible in our use of
the code. If this *does* happen anyway, it's a bug in the floor
finder; find the cause of the confusion (probably a single bin
spike or accidental near-float-limit resolution problems) and
correct it. */
#include <math.h>
#include <string.h>
#include <stdlib.h>
#include "lsp.h"
#include "os.h"
#include "misc.h"
#include "lookup.h"
#include "scales.h"
/* three possible LSP to f curve functions; the exact computation
(float), a lookup based float implementation, and an integer
implementation. The float lookup is likely the optimal choice on
any machine with an FPU. The integer implementation is *not* fixed
point (due to the need for a large dynamic range and thus a
separately tracked exponent) and thus much more complex than the
relatively simple float implementations. It's mostly for future
work on a fully fixed point implementation for processors like the
ARM family. */
/* define either of these (preferably FLOAT_LOOKUP) to have faster
but less precise implementation. */
#undef FLOAT_LOOKUP
#undef INT_LOOKUP
#ifdef FLOAT_LOOKUP
#include "lookup.c" /* catch this in the build system; we #include for
compilers (like gcc) that can't inline across
modules */
/* side effect: changes *lsp to cosines of lsp */
void vorbis_lsp_to_curve(float *curve,int *map,int n,int ln,float *lsp,int m,
float amp,float ampoffset){
int i;
float wdel=M_PI/ln;
vorbis_fpu_control fpu;
vorbis_fpu_setround(&fpu);
for(i=0;i<m;i++)lsp[i]=vorbis_coslook(lsp[i]);
i=0;
while(i<n){
int k=map[i];
int qexp;
float p=.7071067812f;
float q=.7071067812f;
float w=vorbis_coslook(wdel*k);
float *ftmp=lsp;
int c=m>>1;
while(c--){
q*=ftmp[0]-w;
p*=ftmp[1]-w;
ftmp+=2;
}
if(m&1){
/* odd order filter; slightly assymetric */
/* the last coefficient */
q*=ftmp[0]-w;
q*=q;
p*=p*(1.f-w*w);
}else{
/* even order filter; still symmetric */
q*=q*(1.f+w);
p*=p*(1.f-w);
}
q=frexp(p+q,&qexp);
q=vorbis_fromdBlook(amp*
vorbis_invsqlook(q)*
vorbis_invsq2explook(qexp+m)-
ampoffset);
do{
curve[i++]*=q;
}while(map[i]==k);
}
vorbis_fpu_restore(fpu);
}
#else
#ifdef INT_LOOKUP
#include "lookup.c" /* catch this in the build system; we #include for
compilers (like gcc) that can't inline across
modules */
static const int MLOOP_1[64]={
0,10,11,11, 12,12,12,12, 13,13,13,13, 13,13,13,13,
14,14,14,14, 14,14,14,14, 14,14,14,14, 14,14,14,14,
15,15,15,15, 15,15,15,15, 15,15,15,15, 15,15,15,15,
15,15,15,15, 15,15,15,15, 15,15,15,15, 15,15,15,15,
};
static const int MLOOP_2[64]={
0,4,5,5, 6,6,6,6, 7,7,7,7, 7,7,7,7,
8,8,8,8, 8,8,8,8, 8,8,8,8, 8,8,8,8,
9,9,9,9, 9,9,9,9, 9,9,9,9, 9,9,9,9,
9,9,9,9, 9,9,9,9, 9,9,9,9, 9,9,9,9,
};
static const int MLOOP_3[8]={0,1,2,2,3,3,3,3};
/* side effect: changes *lsp to cosines of lsp */
void vorbis_lsp_to_curve(float *curve,int *map,int n,int ln,float *lsp,int m,
float amp,float ampoffset){
/* 0 <= m < 256 */
/* set up for using all int later */
int i;
int ampoffseti=rint(ampoffset*4096.f);
int ampi=rint(amp*16.f);
long *ilsp=(long*)alloca(m*sizeof(*ilsp));
for(i=0;i<m;i++)ilsp[i]=vorbis_coslook_i(lsp[i]/M_PI*65536.f+.5f);
i=0;
while(i<n){
int j,k=map[i];
unsigned long pi=46341; /* 2**-.5 in 0.16 */
unsigned long qi=46341;
int qexp=0,shift;
long wi=vorbis_coslook_i(k*65536/ln);
qi*=labs(ilsp[0]-wi);
pi*=labs(ilsp[1]-wi);
for(j=3;j<m;j+=2){
if(!(shift=MLOOP_1[(pi|qi)>>25]))
if(!(shift=MLOOP_2[(pi|qi)>>19]))
shift=MLOOP_3[(pi|qi)>>16];
qi=(qi>>shift)*labs(ilsp[j-1]-wi);
pi=(pi>>shift)*labs(ilsp[j]-wi);
qexp+=shift;
}
if(!(shift=MLOOP_1[(pi|qi)>>25]))
if(!(shift=MLOOP_2[(pi|qi)>>19]))
shift=MLOOP_3[(pi|qi)>>16];
/* pi,qi normalized collectively, both tracked using qexp */
if(m&1){
/* odd order filter; slightly assymetric */
/* the last coefficient */
qi=(qi>>shift)*labs(ilsp[j-1]-wi);
pi=(pi>>shift)<<14;
qexp+=shift;
if(!(shift=MLOOP_1[(pi|qi)>>25]))
if(!(shift=MLOOP_2[(pi|qi)>>19]))
shift=MLOOP_3[(pi|qi)>>16];
pi>>=shift;
qi>>=shift;
qexp+=shift-14*((m+1)>>1);
pi=((pi*pi)>>16);
qi=((qi*qi)>>16);
qexp=qexp*2+m;
pi*=(1<<14)-((wi*wi)>>14);
qi+=pi>>14;
}else{
/* even order filter; still symmetric */
/* p*=p(1-w), q*=q(1+w), let normalization drift because it isn't
worth tracking step by step */
pi>>=shift;
qi>>=shift;
qexp+=shift-7*m;
pi=((pi*pi)>>16);
qi=((qi*qi)>>16);
qexp=qexp*2+m;
pi*=(1<<14)-wi;
qi*=(1<<14)+wi;
qi=(qi+pi)>>14;
}
/* we've let the normalization drift because it wasn't important;
however, for the lookup, things must be normalized again. We
need at most one right shift or a number of left shifts */
if(qi&0xffff0000){ /* checks for 1.xxxxxxxxxxxxxxxx */
qi>>=1; qexp++;
}else
while(qi && !(qi&0x8000)){ /* checks for 0.0xxxxxxxxxxxxxxx or less*/
qi<<=1; qexp--;
}
amp=vorbis_fromdBlook_i(ampi* /* n.4 */
vorbis_invsqlook_i(qi,qexp)-
/* m.8, m+n<=8 */
ampoffseti); /* 8.12[0] */
curve[i]*=amp;
while(map[++i]==k)curve[i]*=amp;
}
}
#else
/* old, nonoptimized but simple version for any poor sap who needs to
figure out what the hell this code does, or wants the other
fraction of a dB precision */
/* side effect: changes *lsp to cosines of lsp */
void vorbis_lsp_to_curve(float *curve,int *map,int n,int ln,float *lsp,int m,
float amp,float ampoffset){
int i;
float wdel=M_PI/ln;
for(i=0;i<m;i++)lsp[i]=2.f*cos(lsp[i]);
i=0;
while(i<n){
int j,k=map[i];
float p=.5f;
float q=.5f;
float w=2.f*cos(wdel*k);
for(j=1;j<m;j+=2){
q *= w-lsp[j-1];
p *= w-lsp[j];
}
if(j==m){
/* odd order filter; slightly assymetric */
/* the last coefficient */
q*=w-lsp[j-1];
p*=p*(4.f-w*w);
q*=q;
}else{
/* even order filter; still symmetric */
p*=p*(2.f-w);
q*=q*(2.f+w);
}
q=fromdB(amp/sqrt(p+q)-ampoffset);
curve[i]*=q;
while(map[++i]==k)curve[i]*=q;
}
}
#endif
#endif
static void cheby(float *g, int ord) {
int i, j;
g[0] *= .5f;
for(i=2; i<= ord; i++) {
for(j=ord; j >= i; j--) {
g[j-2] -= g[j];
g[j] += g[j];
}
}
}
static int JUCE_CDECL comp(const void *a,const void *b){
return (*(float *)a<*(float *)b)-(*(float *)a>*(float *)b);
}
/* Newton-Raphson-Maehly actually functioned as a decent root finder,
but there are root sets for which it gets into limit cycles
(exacerbated by zero suppression) and fails. We can't afford to
fail, even if the failure is 1 in 100,000,000, so we now use
Laguerre and later polish with Newton-Raphson (which can then
afford to fail) */
#define EPSILON 10e-7
static int Laguerre_With_Deflation(float *a,int ord,float *r){
int i,m;
double *defl=(double*)alloca(sizeof(*defl)*(ord+1));
for(i=0;i<=ord;i++)defl[i]=a[i];
for(m=ord;m>0;m--){
double newx=0.f,delta;
/* iterate a root */
while(1){
double p=defl[m],pp=0.f,ppp=0.f,denom;
/* eval the polynomial and its first two derivatives */
for(i=m;i>0;i--){
ppp = newx*ppp + pp;
pp = newx*pp + p;
p = newx*p + defl[i-1];
}
/* Laguerre's method */
denom=(m-1) * ((m-1)*pp*pp - m*p*ppp);
if(denom<0)
return(-1); /* complex root! The LPC generator handed us a bad filter */
if(pp>0){
denom = pp + sqrt(denom);
if(denom<EPSILON)denom=EPSILON;
}else{
denom = pp - sqrt(denom);
if(denom>-(EPSILON))denom=-(EPSILON);
}
delta = m*p/denom;
newx -= delta;
if(delta<0.f)delta*=-1;
if(fabs(delta/newx)<10e-12)break;
}
r[m-1]=newx;
/* forward deflation */
for(i=m;i>0;i--)
defl[i-1]+=newx*defl[i];
defl++;
}
return(0);
}
/* for spit-and-polish only */
static int Newton_Raphson(float *a,int ord,float *r){
int i, k, count=0;
double error=1.f;
double *root=(double*)alloca(ord*sizeof(*root));
for(i=0; i<ord;i++) root[i] = r[i];
while(error>1e-20){
error=0;
for(i=0; i<ord; i++) { /* Update each point. */
double pp=0.,delta;
double rooti=root[i];
double p=a[ord];
for(k=ord-1; k>= 0; k--) {
pp= pp* rooti + p;
p = p * rooti + a[k];
}
delta = p/pp;
root[i] -= delta;
error+= delta*delta;
}
if(count>40)return(-1);
count++;
}
/* Replaced the original bubble sort with a real sort. With your
help, we can eliminate the bubble sort in our lifetime. --Monty */
for(i=0; i<ord;i++) r[i] = root[i];
return(0);
}
/* Convert lpc coefficients to lsp coefficients */
int vorbis_lpc_to_lsp(float *lpc,float *lsp,int m){
int order2=(m+1)>>1;
int g1_order,g2_order;
float *g1=(float*)alloca(sizeof(*g1)*(order2+1));
float *g2=(float*)alloca(sizeof(*g2)*(order2+1));
float *g1r=(float*)alloca(sizeof(*g1r)*(order2+1));
float *g2r=(float*)alloca(sizeof(*g2r)*(order2+1));
int i;
/* even and odd are slightly different base cases */
g1_order=(m+1)>>1;
g2_order=(m) >>1;
/* Compute the lengths of the x polynomials. */
/* Compute the first half of K & R F1 & F2 polynomials. */
/* Compute half of the symmetric and antisymmetric polynomials. */
/* Remove the roots at +1 and -1. */
g1[g1_order] = 1.f;
for(i=1;i<=g1_order;i++) g1[g1_order-i] = lpc[i-1]+lpc[m-i];
g2[g2_order] = 1.f;
for(i=1;i<=g2_order;i++) g2[g2_order-i] = lpc[i-1]-lpc[m-i];
if(g1_order>g2_order){
for(i=2; i<=g2_order;i++) g2[g2_order-i] += g2[g2_order-i+2];
}else{
for(i=1; i<=g1_order;i++) g1[g1_order-i] -= g1[g1_order-i+1];
for(i=1; i<=g2_order;i++) g2[g2_order-i] += g2[g2_order-i+1];
}
/* Convert into polynomials in cos(alpha) */
cheby(g1,g1_order);
cheby(g2,g2_order);
/* Find the roots of the 2 even polynomials.*/
if(Laguerre_With_Deflation(g1,g1_order,g1r) ||
Laguerre_With_Deflation(g2,g2_order,g2r))
return(-1);
Newton_Raphson(g1,g1_order,g1r); /* if it fails, it leaves g1r alone */
Newton_Raphson(g2,g2_order,g2r); /* if it fails, it leaves g2r alone */
qsort(g1r,g1_order,sizeof(*g1r),comp);
qsort(g2r,g2_order,sizeof(*g2r),comp);
for(i=0;i<g1_order;i++)
lsp[i*2] = acos(g1r[i]);
for(i=0;i<g2_order;i++)
lsp[i*2+1] = acos(g2r[i]);
return(0);
}
JuceLibraryCode/modules/juce_audio_formats/codecs/oggvorbis/libvorbis-1.3.2/lib/lsp.h
+28
View File
@@ -0,0 +1,28 @@
/********************************************************************
* *
* THIS FILE IS PART OF THE OggVorbis SOFTWARE CODEC SOURCE CODE. *
* USE, DISTRIBUTION AND REPRODUCTION OF THIS LIBRARY SOURCE IS *
* GOVERNED BY A BSD-STYLE SOURCE LICENSE INCLUDED WITH THIS SOURCE *
* IN 'COPYING'. PLEASE READ THESE TERMS BEFORE DISTRIBUTING. *
* *
* THE OggVorbis SOURCE CODE IS (C) COPYRIGHT 1994-2009 *
* by the Xiph.Org Foundation http://www.xiph.org/ *
* *
********************************************************************
function: LSP (also called LSF) conversion routines
last mod: $Id: lsp.h 16227 2009-07-08 06:58:46Z xiphmont $
********************************************************************/
#ifndef _V_LSP_H_
#define _V_LSP_H_
extern int vorbis_lpc_to_lsp(float *lpc,float *lsp,int m);
extern void vorbis_lsp_to_curve(float *curve,int *map,int n,int ln,
float *lsp,int m,
float amp,float ampoffset);
#endif

Some files were not shown because too many files have changed in this diff Show More