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Updated FFmpeg.

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This commit is contained in:
Christopher Snowhill 2017-09-24 16:35:45 -07:00
parent d3ef15de82
commit ba78e14909
16 changed files with 685 additions and 13 deletions
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4
Frameworks/vgmstream/vgmstream.xcodeproj/project.pbxproj
View file

@ -391,6 +391,7 @@
83BAFB6C19F45EB3005DAB60 /* bfstm.c in Sources */ = {isa = PBXBuildFile; fileRef = 83BAFB6B19F45EB3005DAB60 /* bfstm.c */; };
83CAB8E21F0B0752001BC993 /* wii_04sw.c in Sources */ = {isa = PBXBuildFile; fileRef = 83CAB8E11F0B0745001BC993 /* wii_04sw.c */; };
83CAB8E31F0B0755001BC993 /* pc_xa30.c in Sources */ = {isa = PBXBuildFile; fileRef = 83CAB8DC1F0B0744001BC993 /* pc_xa30.c */; };
83CD428A1F787879000F77BE /* libswresample.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 83CD42851F787878000F77BE /* libswresample.a */; };
83D731101A7394BF00CA1366 /* g7221.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 83D730EB1A738EB300CA1366 /* g7221.framework */; };
83D731111A7394D300CA1366 /* g7221.framework in CopyFiles */ = {isa = PBXBuildFile; fileRef = 83D730EB1A738EB300CA1366 /* g7221.framework */; settings = {ATTRIBUTES = (CodeSignOnCopy, RemoveHeadersOnCopy, ); }; };
83D731891A749D1500CA1366 /* g719.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 83D7313E1A74968A00CA1366 /* g719.framework */; };
@ -890,6 +891,7 @@
83BAFB6B19F45EB3005DAB60 /* bfstm.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = bfstm.c; sourceTree = "<group>"; };
83CAB8DC1F0B0744001BC993 /* pc_xa30.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; path = pc_xa30.c; sourceTree = "<group>"; };
83CAB8E11F0B0745001BC993 /* wii_04sw.c */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.c; path = wii_04sw.c; sourceTree = "<group>"; };
83CD42851F787878000F77BE /* libswresample.a */ = {isa = PBXFileReference; lastKnownFileType = archive.ar; name = libswresample.a; path = ../../ThirdParty/ffmpeg/lib/libswresample.a; sourceTree = "<group>"; };
83D730E51A738EB200CA1366 /* g7221.xcodeproj */ = {isa = PBXFileReference; lastKnownFileType = "wrapper.pb-project"; name = g7221.xcodeproj; path = ../g7221/g7221.xcodeproj; sourceTree = "<group>"; };
83D731381A74968900CA1366 /* g719.xcodeproj */ = {isa = PBXFileReference; lastKnownFileType = "wrapper.pb-project"; name = g719.xcodeproj; path = ../g719/g719.xcodeproj; sourceTree = "<group>"; };
83D7318B1A749EEE00CA1366 /* g719_decoder.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = g719_decoder.c; sourceTree = "<group>"; };
@ -916,6 +918,7 @@
838BDB711D3B1F990022CA6F /* CoreFoundation.framework in Frameworks */,
838BDB6A1D3AF7140022CA6F /* libiconv.tbd in Frameworks */,
838BDB681D3AF70D0022CA6F /* libz.tbd in Frameworks */,
83CD428A1F787879000F77BE /* libswresample.a in Frameworks */,
83D731891A749D1500CA1366 /* g719.framework in Frameworks */,
83D731101A7394BF00CA1366 /* g7221.framework in Frameworks */,
838BDB641D3AF08C0022CA6F /* libavcodec.a in Frameworks */,
@ -988,6 +991,7 @@
838BDB611D3AF08C0022CA6F /* libavcodec.a */,
838BDB621D3AF08C0022CA6F /* libavformat.a */,
838BDB631D3AF08C0022CA6F /* libavutil.a */,
83CD42851F787878000F77BE /* libswresample.a */,
8313E33D1901FBDC00B4B6F1 /* mpg123.xcodeproj */,
83F412871E932F9A002E37D0 /* Vorbis.xcodeproj */,
83D730E51A738EB200CA1366 /* g7221.xcodeproj */,

4
Plugins/FFMPEG/FFMPEG.xcodeproj/project.pbxproj
View file

@ -19,6 +19,7 @@
8352D49B1CDDB8B2009D16AA /* VideoToolbox.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 8352D49A1CDDB8B2009D16AA /* VideoToolbox.framework */; };
8352D49D1CDDB8C0009D16AA /* CoreMedia.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 8352D49C1CDDB8C0009D16AA /* CoreMedia.framework */; };
8352D49F1CDDB8D7009D16AA /* CoreVideo.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 8352D49E1CDDB8D7009D16AA /* CoreVideo.framework */; };
83CD428C1F7878A0000F77BE /* libswresample.a in Frameworks */ = {isa = PBXBuildFile; fileRef = 83CD428B1F78789F000F77BE /* libswresample.a */; };
8D5B49B4048680CD000E48DA /* Cocoa.framework in Frameworks */ = {isa = PBXBuildFile; fileRef = 1058C7ADFEA557BF11CA2CBB /* Cocoa.framework */; };
B09E942F0D747F410064F138 /* FFMPEGDecoder.m in Sources */ = {isa = PBXBuildFile; fileRef = B09E942E0D747F410064F138 /* FFMPEGDecoder.m */; };
/* End PBXBuildFile section */
@ -41,6 +42,7 @@
8352D49C1CDDB8C0009D16AA /* CoreMedia.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = CoreMedia.framework; path = System/Library/Frameworks/CoreMedia.framework; sourceTree = SDKROOT; };
8352D49E1CDDB8D7009D16AA /* CoreVideo.framework */ = {isa = PBXFileReference; lastKnownFileType = wrapper.framework; name = CoreVideo.framework; path = System/Library/Frameworks/CoreVideo.framework; sourceTree = SDKROOT; };
8384913818081F6C00E7332D /* Logging.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = Logging.h; path = ../../Utils/Logging.h; sourceTree = "<group>"; };
83CD428B1F78789F000F77BE /* libswresample.a */ = {isa = PBXFileReference; lastKnownFileType = archive.ar; name = libswresample.a; path = ../../ThirdParty/ffmpeg/lib/libswresample.a; sourceTree = "<group>"; };
8D5B49B6048680CD000E48DA /* FFMPEG.bundle */ = {isa = PBXFileReference; explicitFileType = wrapper.cfbundle; includeInIndex = 0; path = FFMPEG.bundle; sourceTree = BUILT_PRODUCTS_DIR; };
8D5B49B7048680CD000E48DA /* Info.plist */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text.plist.xml; path = Info.plist; sourceTree = "<group>"; };
B09E942D0D747F410064F138 /* FFMPEGDecoder.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = FFMPEGDecoder.h; sourceTree = "<group>"; };
@ -67,6 +69,7 @@
8352D48C1CDDAEDD009D16AA /* libavformat.a in Frameworks */,
8352D48D1CDDAEDD009D16AA /* libavutil.a in Frameworks */,
8352D48B1CDDAEDD009D16AA /* libavcodec.a in Frameworks */,
83CD428C1F7878A0000F77BE /* libswresample.a in Frameworks */,
8352D4951CDDB03E009D16AA /* libiconv.tbd in Frameworks */,
);
runOnlyForDeploymentPostprocessing = 0;
@ -95,6 +98,7 @@
8352D4881CDDAEDD009D16AA /* libavcodec.a */,
8352D4891CDDAEDD009D16AA /* libavformat.a */,
8352D48A1CDDAEDD009D16AA /* libavutil.a */,
83CD428B1F78789F000F77BE /* libswresample.a */,
1058C7ACFEA557BF11CA2CBB /* Linked Frameworks */,
1058C7AEFEA557BF11CA2CBB /* Other Frameworks */,
);

7
Scripts/ffmpeg-build.sh
View file

@ -3,12 +3,13 @@
--enable-static --disable-shared --prefix=$HOME/Source/Repos/cog/ThirdParty/ffmpeg\
--enable-pic --enable-gpl --disable-doc --disable-ffplay\
--disable-ffprobe --disable-ffserver --disable-avdevice --disable-ffmpeg\
--disable-postproc --disable-swresample --disable-avfilter\
--disable-postproc --disable-avfilter\
--disable-swscale --disable-network --disable-swscale-alpha --disable-vdpau\
--disable-dxva2 --disable-everything --enable-hwaccels\
--enable-parser=ac3,mpegaudio,xma,vorbis\
--enable-swresample\
--enable-parser=ac3,mpegaudio,xma,vorbis,opus\
--enable-demuxer=ac3,asf,xwma,mov,oma,ogg,tak,dsf,wav,aac,dts,dtshd,mp3,bink,flac,msf,xmv,caf,ape,smacker\
--enable-decoder=ac3,wmapro,wmav1,wmav2,wmavoice,wmalossless,xma1,xma2,dca,tak,dsd_lsbf,dsd_lsbf_planar,dsd_mbf,dsd_msbf_planar,aac,atrac3,atrac3p,mp3float,bink,binkaudio_dct,binkaudio_rdft,flac,pcm_s16be,pcm_s16be_planar,pcm_s16le,pcm_s16le_planar,vorbis,ape,adpcm_ima_qt,smackaud\
--enable-decoder=ac3,wmapro,wmav1,wmav2,wmavoice,wmalossless,xma1,xma2,dca,tak,dsd_lsbf,dsd_lsbf_planar,dsd_mbf,dsd_msbf_planar,aac,atrac3,atrac3p,mp3float,bink,binkaudio_dct,binkaudio_rdft,flac,pcm_s16be,pcm_s16be_planar,pcm_s16le,pcm_s16le_planar,vorbis,ape,adpcm_ima_qt,smackaud,opus\
--disable-parser=mpeg4video,h263\
--disable-decoder=mpeg2video,h263,h264,mpeg1video,mpeg2video,mpeg4,hevc,vp9\
--disable-version3

2
ThirdParty/ffmpeg/include/libavformat/version.h vendored
View file

@ -33,7 +33,7 @@
// Also please add any ticket numbers that you believe might be affected here
#define LIBAVFORMAT_VERSION_MAJOR 57
#define LIBAVFORMAT_VERSION_MINOR 82
#define LIBAVFORMAT_VERSION_MICRO 100
#define LIBAVFORMAT_VERSION_MICRO 101
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
LIBAVFORMAT_VERSION_MINOR, \

2
ThirdParty/ffmpeg/include/libavutil/ffversion.h vendored
View file

@ -1,5 +1,5 @@
/* Automatically generated by version.sh, do not manually edit! */
#ifndef AVUTIL_FFVERSION_H
#define AVUTIL_FFVERSION_H
#define FFMPEG_VERSION "N-87323-gc3a00d516f"
#define FFMPEG_VERSION "N-87392-gebb9733867"
#endif /* AVUTIL_FFVERSION_H */

25
ThirdParty/ffmpeg/include/libavutil/pixdesc.h vendored
View file

@ -250,26 +250,51 @@ int av_pix_fmt_count_planes(enum AVPixelFormat pix_fmt);
*/
const char *av_color_range_name(enum AVColorRange range);
/**
* @return the AVColorRange value for name or an AVError if not found.
*/
int av_color_range_from_name(const char *name);
/**
* @return the name for provided color primaries or NULL if unknown.
*/
const char *av_color_primaries_name(enum AVColorPrimaries primaries);
/**
* @return the AVColorPrimaries value for name or an AVError if not found.
*/
int av_color_primaries_from_name(const char *name);
/**
* @return the name for provided color transfer or NULL if unknown.
*/
const char *av_color_transfer_name(enum AVColorTransferCharacteristic transfer);
/**
* @return the AVColorTransferCharacteristic value for name or an AVError if not found.
*/
int av_color_transfer_from_name(const char *name);
/**
* @return the name for provided color space or NULL if unknown.
*/
const char *av_color_space_name(enum AVColorSpace space);
/**
* @return the AVColorSpace value for name or an AVError if not found.
*/
int av_color_space_from_name(const char *name);
/**
* @return the name for provided chroma location or NULL if unknown.
*/
const char *av_chroma_location_name(enum AVChromaLocation location);
/**
* @return the AVChromaLocation value for name or an AVError if not found.
*/
int av_chroma_location_from_name(const char *name);
/**
* Return the pixel format corresponding to name.
*

2
ThirdParty/ffmpeg/include/libavutil/version.h vendored
View file

@ -80,7 +80,7 @@
#define LIBAVUTIL_VERSION_MAJOR 55
#define LIBAVUTIL_VERSION_MINOR 75
#define LIBAVUTIL_VERSION_MINOR 76
#define LIBAVUTIL_VERSION_MICRO 100
#define LIBAVUTIL_VERSION_INT AV_VERSION_INT(LIBAVUTIL_VERSION_MAJOR, \

579
ThirdParty/ffmpeg/include/libswresample/swresample.h vendored Normal file
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@ -0,0 +1,579 @@
/*
* Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef SWRESAMPLE_SWRESAMPLE_H
#define SWRESAMPLE_SWRESAMPLE_H
/**
* @file
* @ingroup lswr
* libswresample public header
*/
/**
* @defgroup lswr libswresample
* @{
*
* Audio resampling, sample format conversion and mixing library.
*
* Interaction with lswr is done through SwrContext, which is
* allocated with swr_alloc() or swr_alloc_set_opts(). It is opaque, so all parameters
* must be set with the @ref avoptions API.
*
* The first thing you will need to do in order to use lswr is to allocate
* SwrContext. This can be done with swr_alloc() or swr_alloc_set_opts(). If you
* are using the former, you must set options through the @ref avoptions API.
* The latter function provides the same feature, but it allows you to set some
* common options in the same statement.
*
* For example the following code will setup conversion from planar float sample
* format to interleaved signed 16-bit integer, downsampling from 48kHz to
* 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing
* matrix). This is using the swr_alloc() function.
* @code
* SwrContext *swr = swr_alloc();
* av_opt_set_channel_layout(swr, "in_channel_layout", AV_CH_LAYOUT_5POINT1, 0);
* av_opt_set_channel_layout(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO, 0);
* av_opt_set_int(swr, "in_sample_rate", 48000, 0);
* av_opt_set_int(swr, "out_sample_rate", 44100, 0);
* av_opt_set_sample_fmt(swr, "in_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
* av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0);
* @endcode
*
* The same job can be done using swr_alloc_set_opts() as well:
* @code
* SwrContext *swr = swr_alloc_set_opts(NULL, // we're allocating a new context
* AV_CH_LAYOUT_STEREO, // out_ch_layout
* AV_SAMPLE_FMT_S16, // out_sample_fmt
* 44100, // out_sample_rate
* AV_CH_LAYOUT_5POINT1, // in_ch_layout
* AV_SAMPLE_FMT_FLTP, // in_sample_fmt
* 48000, // in_sample_rate
* 0, // log_offset
* NULL); // log_ctx
* @endcode
*
* Once all values have been set, it must be initialized with swr_init(). If
* you need to change the conversion parameters, you can change the parameters
* using @ref AVOptions, as described above in the first example; or by using
* swr_alloc_set_opts(), but with the first argument the allocated context.
* You must then call swr_init() again.
*
* The conversion itself is done by repeatedly calling swr_convert().
* Note that the samples may get buffered in swr if you provide insufficient
* output space or if sample rate conversion is done, which requires "future"
* samples. Samples that do not require future input can be retrieved at any
* time by using swr_convert() (in_count can be set to 0).
* At the end of conversion the resampling buffer can be flushed by calling
* swr_convert() with NULL in and 0 in_count.
*
* The samples used in the conversion process can be managed with the libavutil
* @ref lavu_sampmanip "samples manipulation" API, including av_samples_alloc()
* function used in the following example.
*
* The delay between input and output, can at any time be found by using
* swr_get_delay().
*
* The following code demonstrates the conversion loop assuming the parameters
* from above and caller-defined functions get_input() and handle_output():
* @code
* uint8_t **input;
* int in_samples;
*
* while (get_input(&input, &in_samples)) {
* uint8_t *output;
* int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) +
* in_samples, 44100, 48000, AV_ROUND_UP);
* av_samples_alloc(&output, NULL, 2, out_samples,
* AV_SAMPLE_FMT_S16, 0);
* out_samples = swr_convert(swr, &output, out_samples,
* input, in_samples);
* handle_output(output, out_samples);
* av_freep(&output);
* }
* @endcode
*
* When the conversion is finished, the conversion
* context and everything associated with it must be freed with swr_free().
* A swr_close() function is also available, but it exists mainly for
* compatibility with libavresample, and is not required to be called.
*
* There will be no memory leak if the data is not completely flushed before
* swr_free().
*/
#include <stdint.h>
#include "libavutil/channel_layout.h"
#include "libavutil/frame.h"
#include "libavutil/samplefmt.h"
#include "libswresample/version.h"
/**
* @name Option constants
* These constants are used for the @ref avoptions interface for lswr.
* @{
*
*/
#define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
//TODO use int resample ?
//long term TODO can we enable this dynamically?
/** Dithering algorithms */
enum SwrDitherType {
SWR_DITHER_NONE = 0,
SWR_DITHER_RECTANGULAR,
SWR_DITHER_TRIANGULAR,
SWR_DITHER_TRIANGULAR_HIGHPASS,
SWR_DITHER_NS = 64, ///< not part of API/ABI
SWR_DITHER_NS_LIPSHITZ,
SWR_DITHER_NS_F_WEIGHTED,
SWR_DITHER_NS_MODIFIED_E_WEIGHTED,
SWR_DITHER_NS_IMPROVED_E_WEIGHTED,
SWR_DITHER_NS_SHIBATA,
SWR_DITHER_NS_LOW_SHIBATA,
SWR_DITHER_NS_HIGH_SHIBATA,
SWR_DITHER_NB, ///< not part of API/ABI
};
/** Resampling Engines */
enum SwrEngine {
SWR_ENGINE_SWR, /**< SW Resampler */
SWR_ENGINE_SOXR, /**< SoX Resampler */
SWR_ENGINE_NB, ///< not part of API/ABI
};
/** Resampling Filter Types */
enum SwrFilterType {
SWR_FILTER_TYPE_CUBIC, /**< Cubic */
SWR_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall windowed sinc */
SWR_FILTER_TYPE_KAISER, /**< Kaiser windowed sinc */
};
/**
* @}
*/
/**
* The libswresample context. Unlike libavcodec and libavformat, this structure
* is opaque. This means that if you would like to set options, you must use
* the @ref avoptions API and cannot directly set values to members of the
* structure.
*/
typedef struct SwrContext SwrContext;
/**
* Get the AVClass for SwrContext. It can be used in combination with
* AV_OPT_SEARCH_FAKE_OBJ for examining options.
*
* @see av_opt_find().
* @return the AVClass of SwrContext
*/
const AVClass *swr_get_class(void);
/**
* @name SwrContext constructor functions
* @{
*/
/**
* Allocate SwrContext.
*
* If you use this function you will need to set the parameters (manually or
* with swr_alloc_set_opts()) before calling swr_init().
*
* @see swr_alloc_set_opts(), swr_init(), swr_free()
* @return NULL on error, allocated context otherwise
*/
struct SwrContext *swr_alloc(void);
/**
* Initialize context after user parameters have been set.
* @note The context must be configured using the AVOption API.
*
* @see av_opt_set_int()
* @see av_opt_set_dict()
*
* @param[in,out] s Swr context to initialize
* @return AVERROR error code in case of failure.
*/
int swr_init(struct SwrContext *s);
/**
* Check whether an swr context has been initialized or not.
*
* @param[in] s Swr context to check
* @see swr_init()
* @return positive if it has been initialized, 0 if not initialized
*/
int swr_is_initialized(struct SwrContext *s);
/**
* Allocate SwrContext if needed and set/reset common parameters.
*
* This function does not require s to be allocated with swr_alloc(). On the
* other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters
* on the allocated context.
*
* @param s existing Swr context if available, or NULL if not
* @param out_ch_layout output channel layout (AV_CH_LAYOUT_*)
* @param out_sample_fmt output sample format (AV_SAMPLE_FMT_*).
* @param out_sample_rate output sample rate (frequency in Hz)
* @param in_ch_layout input channel layout (AV_CH_LAYOUT_*)
* @param in_sample_fmt input sample format (AV_SAMPLE_FMT_*).
* @param in_sample_rate input sample rate (frequency in Hz)
* @param log_offset logging level offset
* @param log_ctx parent logging context, can be NULL
*
* @see swr_init(), swr_free()
* @return NULL on error, allocated context otherwise
*/
struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
int64_t in_ch_layout, enum AVSampleFormat in_sample_fmt, int in_sample_rate,
int log_offset, void *log_ctx);
/**
* @}
*
* @name SwrContext destructor functions
* @{
*/
/**
* Free the given SwrContext and set the pointer to NULL.
*
* @param[in] s a pointer to a pointer to Swr context
*/
void swr_free(struct SwrContext **s);
/**
* Closes the context so that swr_is_initialized() returns 0.
*
* The context can be brought back to life by running swr_init(),
* swr_init() can also be used without swr_close().
* This function is mainly provided for simplifying the usecase
* where one tries to support libavresample and libswresample.
*
* @param[in,out] s Swr context to be closed
*/
void swr_close(struct SwrContext *s);
/**
* @}
*
* @name Core conversion functions
* @{
*/
/** Convert audio.
*
* in and in_count can be set to 0 to flush the last few samples out at the
* end.
*
* If more input is provided than output space, then the input will be buffered.
* You can avoid this buffering by using swr_get_out_samples() to retrieve an
* upper bound on the required number of output samples for the given number of
* input samples. Conversion will run directly without copying whenever possible.
*
* @param s allocated Swr context, with parameters set
* @param out output buffers, only the first one need be set in case of packed audio
* @param out_count amount of space available for output in samples per channel
* @param in input buffers, only the first one need to be set in case of packed audio
* @param in_count number of input samples available in one channel
*
* @return number of samples output per channel, negative value on error
*/
int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,
const uint8_t **in , int in_count);
/**
* Convert the next timestamp from input to output
* timestamps are in 1/(in_sample_rate * out_sample_rate) units.
*
* @note There are 2 slightly differently behaving modes.
* @li When automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
* in this case timestamps will be passed through with delays compensated
* @li When automatic timestamp compensation is used, (min_compensation < FLT_MAX)
* in this case the output timestamps will match output sample numbers.
* See ffmpeg-resampler(1) for the two modes of compensation.
*
* @param s[in] initialized Swr context
* @param pts[in] timestamp for the next input sample, INT64_MIN if unknown
* @see swr_set_compensation(), swr_drop_output(), and swr_inject_silence() are
* function used internally for timestamp compensation.
* @return the output timestamp for the next output sample
*/
int64_t swr_next_pts(struct SwrContext *s, int64_t pts);
/**
* @}
*
* @name Low-level option setting functions
* These functons provide a means to set low-level options that is not possible
* with the AVOption API.
* @{
*/
/**
* Activate resampling compensation ("soft" compensation). This function is
* internally called when needed in swr_next_pts().
*
* @param[in,out] s allocated Swr context. If it is not initialized,
* or SWR_FLAG_RESAMPLE is not set, swr_init() is
* called with the flag set.
* @param[in] sample_delta delta in PTS per sample
* @param[in] compensation_distance number of samples to compensate for
* @return >= 0 on success, AVERROR error codes if:
* @li @c s is NULL,
* @li @c compensation_distance is less than 0,
* @li @c compensation_distance is 0 but sample_delta is not,
* @li compensation unsupported by resampler, or
* @li swr_init() fails when called.
*/
int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance);
/**
* Set a customized input channel mapping.
*
* @param[in,out] s allocated Swr context, not yet initialized
* @param[in] channel_map customized input channel mapping (array of channel
* indexes, -1 for a muted channel)
* @return >= 0 on success, or AVERROR error code in case of failure.
*/
int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map);
/**
* Generate a channel mixing matrix.
*
* This function is the one used internally by libswresample for building the
* default mixing matrix. It is made public just as a utility function for
* building custom matrices.
*
* @param in_layout input channel layout
* @param out_layout output channel layout
* @param center_mix_level mix level for the center channel
* @param surround_mix_level mix level for the surround channel(s)
* @param lfe_mix_level mix level for the low-frequency effects channel
* @param rematrix_maxval if 1.0, coefficients will be normalized to prevent
* overflow. if INT_MAX, coefficients will not be
* normalized.
* @param[out] matrix mixing coefficients; matrix[i + stride * o] is
* the weight of input channel i in output channel o.
* @param stride distance between adjacent input channels in the
* matrix array
* @param matrix_encoding matrixed stereo downmix mode (e.g. dplii)
* @param log_ctx parent logging context, can be NULL
* @return 0 on success, negative AVERROR code on failure
*/
int swr_build_matrix(uint64_t in_layout, uint64_t out_layout,
double center_mix_level, double surround_mix_level,
double lfe_mix_level, double rematrix_maxval,
double rematrix_volume, double *matrix,
int stride, enum AVMatrixEncoding matrix_encoding,
void *log_ctx);
/**
* Set a customized remix matrix.
*
* @param s allocated Swr context, not yet initialized
* @param matrix remix coefficients; matrix[i + stride * o] is
* the weight of input channel i in output channel o
* @param stride offset between lines of the matrix
* @return >= 0 on success, or AVERROR error code in case of failure.
*/
int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride);
/**
* @}
*
* @name Sample handling functions
* @{
*/
/**
* Drops the specified number of output samples.
*
* This function, along with swr_inject_silence(), is called by swr_next_pts()
* if needed for "hard" compensation.
*
* @param s allocated Swr context
* @param count number of samples to be dropped
*
* @return >= 0 on success, or a negative AVERROR code on failure
*/
int swr_drop_output(struct SwrContext *s, int count);
/**
* Injects the specified number of silence samples.
*
* This function, along with swr_drop_output(), is called by swr_next_pts()
* if needed for "hard" compensation.
*
* @param s allocated Swr context
* @param count number of samples to be dropped
*
* @return >= 0 on success, or a negative AVERROR code on failure
*/
int swr_inject_silence(struct SwrContext *s, int count);
/**
* Gets the delay the next input sample will experience relative to the next output sample.
*
* Swresample can buffer data if more input has been provided than available
* output space, also converting between sample rates needs a delay.
* This function returns the sum of all such delays.
* The exact delay is not necessarily an integer value in either input or
* output sample rate. Especially when downsampling by a large value, the
* output sample rate may be a poor choice to represent the delay, similarly
* for upsampling and the input sample rate.
*
* @param s swr context
* @param base timebase in which the returned delay will be:
* @li if it's set to 1 the returned delay is in seconds
* @li if it's set to 1000 the returned delay is in milliseconds
* @li if it's set to the input sample rate then the returned
* delay is in input samples
* @li if it's set to the output sample rate then the returned
* delay is in output samples
* @li if it's the least common multiple of in_sample_rate and
* out_sample_rate then an exact rounding-free delay will be
* returned
* @returns the delay in 1 / @c base units.
*/
int64_t swr_get_delay(struct SwrContext *s, int64_t base);
/**
* Find an upper bound on the number of samples that the next swr_convert
* call will output, if called with in_samples of input samples. This
* depends on the internal state, and anything changing the internal state
* (like further swr_convert() calls) will may change the number of samples
* swr_get_out_samples() returns for the same number of input samples.
*
* @param in_samples number of input samples.
* @note any call to swr_inject_silence(), swr_convert(), swr_next_pts()
* or swr_set_compensation() invalidates this limit
* @note it is recommended to pass the correct available buffer size
* to all functions like swr_convert() even if swr_get_out_samples()
* indicates that less would be used.
* @returns an upper bound on the number of samples that the next swr_convert
* will output or a negative value to indicate an error
*/
int swr_get_out_samples(struct SwrContext *s, int in_samples);
/**
* @}
*
* @name Configuration accessors
* @{
*/
/**
* Return the @ref LIBSWRESAMPLE_VERSION_INT constant.
*
* This is useful to check if the build-time libswresample has the same version
* as the run-time one.
*
* @returns the unsigned int-typed version
*/
unsigned swresample_version(void);
/**
* Return the swr build-time configuration.
*
* @returns the build-time @c ./configure flags
*/
const char *swresample_configuration(void);
/**
* Return the swr license.
*
* @returns the license of libswresample, determined at build-time
*/
const char *swresample_license(void);
/**
* @}
*
* @name AVFrame based API
* @{
*/
/**
* Convert the samples in the input AVFrame and write them to the output AVFrame.
*
* Input and output AVFrames must have channel_layout, sample_rate and format set.
*
* If the output AVFrame does not have the data pointers allocated the nb_samples
* field will be set using av_frame_get_buffer()
* is called to allocate the frame.
*
* The output AVFrame can be NULL or have fewer allocated samples than required.
* In this case, any remaining samples not written to the output will be added
* to an internal FIFO buffer, to be returned at the next call to this function
* or to swr_convert().
*
* If converting sample rate, there may be data remaining in the internal
* resampling delay buffer. swr_get_delay() tells the number of
* remaining samples. To get this data as output, call this function or
* swr_convert() with NULL input.
*
* If the SwrContext configuration does not match the output and
* input AVFrame settings the conversion does not take place and depending on
* which AVFrame is not matching AVERROR_OUTPUT_CHANGED, AVERROR_INPUT_CHANGED
* or the result of a bitwise-OR of them is returned.
*
* @see swr_delay()
* @see swr_convert()
* @see swr_get_delay()
*
* @param swr audio resample context
* @param output output AVFrame
* @param input input AVFrame
* @return 0 on success, AVERROR on failure or nonmatching
* configuration.
*/
int swr_convert_frame(SwrContext *swr,
AVFrame *output, const AVFrame *input);
/**
* Configure or reconfigure the SwrContext using the information
* provided by the AVFrames.
*
* The original resampling context is reset even on failure.
* The function calls swr_close() internally if the context is open.
*
* @see swr_close();
*
* @param swr audio resample context
* @param output output AVFrame
* @param input input AVFrame
* @return 0 on success, AVERROR on failure.
*/
int swr_config_frame(SwrContext *swr, const AVFrame *out, const AVFrame *in);
/**
* @}
* @}
*/
#endif /* SWRESAMPLE_SWRESAMPLE_H */

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/*
* Version macros.
*
* This file is part of libswresample
*
* libswresample is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* libswresample is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with libswresample; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef SWRESAMPLE_VERSION_H
#define SWRESAMPLE_VERSION_H
/**
* @file
* Libswresample version macros
*/
#include "libavutil/avutil.h"
#define LIBSWRESAMPLE_VERSION_MAJOR 2
#define LIBSWRESAMPLE_VERSION_MINOR 8
#define LIBSWRESAMPLE_VERSION_MICRO 100
#define LIBSWRESAMPLE_VERSION_INT AV_VERSION_INT(LIBSWRESAMPLE_VERSION_MAJOR, \
LIBSWRESAMPLE_VERSION_MINOR, \
LIBSWRESAMPLE_VERSION_MICRO)
#define LIBSWRESAMPLE_VERSION AV_VERSION(LIBSWRESAMPLE_VERSION_MAJOR, \
LIBSWRESAMPLE_VERSION_MINOR, \
LIBSWRESAMPLE_VERSION_MICRO)
#define LIBSWRESAMPLE_BUILD LIBSWRESAMPLE_VERSION_INT
#define LIBSWRESAMPLE_IDENT "SwR" AV_STRINGIFY(LIBSWRESAMPLE_VERSION)
#endif /* SWRESAMPLE_VERSION_H */

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prefix=/usr/local
prefix=/Users/chris/Source/Repos/cog/ThirdParty/ffmpeg
exec_prefix=${prefix}
libdir=/usr/local/lib
includedir=/usr/local/include
libdir=/Users/chris/Source/Repos/cog/ThirdParty/ffmpeg/lib
includedir=/Users/chris/Source/Repos/cog/ThirdParty/ffmpeg/include
Name: libavformat
Description: FFmpeg container format library
Version: 57.82.100
Requires: libavcodec >= 57.105.100, libswresample >= 2.8.100, libavutil >= 55.74.100
Version: 57.82.101
Requires: libavcodec >= 57.106.101, libswresample >= 2.8.100, libavutil >= 55.76.100
Requires.private:
Conflicts:
Libs: -L${libdir} -lavformat -framework OpenGL -framework AppKit -framework OpenGL -framework AppKit -framework Foundation -framework CoreVideo -framework CoreMedia -L/usr/local/lib -lSDL2 -framework CoreFoundation -framework VideoToolbox -framework CoreMedia -framework CoreVideo -L/usr/local/lib -lSDL2 -liconv -Wl,-framework,CoreFoundation -Wl,-framework,Security -L/usr/local/lib -lSDL2 -lm -llzma -lbz2 -lz -pthread -pthread -framework CoreServices -framework CoreGraphics -framework VideoToolbox -framework CoreImage -framework AVFoundation -framework AudioToolbox
Libs: -L${libdir} -lavformat -framework CoreFoundation -framework VideoToolbox -framework CoreMedia -framework CoreVideo -liconv -Wl,-framework,CoreFoundation -Wl,-framework,Security -lm -lbz2 -lz -pthread -pthread -framework CoreServices -framework CoreGraphics -framework VideoToolbox -framework CoreImage -framework AVFoundation -framework AudioToolbox -framework AppKit
Libs.private:
Cflags: -I${includedir}

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Name: libavutil
Description: FFmpeg utility library
Version: 55.75.100
Version: 55.76.100
Requires:
Requires.private:
Conflicts:

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prefix=/usr/local
exec_prefix=${prefix}
libdir=/usr/local/lib
includedir=/usr/local/include
Name: libswresample
Description: FFmpeg audio resampling library
Version: 2.8.100
Requires: libavutil >= 55.74.100
Requires.private:
Conflicts:
Libs: -L${libdir} -lswresample -lm
Libs.private:
Cflags: -I${includedir}