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https://github.com/FFmpeg/FFmpeg.git
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use common aac sample rate tables
Originally committed as revision 12671 to svn://svn.ffmpeg.org/ffmpeg/trunk
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@ -306,8 +306,9 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER) += adpcm.o
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# libavformat dependencies
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# libavformat dependencies
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OBJS-$(CONFIG_GXF_DEMUXER) += mpeg12data.o
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OBJS-$(CONFIG_GXF_DEMUXER) += mpeg12data.o
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OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER) += xiph.o
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OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER) += xiph.o mpeg4audio.o
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OBJS-$(CONFIG_MATROSKA_MUXER) += xiph.o
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OBJS-$(CONFIG_MATROSKA_DEMUXER) += mpeg4audio.o
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OBJS-$(CONFIG_MATROSKA_MUXER) += xiph.o mpeg4audio.o
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OBJS-$(CONFIG_NUT_MUXER) += mpegaudiodata.o
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OBJS-$(CONFIG_NUT_MUXER) += mpegaudiodata.o
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OBJS-$(CONFIG_OGG_MUXER) += xiph.o
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OBJS-$(CONFIG_OGG_MUXER) += xiph.o
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OBJS-$(CONFIG_RTP_MUXER) += mpegvideo.o
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OBJS-$(CONFIG_RTP_MUXER) += mpegvideo.o
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@ -33,6 +33,7 @@
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#include "riff.h"
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#include "riff.h"
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#include "intfloat_readwrite.h"
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#include "intfloat_readwrite.h"
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#include "matroska.h"
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#include "matroska.h"
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#include "libavcodec/mpeg4audio.h"
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typedef struct Track {
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typedef struct Track {
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MatroskaTrackType type;
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MatroskaTrackType type;
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@ -1997,14 +1998,10 @@ matroska_aac_profile (char *codec_id)
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static int
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static int
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matroska_aac_sri (int samplerate)
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matroska_aac_sri (int samplerate)
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{
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{
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static const int aac_sample_rates[] = {
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96000, 88200, 64000, 48000, 44100, 32000,
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24000, 22050, 16000, 12000, 11025, 8000,
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};
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int sri;
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int sri;
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for (sri=0; sri<ARRAY_SIZE(aac_sample_rates); sri++)
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for (sri=0; sri<ARRAY_SIZE(ff_mpeg4audio_sample_rates); sri++)
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if (aac_sample_rates[sri] == samplerate)
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if (ff_mpeg4audio_sample_rates[sri] == samplerate)
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break;
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break;
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return sri;
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return sri;
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}
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}
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@ -25,6 +25,7 @@
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#include "xiph.h"
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#include "xiph.h"
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#include "matroska.h"
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#include "matroska.h"
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#include "avc.h"
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#include "avc.h"
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#include "libavcodec/mpeg4audio.h"
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typedef struct ebml_master {
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typedef struct ebml_master {
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offset_t pos; ///< absolute offset in the file where the master's elements start
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offset_t pos; ///< absolute offset in the file where the master's elements start
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@ -438,10 +439,6 @@ static int put_flac_codecpriv(AVFormatContext *s, ByteIOContext *pb, AVCodecCont
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static void get_aac_sample_rates(AVFormatContext *s, AVCodecContext *codec, int *sample_rate, int *output_sample_rate)
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static void get_aac_sample_rates(AVFormatContext *s, AVCodecContext *codec, int *sample_rate, int *output_sample_rate)
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{
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{
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static const int aac_sample_rates[] = {
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96000, 88200, 64000, 48000, 44100, 32000,
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24000, 22050, 16000, 12000, 11025, 8000,
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};
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int sri;
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int sri;
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if (codec->extradata_size < 2) {
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if (codec->extradata_size < 2) {
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@ -454,7 +451,7 @@ static void get_aac_sample_rates(AVFormatContext *s, AVCodecContext *codec, int
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av_log(s, AV_LOG_WARNING, "AAC samplerate index out of bounds\n");
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av_log(s, AV_LOG_WARNING, "AAC samplerate index out of bounds\n");
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return;
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return;
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}
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}
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*sample_rate = aac_sample_rates[sri];
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*sample_rate = ff_mpeg4audio_sample_rates[sri];
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// if sbr, get output sample rate as well
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// if sbr, get output sample rate as well
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if (codec->extradata_size == 5) {
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if (codec->extradata_size == 5) {
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@ -463,7 +460,7 @@ static void get_aac_sample_rates(AVFormatContext *s, AVCodecContext *codec, int
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av_log(s, AV_LOG_WARNING, "AAC output samplerate index out of bounds\n");
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av_log(s, AV_LOG_WARNING, "AAC output samplerate index out of bounds\n");
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return;
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return;
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}
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}
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*output_sample_rate = aac_sample_rates[sri];
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*output_sample_rate = ff_mpeg4audio_sample_rates[sri];
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}
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}
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}
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}
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