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use common aac sample rate tables

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Originally committed as revision 12671 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Aurelien Jacobs 2008-04-02 21:41:48 +00:00
parent b2fec0fc25
commit 7bfacd4e75
3 changed files with 9 additions and 14 deletions
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5
libavcodec/Makefile
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@ -306,8 +306,9 @@ OBJS-$(CONFIG_ADPCM_YAMAHA_ENCODER) += adpcm.o
# libavformat dependencies
OBJS-$(CONFIG_GXF_DEMUXER) += mpeg12data.o
OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER) += xiph.o
OBJS-$(CONFIG_MATROSKA_MUXER) += xiph.o
OBJS-$(CONFIG_MATROSKA_AUDIO_MUXER) += xiph.o mpeg4audio.o
OBJS-$(CONFIG_MATROSKA_DEMUXER) += mpeg4audio.o
OBJS-$(CONFIG_MATROSKA_MUXER) += xiph.o mpeg4audio.o
OBJS-$(CONFIG_NUT_MUXER) += mpegaudiodata.o
OBJS-$(CONFIG_OGG_MUXER) += xiph.o
OBJS-$(CONFIG_RTP_MUXER) += mpegvideo.o

9
libavformat/matroskadec.c
View File

@ -33,6 +33,7 @@
#include "riff.h"
#include "intfloat_readwrite.h"
#include "matroska.h"
#include "libavcodec/mpeg4audio.h"
typedef struct Track {
MatroskaTrackType type;
@ -1997,14 +1998,10 @@ matroska_aac_profile (char *codec_id)
static int
matroska_aac_sri (int samplerate)
{
static const int aac_sample_rates[] = {
96000, 88200, 64000, 48000, 44100, 32000,
24000, 22050, 16000, 12000, 11025, 8000,
};
int sri;
for (sri=0; sri<ARRAY_SIZE(aac_sample_rates); sri++)
if (aac_sample_rates[sri] == samplerate)
for (sri=0; sri<ARRAY_SIZE(ff_mpeg4audio_sample_rates); sri++)
if (ff_mpeg4audio_sample_rates[sri] == samplerate)
break;
return sri;
}

9
libavformat/matroskaenc.c
View File

@ -25,6 +25,7 @@
#include "xiph.h"
#include "matroska.h"
#include "avc.h"
#include "libavcodec/mpeg4audio.h"
typedef struct ebml_master {
offset_t pos; ///< absolute offset in the file where the master's elements start
@ -438,10 +439,6 @@ static int put_flac_codecpriv(AVFormatContext *s, ByteIOContext *pb, AVCodecCont
static void get_aac_sample_rates(AVFormatContext *s, AVCodecContext *codec, int *sample_rate, int *output_sample_rate)
{
static const int aac_sample_rates[] = {
96000, 88200, 64000, 48000, 44100, 32000,
24000, 22050, 16000, 12000, 11025, 8000,
};
int sri;
if (codec->extradata_size < 2) {
@ -454,7 +451,7 @@ static void get_aac_sample_rates(AVFormatContext *s, AVCodecContext *codec, int
av_log(s, AV_LOG_WARNING, "AAC samplerate index out of bounds\n");
return;
}
*sample_rate = aac_sample_rates[sri];
*sample_rate = ff_mpeg4audio_sample_rates[sri];
// if sbr, get output sample rate as well
if (codec->extradata_size == 5) {
@ -463,7 +460,7 @@ static void get_aac_sample_rates(AVFormatContext *s, AVCodecContext *codec, int
av_log(s, AV_LOG_WARNING, "AAC output samplerate index out of bounds\n");
return;
}
*output_sample_rate = aac_sample_rates[sri];
*output_sample_rate = ff_mpeg4audio_sample_rates[sri];
}
}