Mercurial > libavformat.hg
changeset 3201:1a16b069daca libavformat
use common aac sample rate tables
| author | aurel |
|---|---|
| date | Wed, 02 Apr 2008 21:41:48 +0000 |
| parents | a437fb5ca080 |
| children | 98e633b72d38 |
| files | matroskadec.c matroskaenc.c |
| diffstat | 2 files changed, 6 insertions(+), 12 deletions(-) [+] |
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--- a/matroskadec.c Wed Apr 02 19:03:59 2008 +0000 +++ b/matroskadec.c Wed Apr 02 21:41:48 2008 +0000 @@ -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 @@ 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; }
--- a/matroskaenc.c Wed Apr 02 19:03:59 2008 +0000 +++ b/matroskaenc.c Wed Apr 02 21:41:48 2008 +0000 @@ -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 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 @@ 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 @@ 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]; } }