[SCM] libav/experimental: Add audio codec 0x1602 (AAC LATM)
siretart at users.alioth.debian.org
siretart at users.alioth.debian.org
Sun Jun 30 17:21:37 UTC 2013
The following commit has been merged in the experimental branch:
commit 50d83b20058fd9564c0c41675717d2a87c36409b
Author: Peter Ross <pross at xvid.org>
Date: Sun Jan 9 02:11:41 2011 +0000
Add audio codec 0x1602 (AAC LATM)
Originally committed as revision 26273 to svn://svn.ffmpeg.org/ffmpeg/trunk
diff --git a/libavformat/riff.c b/libavformat/riff.c
index 7388f9c..4edccce 100644
--- a/libavformat/riff.c
+++ b/libavformat/riff.c
@@ -298,6 +298,7 @@ const AVCodecTag ff_codec_wav_tags[] = {
{ CODEC_ID_IMC, 0x0401 },
{ CODEC_ID_GSM_MS, 0x1500 },
{ CODEC_ID_TRUESPEECH, 0x1501 },
+ { CODEC_ID_AAC_LATM, 0x1602 },
{ CODEC_ID_AC3, 0x2000 },
{ CODEC_ID_DTS, 0x2001 },
{ CODEC_ID_SONIC, 0x2048 },
@@ -515,6 +516,11 @@ void ff_get_wav_header(ByteIOContext *pb, AVCodecContext *codec, int size)
url_fskip(pb, size);
}
codec->codec_id = ff_wav_codec_get_id(id, codec->bits_per_coded_sample);
+ if (codec->codec_id == CODEC_ID_AAC_LATM) {
+ /* channels and sample_rate values are those prior to applying SBR and/or PS */
+ codec->channels = 0;
+ codec->sample_rate = 0;
+ }
}
--
Libav/FFmpeg packaging
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