[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



More information about the pkg-multimedia-commits mailing list