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authorPhilip Gladstone <philipjsg@users.sourceforge.net>2002-05-16 01:57:55 +0000
committerPhilip Gladstone <philipjsg@users.sourceforge.net>2002-05-16 01:57:55 +0000
commit3884a3c322285f8795ca4cb64b1f41b9162811c6 (patch)
treeb344d5c316cc6ef5b4565a5717d2a2da6ab68d85 /libav/ffm.c
parent218ad65d4db44b629947737714a58760f14ef4f1 (diff)
downloadffmpeg-3884a3c322285f8795ca4cb64b1f41b9162811c6.tar.gz
Pass over the bit_rate_tolerance field to ffmpeg
Originally committed as revision 500 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libav/ffm.c')
-rw-r--r--libav/ffm.c2
1 files changed, 2 insertions, 0 deletions
diff --git a/libav/ffm.c b/libav/ffm.c
index 3f444c2815..26572cf555 100644
--- a/libav/ffm.c
+++ b/libav/ffm.c
@@ -166,6 +166,7 @@ static int ffm_write_header(AVFormatContext *s)
put_byte(pb, codec->max_qdiff);
put_be16(pb, (int) (codec->qcompress * 10000.0));
put_be16(pb, (int) (codec->qblur * 10000.0));
+ put_be32(pb, codec->bit_rate_tolerance);
break;
case CODEC_TYPE_AUDIO:
put_be32(pb, codec->sample_rate);
@@ -390,6 +391,7 @@ static int ffm_read_header(AVFormatContext *s, AVFormatParameters *ap)
codec->max_qdiff = get_byte(pb);
codec->qcompress = get_be16(pb) / 10000.0;
codec->qblur = get_be16(pb) / 10000.0;
+ codec->bit_rate_tolerance = get_be32(pb);
break;
case CODEC_TYPE_AUDIO:
codec->sample_rate = get_be32(pb);