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| author | Philip Gladstone <philipjsg@users.sourceforge.net> | 2002-05-16 05:57:55 +0400 |
|---|---|---|
| committer | Philip Gladstone <philipjsg@users.sourceforge.net> | 2002-05-16 05:57:55 +0400 |
| commit | 3884a3c322285f8795ca4cb64b1f41b9162811c6 (patch) | |
| tree | b344d5c316cc6ef5b4565a5717d2a2da6ab68d85 | |
| parent | 218ad65d4db44b629947737714a58760f14ef4f1 (diff) | |
Pass over the bit_rate_tolerance field to ffmpeg
Originally committed as revision 500 to svn://svn.ffmpeg.org/ffmpeg/trunk
| -rw-r--r-- | libav/ffm.c | 2 |
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); |