diff options
| author | Luca Abeni <lucabe72@email.it> | 2008-09-08 14:24:59 +0000 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2008-09-08 14:24:59 +0000 |
| commit | dd1c8f3e6e5380f993c86750bb09fd42e130143f (patch) | |
| tree | 2ae5b2bbda1b685069f4db6ed408097288590dde /libavformat/bethsoftvid.c | |
| parent | 71375e05006e68fecdeb8d5fa80c3cce52a5cf86 (diff) | |
Bump Major version, this commit is almost just renaming bits_per_sample to
bits_per_coded_sample but that cannot be done seperately.
Patch by Luca Abeni
Also reset the minor version and fix the forgotton change to libfaad.
Note: The API/ABI should not be considered stable yet, there still may
be a change done here or there if some developer has some cleanup ideas and
patches!
Originally committed as revision 15262 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/bethsoftvid.c')
| -rw-r--r-- | libavformat/bethsoftvid.c | 6 |
1 files changed, 3 insertions, 3 deletions
diff --git a/libavformat/bethsoftvid.c b/libavformat/bethsoftvid.c index e3024c22b8..a80b105e2c 100644 --- a/libavformat/bethsoftvid.c +++ b/libavformat/bethsoftvid.c @@ -89,8 +89,8 @@ static int vid_read_header(AVFormatContext *s, stream->codec->codec_id = CODEC_ID_PCM_U8; stream->codec->channels = 1; stream->codec->sample_rate = 11025; - stream->codec->bits_per_sample = 8; - stream->codec->bit_rate = stream->codec->channels * stream->codec->sample_rate * stream->codec->bits_per_sample; + stream->codec->bits_per_coded_sample = 8; + stream->codec->bit_rate = stream->codec->channels * stream->codec->sample_rate * stream->codec->bits_per_coded_sample; return 0; } @@ -197,7 +197,7 @@ static int vid_read_packet(AVFormatContext *s, get_le16(pb); // soundblaster DAC used for sample rate, as on specification page (link above) s->streams[1]->codec->sample_rate = 1000000 / (256 - get_byte(pb)); - s->streams[1]->codec->bit_rate = s->streams[1]->codec->channels * s->streams[1]->codec->sample_rate * s->streams[1]->codec->bits_per_sample; + s->streams[1]->codec->bit_rate = s->streams[1]->codec->channels * s->streams[1]->codec->sample_rate * s->streams[1]->codec->bits_per_coded_sample; case AUDIO_BLOCK: audio_length = get_le16(pb); ret_value = av_get_packet(pb, pkt, audio_length); |