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/westwood.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/westwood.c')
| -rw-r--r-- | libavformat/westwood.c | 12 |
1 files changed, 6 insertions, 6 deletions
diff --git a/libavformat/westwood.c b/libavformat/westwood.c index d1ac63c762..600863e837 100644 --- a/libavformat/westwood.c +++ b/libavformat/westwood.c @@ -154,10 +154,10 @@ static int wsaud_read_header(AVFormatContext *s, st->codec->codec_tag = 0; /* no tag */ st->codec->channels = wsaud->audio_channels; st->codec->sample_rate = wsaud->audio_samplerate; - st->codec->bits_per_sample = wsaud->audio_bits; + st->codec->bits_per_coded_sample = wsaud->audio_bits; st->codec->bit_rate = st->codec->channels * st->codec->sample_rate * - st->codec->bits_per_sample / 4; - st->codec->block_align = st->codec->channels * st->codec->bits_per_sample; + st->codec->bits_per_coded_sample / 4; + st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample; wsaud->audio_stream_index = st->index; wsaud->audio_frame_counter = 0; @@ -264,10 +264,10 @@ static int wsvqa_read_header(AVFormatContext *s, st->codec->channels = header[26]; if (!st->codec->channels) st->codec->channels = 1; - st->codec->bits_per_sample = 16; + st->codec->bits_per_coded_sample = 16; st->codec->bit_rate = st->codec->channels * st->codec->sample_rate * - st->codec->bits_per_sample / 4; - st->codec->block_align = st->codec->channels * st->codec->bits_per_sample; + st->codec->bits_per_coded_sample / 4; + st->codec->block_align = st->codec->channels * st->codec->bits_per_coded_sample; wsvqa->audio_stream_index = st->index; wsvqa->audio_samplerate = st->codec->sample_rate; |