Welcome to mirror list, hosted at ThFree Co, Russian Federation.

github.com/FFmpeg/FFmpeg.git - Unnamed repository; edit this file 'description' to name the repository.
summaryrefslogtreecommitdiff
diff options
context:
space:
mode:
authorDerek Buitenhuis <derek.buitenhuis@gmail.com>2016-04-10 22:58:15 +0300
committerDerek Buitenhuis <derek.buitenhuis@gmail.com>2016-04-10 22:59:55 +0300
commit6f69f7a8bf6a0d013985578df2ef42ee6b1c7994 (patch)
tree0c2ec8349ff1763d5f48454b8b9f26374dbd80b0 /libavformat/msf.c
parent60b75186b2c878b6257b43c8fcc0b1356ada218e (diff)
parent9200514ad8717c63f82101dc394f4378854325bf (diff)
Merge commit '9200514ad8717c63f82101dc394f4378854325bf'
* commit '9200514ad8717c63f82101dc394f4378854325bf': lavf: replace AVStream.codec with AVStream.codecpar This has been a HUGE effort from: - Derek Buitenhuis <derek.buitenhuis@gmail.com> - Hendrik Leppkes <h.leppkes@gmail.com> - wm4 <nfxjfg@googlemail.com> - Clément Bœsch <clement@stupeflix.com> - James Almer <jamrial@gmail.com> - Michael Niedermayer <michael@niedermayer.cc> - Rostislav Pehlivanov <atomnuker@gmail.com> Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Diffstat (limited to 'libavformat/msf.c')
-rw-r--r--libavformat/msf.c30
1 files changed, 15 insertions, 15 deletions
diff --git a/libavformat/msf.c b/libavformat/msf.c
index 97a6dc6929..0551e9bc24 100644
--- a/libavformat/msf.c
+++ b/libavformat/msf.c
@@ -51,41 +51,41 @@ static int msf_read_header(AVFormatContext *s)
if (!st)
return AVERROR(ENOMEM);
- st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+ st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
codec = avio_rb32(s->pb);
- st->codec->channels = avio_rb32(s->pb);
- if (st->codec->channels <= 0 || st->codec->channels >= INT_MAX / 1024)
+ st->codecpar->channels = avio_rb32(s->pb);
+ if (st->codecpar->channels <= 0 || st->codecpar->channels >= INT_MAX / 1024)
return AVERROR_INVALIDDATA;
size = avio_rb32(s->pb);
- st->codec->sample_rate = avio_rb32(s->pb);
- if (st->codec->sample_rate <= 0)
+ st->codecpar->sample_rate = avio_rb32(s->pb);
+ if (st->codecpar->sample_rate <= 0)
return AVERROR_INVALIDDATA;
align = avio_rb32(s->pb) ;
- if (align > INT_MAX / st->codec->channels)
+ if (align > INT_MAX / st->codecpar->channels)
return AVERROR_INVALIDDATA;
- st->codec->block_align = align;
+ st->codecpar->block_align = align;
switch (codec) {
- case 0: st->codec->codec_id = AV_CODEC_ID_PCM_S16BE; break;
- case 3: st->codec->block_align = 16 * st->codec->channels;
- st->codec->codec_id = AV_CODEC_ID_ADPCM_PSX; break;
+ case 0: st->codecpar->codec_id = AV_CODEC_ID_PCM_S16BE; break;
+ case 3: st->codecpar->block_align = 16 * st->codecpar->channels;
+ st->codecpar->codec_id = AV_CODEC_ID_ADPCM_PSX; break;
case 7: st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
- st->codec->codec_id = AV_CODEC_ID_MP3; break;
+ st->codecpar->codec_id = AV_CODEC_ID_MP3; break;
default:
avpriv_request_sample(s, "Codec %d", codec);
return AVERROR_PATCHWELCOME;
}
- st->duration = av_get_audio_frame_duration(st->codec, size);
+ st->duration = av_get_audio_frame_duration2(st->codecpar, size);
avio_skip(s->pb, 0x40 - avio_tell(s->pb));
- avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
+ avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
return 0;
}
static int msf_read_packet(AVFormatContext *s, AVPacket *pkt)
{
- AVCodecContext *codec = s->streams[0]->codec;
+ AVCodecParameters *par = s->streams[0]->codecpar;
- return av_get_packet(s->pb, pkt, codec->block_align ? codec->block_align : 1024 * codec->channels);
+ return av_get_packet(s->pb, pkt, par->block_align ? par->block_align : 1024 * par->channels);
}
AVInputFormat ff_msf_demuxer = {