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| author | Derek Buitenhuis <derek.buitenhuis@gmail.com> | 2016-04-10 20:58:15 +0100 |
|---|---|---|
| committer | Derek Buitenhuis <derek.buitenhuis@gmail.com> | 2016-04-10 20:59:55 +0100 |
| commit | 6f69f7a8bf6a0d013985578df2ef42ee6b1c7994 (patch) | |
| tree | 0c2ec8349ff1763d5f48454b8b9f26374dbd80b0 /libavformat/voc_packet.c | |
| parent | 60b75186b2c878b6257b43c8fcc0b1356ada218e (diff) | |
| parent | 9200514ad8717c63f82101dc394f4378854325bf (diff) | |
| download | ffmpeg-6f69f7a8bf6a0d013985578df2ef42ee6b1c7994.tar.gz | |
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/voc_packet.c')
| -rw-r--r-- | libavformat/voc_packet.c | 36 |
1 files changed, 18 insertions, 18 deletions
diff --git a/libavformat/voc_packet.c b/libavformat/voc_packet.c index f4b82312f7..5833a79b56 100644 --- a/libavformat/voc_packet.c +++ b/libavformat/voc_packet.c @@ -27,7 +27,7 @@ int ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size) { VocDecContext *voc = s->priv_data; - AVCodecContext *dec = st->codec; + AVCodecParameters *par = st->codecpar; AVIOContext *pb = s->pb; VocType type; int size, tmp_codec=-1; @@ -57,13 +57,13 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size) switch (type) { case VOC_TYPE_VOICE_DATA: - if (!dec->sample_rate) { - dec->sample_rate = 1000000 / (256 - avio_r8(pb)); + if (!par->sample_rate) { + par->sample_rate = 1000000 / (256 - avio_r8(pb)); if (sample_rate) - dec->sample_rate = sample_rate; - avpriv_set_pts_info(st, 64, 1, dec->sample_rate); - dec->channels = channels; - dec->bits_per_coded_sample = av_get_bits_per_sample(dec->codec_id); + par->sample_rate = sample_rate; + avpriv_set_pts_info(st, 64, 1, par->sample_rate); + par->channels = channels; + par->bits_per_coded_sample = av_get_bits_per_sample(par->codec_id); } else avio_skip(pb, 1); tmp_codec = avio_r8(pb); @@ -85,11 +85,11 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size) break; case VOC_TYPE_NEW_VOICE_DATA: - if (!dec->sample_rate) { - dec->sample_rate = avio_rl32(pb); - avpriv_set_pts_info(st, 64, 1, dec->sample_rate); - dec->bits_per_coded_sample = avio_r8(pb); - dec->channels = avio_r8(pb); + if (!par->sample_rate) { + par->sample_rate = avio_rl32(pb); + avpriv_set_pts_info(st, 64, 1, par->sample_rate); + par->bits_per_coded_sample = avio_r8(pb); + par->channels = avio_r8(pb); } else avio_skip(pb, 6); tmp_codec = avio_rl16(pb); @@ -108,11 +108,11 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size) if (tmp_codec >= 0) { tmp_codec = ff_codec_get_id(ff_voc_codec_tags, tmp_codec); - if (dec->codec_id == AV_CODEC_ID_NONE) - dec->codec_id = tmp_codec; - else if (dec->codec_id != tmp_codec) + if (par->codec_id == AV_CODEC_ID_NONE) + par->codec_id = tmp_codec; + else if (par->codec_id != tmp_codec) av_log(s, AV_LOG_WARNING, "Ignoring mid-stream change in audio codec\n"); - if (dec->codec_id == AV_CODEC_ID_NONE) { + if (par->codec_id == AV_CODEC_ID_NONE) { if (s->audio_codec_id == AV_CODEC_ID_NONE) { av_log(s, AV_LOG_ERROR, "unknown codec tag\n"); return AVERROR(EINVAL); @@ -121,7 +121,7 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size) } } - dec->bit_rate = dec->sample_rate * dec->channels * dec->bits_per_coded_sample; + par->bit_rate = par->sample_rate * par->channels * par->bits_per_coded_sample; if (max_size <= 0) max_size = 2048; @@ -131,7 +131,7 @@ ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size) ret = av_get_packet(pb, pkt, size); pkt->dts = pkt->pts = voc->pts; - duration = av_get_audio_frame_duration(st->codec, size); + duration = av_get_audio_frame_duration2(st->codecpar, size); if (duration > 0 && voc->pts != AV_NOPTS_VALUE) voc->pts += duration; else |