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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/rtp.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/rtp.c')
| -rw-r--r-- | libavformat/rtp.c | 28 |
1 files changed, 14 insertions, 14 deletions
diff --git a/libavformat/rtp.c b/libavformat/rtp.c index 4d41350f3c..4745e54bb0 100644 --- a/libavformat/rtp.c +++ b/libavformat/rtp.c @@ -68,19 +68,19 @@ static const struct { {-1, "", AVMEDIA_TYPE_UNKNOWN, AV_CODEC_ID_NONE, -1, -1} }; -int ff_rtp_get_codec_info(AVCodecContext *codec, int payload_type) +int ff_rtp_get_codec_info(AVCodecParameters *par, int payload_type) { int i = 0; for (i = 0; rtp_payload_types[i].pt >= 0; i++) if (rtp_payload_types[i].pt == payload_type) { if (rtp_payload_types[i].codec_id != AV_CODEC_ID_NONE) { - codec->codec_type = rtp_payload_types[i].codec_type; - codec->codec_id = rtp_payload_types[i].codec_id; + par->codec_type = rtp_payload_types[i].codec_type; + par->codec_id = rtp_payload_types[i].codec_id; if (rtp_payload_types[i].audio_channels > 0) - codec->channels = rtp_payload_types[i].audio_channels; + par->channels = rtp_payload_types[i].audio_channels; if (rtp_payload_types[i].clock_rate > 0) - codec->sample_rate = rtp_payload_types[i].clock_rate; + par->sample_rate = rtp_payload_types[i].clock_rate; return 0; } } @@ -88,7 +88,7 @@ int ff_rtp_get_codec_info(AVCodecContext *codec, int payload_type) } int ff_rtp_get_payload_type(AVFormatContext *fmt, - AVCodecContext *codec, int idx) + AVCodecParameters *par, int idx) { int i; AVOutputFormat *ofmt = fmt ? fmt->oformat : NULL; @@ -103,27 +103,27 @@ int ff_rtp_get_payload_type(AVFormatContext *fmt, /* static payload type */ for (i = 0; rtp_payload_types[i].pt >= 0; ++i) - if (rtp_payload_types[i].codec_id == codec->codec_id) { - if (codec->codec_id == AV_CODEC_ID_H263 && (!fmt || !fmt->oformat || + if (rtp_payload_types[i].codec_id == par->codec_id) { + if (par->codec_id == AV_CODEC_ID_H263 && (!fmt || !fmt->oformat || !fmt->oformat->priv_class || !fmt->priv_data || !av_opt_flag_is_set(fmt->priv_data, "rtpflags", "rfc2190"))) continue; /* G722 has 8000 as nominal rate even if the sample rate is 16000, * see section 4.5.2 in RFC 3551. */ - if (codec->codec_id == AV_CODEC_ID_ADPCM_G722 && - codec->sample_rate == 16000 && codec->channels == 1) + if (par->codec_id == AV_CODEC_ID_ADPCM_G722 && + par->sample_rate == 16000 && par->channels == 1) return rtp_payload_types[i].pt; - if (codec->codec_type == AVMEDIA_TYPE_AUDIO && + if (par->codec_type == AVMEDIA_TYPE_AUDIO && ((rtp_payload_types[i].clock_rate > 0 && - codec->sample_rate != rtp_payload_types[i].clock_rate) || + par->sample_rate != rtp_payload_types[i].clock_rate) || (rtp_payload_types[i].audio_channels > 0 && - codec->channels != rtp_payload_types[i].audio_channels))) + par->channels != rtp_payload_types[i].audio_channels))) continue; return rtp_payload_types[i].pt; } if (idx < 0) - idx = codec->codec_type == AVMEDIA_TYPE_AUDIO; + idx = par->codec_type == AVMEDIA_TYPE_AUDIO; /* dynamic payload type */ return RTP_PT_PRIVATE + idx; |