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| author | Michael Niedermayer <michaelni@gmx.at> | 2012-07-24 20:43:07 +0200 |
|---|---|---|
| committer | Michael Niedermayer <michaelni@gmx.at> | 2012-07-24 21:15:57 +0200 |
| commit | 93342de1d88081822dad55af70dd4738cd15d242 (patch) | |
| tree | 7dd3f6fe644743880f7be2554570bda7497688c7 /libavformat/rtp.c | |
| parent | 3ccf22c64a0065fa08fe642500f193394fd83d01 (diff) | |
| parent | 6a433fdba82ff241be2e9193f66a43689766e4d7 (diff) | |
| download | ffmpeg-streaming-93342de1d88081822dad55af70dd4738cd15d242.zip ffmpeg-streaming-93342de1d88081822dad55af70dd4738cd15d242.tar.gz | |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
rtmp: Add credit/copyright to librtmp authors for parts of the RTMPE code
rtmp: Move the CONFIG_ condition into the if conditions
aac: Mention abbreviation as well in long_name
build: Skip compiling rtmpdh.h if ffrtmpcrypt protocol is not enabled
doc: Add Git configuration section
configure: Add a dependency on https for rtmpts
rtp: Only choose static payload types if the sample rate and channels are right
Conflicts:
doc/git-howto.texi
libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavformat/rtp.c')
| -rw-r--r-- | libavformat/rtp.c | 14 |
1 files changed, 11 insertions, 3 deletions
diff --git a/libavformat/rtp.c b/libavformat/rtp.c index a5484ae..2a80162 100644 --- a/libavformat/rtp.c +++ b/libavformat/rtp.c @@ -110,9 +110,17 @@ int ff_rtp_get_payload_type(AVFormatContext *fmt, AVCodecContext *codec) !fmt->oformat->priv_class || !av_opt_flag_is_set(fmt->priv_data, "rtpflags", "rfc2190"))) continue; - if (codec->codec_id == CODEC_ID_PCM_S16BE) - if (codec->channels != AVRtpPayloadTypes[i].audio_channels) - 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 == CODEC_ID_ADPCM_G722 && + codec->sample_rate == 16000 && codec->channels == 1) + return AVRtpPayloadTypes[i].pt; + if (codec->codec_type == AVMEDIA_TYPE_AUDIO && + ((AVRtpPayloadTypes[i].clock_rate > 0 && + codec->sample_rate != AVRtpPayloadTypes[i].clock_rate) || + (AVRtpPayloadTypes[i].audio_channels > 0 && + codec->channels != AVRtpPayloadTypes[i].audio_channels))) + continue; return AVRtpPayloadTypes[i].pt; } |
