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Merge commit '54cb096ee4558b3bfc28c2fcd6418ce82dc39fe1' 

* commit '54cb096ee4558b3bfc28c2fcd6418ce82dc39fe1':
  rtsp: Remove an outdated comment
  rtsp: Remove references to weirdly named variables in other files

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2013-01-12 13:14:00 +01:00
parent dda2d29748 54cb096ee4
commit e730c3a2cb
1 changed files with 1 additions and 7 deletions
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8
libavformat/rtsp.c
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@ -201,8 +201,7 @@ static int sdp_parse_rtpmap(AVFormatContext *s,
AVCodec *c; AVCodec *c;
const char *c_name; const char *c_name;
/* Loop into AVRtpDynamicPayloadTypes[] and AVRtpPayloadTypes[] and /* See if we can handle this kind of payload.
* see if we can handle this kind of payload.
* The space should normally not be there but some Real streams or * The space should normally not be there but some Real streams or
* particular servers ("RealServer Version 6.1.3.970", see issue 1658) * particular servers ("RealServer Version 6.1.3.970", see issue 1658)
* have a trailing space. */ * have a trailing space. */
@ -210,7 +209,6 @@ static int sdp_parse_rtpmap(AVFormatContext *s,
if (payload_type < RTP_PT_PRIVATE) { if (payload_type < RTP_PT_PRIVATE) {
/* We are in a standard case /* We are in a standard case
* (from http://www.iana.org/assignments/rtp-parameters). */ * (from http://www.iana.org/assignments/rtp-parameters). */
/* search into AVRtpPayloadTypes[] */
codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type); codec->codec_id = ff_rtp_codec_id(buf, codec->codec_type);
} }
@ -246,10 +244,6 @@ static int sdp_parse_rtpmap(AVFormatContext *s,
i = atoi(buf); i = atoi(buf);
if (i > 0) if (i > 0)
codec->channels = i; codec->channels = i;
// TODO: there is a bug here; if it is a mono stream, and
// less than 22000Hz, faad upconverts to stereo and twice
// the frequency. No problem, but the sample rate is being
// set here by the sdp line. Patch on its way. (rdm)
} }
av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n", av_log(s, AV_LOG_DEBUG, "audio samplerate set to: %i\n",
codec->sample_rate); codec->sample_rate);