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authorMichael Niedermayer <michaelni@gmx.at>2013-01-12 13:14:00 +0100
committerMichael Niedermayer <michaelni@gmx.at>2013-01-12 13:14:05 +0100
commite730c3a2cbc4070f3e1799d2a6789f239f7377b1 (patch)
tree6e63062ae1b712ab9fa91fa35658a54b019bac91 /libavformat
parentdda2d2974863cb862f63fc1927d1d9ad5733555b (diff)
parent54cb096ee4558b3bfc28c2fcd6418ce82dc39fe1 (diff)
downloadffmpeg-e730c3a2cbc4070f3e1799d2a6789f239f7377b1.tar.gz
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* commit '54cb096ee4558b3bfc28c2fcd6418ce82dc39fe1': rtsp: Remove an outdated comment rtsp: Remove references to weirdly named variables in other files Merged-by: Michael Niedermayer 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>
Diffstat (limited to 'libavformat')
-rw-r--r--libavformat/rtsp.c8
1 files changed, 1 insertions, 7 deletions
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index dcd5085310..0582a00f8c 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -201,8 +201,7 @@ static int sdp_parse_rtpmap(AVFormatContext *s,
AVCodec *c;
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
* particular servers ("RealServer Version 6.1.3.970", see issue 1658)
* have a trailing space. */
@@ -210,7 +209,6 @@ static int sdp_parse_rtpmap(AVFormatContext *s,
if (payload_type < RTP_PT_PRIVATE) {
/* We are in a standard case
* (from http://www.iana.org/assignments/rtp-parameters). */
- /* search into AVRtpPayloadTypes[] */
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);
if (i > 0)
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",
codec->sample_rate);