Merge remote-tracking branch 'qatar/master'

[ffmpeg] / libavformat / rtsp.c

diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index dcd50853101ded1c885c6f130902b741db738f1f..0582a00f8cec819d95c6246350caabcca57cf06f 100644 (file)
--- 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);