Merge remote-tracking branch 'qatar/master'

[ffmpeg] / libavformat / rtsp.c

diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index ec343d6070a5e94a9ad8a6df413c7374bf4bb9d9..0582a00f8cec819d95c6246350caabcca57cf06f 100644 (file)
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -2,23 +2,24 @@
  * RTSP/SDP client
  * Copyright (c) 2002 Fabrice Bellard
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
+#include "libavutil/avassert.h"
 #include "libavutil/base64.h"
 #include "libavutil/avstring.h"
 #include "libavutil/intreadwrite.h"
@@ -243,10 +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);
@@ -629,7 +626,7 @@ int ff_rtsp_open_transport_ctx(AVFormatContext *s, RTSPStream *rtsp_st)
         s->ctx_flags |= AVFMTCTX_NOHEADER;
 
     if (s->oformat && CONFIG_RTSP_MUXER) {
-        int ret = ff_rtp_chain_mux_open(&rtsp_st->transport_priv, s, st,
+        int ret = ff_rtp_chain_mux_open((AVFormatContext **)&rtsp_st->transport_priv, s, st,
                                         rtsp_st->rtp_handle,
                                         RTSP_TCP_MAX_PACKET_SIZE,
                                         rtsp_st->stream_index);
@@ -1227,10 +1224,6 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
     /* default timeout: 1 minute */
     rt->timeout = 60;
 
-    /* for each stream, make the setup request */
-    /* XXX: we assume the same server is used for the control of each
-     * RTSP stream */
-
     /* Choose a random starting offset within the first half of the
      * port range, to allow for a number of ports to try even if the offset
      * happens to be at the end of the random range. */
@@ -1284,7 +1277,6 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const char *host, int port,
                                &s->interrupt_callback, NULL))
                     goto rtp_opened;
             }
-
             av_log(s, AV_LOG_ERROR, "Unable to open an input RTP port\n");
             err = AVERROR(EIO);
             goto fail;