Merge remote-tracking branch 'qatar/master'

* qatar/master: (44 commits)
  replacement Indeo 3 decoder
  gsm demuxer: do not allocate packet twice.
  flvenc: use first packet delay as global delay.
  ac3enc: doxygen update.
  imc: return error codes instead of 0 for error conditions.
  imc: return meaningful error codes instead of -1
  imc: do not set channel layout for stereo
  imc: validate channel count
  imc: check for ff_fft_init() failure
  imc: check output buffer size before decoding
  imc: use DSPContext.bswap16_buf() to byte-swap packet data
  rtsp: add allowed_media_types option
  libgsm: add flush function to reset the decoder state when seeking
  libgsm: simplify decoding by using a loop
  gsm: log error message when packet is too small
  libgsmdec: do not needlessly set *data_size to 0
  gsmdec: do not needlessly set *data_size to 0
  gsmdec: add flush function to reset the decoder state when seeking
  libgsmdec: check output buffer size before decoding
  gsmdec: log error message when output buffer is too small.
  ...

Conflicts:
	Changelog
	ffplay.c
	libavcodec/indeo3.c
	libavcodec/mjpeg_parser.c
	libavcodec/vp3.c
	libavformat/cutils.c
	libavformat/id3v2.c
	libavutil/parseutils.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 11 insertions, 1 deletions

diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 11533968eb..a80a9da7bf 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -65,6 +65,12 @@
     { name, longname, OFFSET(rtsp_flags), AV_OPT_TYPE_FLAGS, {0}, INT_MIN, INT_MAX, DEC, "rtsp_flags" }, \
     { "filter_src", "Only receive packets from the negotiated peer IP", 0, AV_OPT_TYPE_CONST, {RTSP_FLAG_FILTER_SRC}, 0, 0, DEC, "rtsp_flags" }
 
+#define RTSP_MEDIATYPE_OPTS(name, longname) \
+    { name, longname, OFFSET(media_type_mask), AV_OPT_TYPE_FLAGS, { (1 << (AVMEDIA_TYPE_DATA+1)) - 1 }, INT_MIN, INT_MAX, DEC, "allowed_media_types" }, \
+    { "video", "Video", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_VIDEO}, 0, 0, DEC, "allowed_media_types" }, \
+    { "audio", "Audio", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_AUDIO}, 0, 0, DEC, "allowed_media_types" }, \
+    { "data", "Data", 0, AV_OPT_TYPE_CONST, {1 << AVMEDIA_TYPE_DATA}, 0, 0, DEC, "allowed_media_types" }
+
 const AVOption ff_rtsp_options[] = {
     { "initial_pause",  "Don't start playing the stream immediately", OFFSET(initial_pause), AV_OPT_TYPE_INT, {0}, 0, 1, DEC },
     FF_RTP_FLAG_OPTS(RTSPState, rtp_muxer_flags),
@@ -74,11 +80,13 @@ const AVOption ff_rtsp_options[] = {
     { "udp_multicast", "UDP multicast", 0, AV_OPT_TYPE_CONST, {1 << RTSP_LOWER_TRANSPORT_UDP_MULTICAST}, 0, 0, DEC, "rtsp_transport" },
     { "http", "HTTP tunneling", 0, AV_OPT_TYPE_CONST, {(1 << RTSP_LOWER_TRANSPORT_HTTP)}, 0, 0, DEC, "rtsp_transport" },
     RTSP_FLAG_OPTS("rtsp_flags", "RTSP flags"),
+    RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
     { NULL },
 };
 
 static const AVOption sdp_options[] = {
     RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
+    RTSP_MEDIATYPE_OPTS("allowed_media_types", "Media types to accept from the server"),
     { NULL },
 };
 
@@ -325,6 +333,7 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
     case 'm':
         /* new stream */
         s1->skip_media = 0;
+        codec_type = AVMEDIA_TYPE_UNKNOWN;
         get_word(st_type, sizeof(st_type), &p);
         if (!strcmp(st_type, "audio")) {
             codec_type = AVMEDIA_TYPE_AUDIO;
@@ -332,7 +341,8 @@ static void sdp_parse_line(AVFormatContext *s, SDPParseState *s1,
             codec_type = AVMEDIA_TYPE_VIDEO;
         } else if (!strcmp(st_type, "application")) {
             codec_type = AVMEDIA_TYPE_DATA;
-        } else {
+        }
+        if (codec_type == AVMEDIA_TYPE_UNKNOWN || !(rt->media_type_mask & (1 << codec_type))) {
             s1->skip_media = 1;
             return;
         }