This avoids exposing a dummy AVStream which won't get any data
and which will make avformat_find_stream_info wait for info about
this stream.
---
libavformat/rtpdec.c | 2 +-
libavformat/rtpdec_asf.c | 2 ++
libavformat/rtsp.c | 22 +++++++++++++++-------
3 files changed, 18 insertions(+), 8 deletions(-)
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index 61653f7..41e6eb4 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -385,7 +385,7 @@ RTPDemuxContext *ff_rtp_parse_open(AVFormatContext *s1,
AVStream *st, URLContext
av_free(s);
return NULL;
}
- } else {
+ } else if (st) {
switch(st->codec->codec_id) {
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
diff --git a/libavformat/rtpdec_asf.c b/libavformat/rtpdec_asf.c
index c1690ef..bbb7609 100644
--- a/libavformat/rtpdec_asf.c
+++ b/libavformat/rtpdec_asf.c
@@ -130,6 +130,8 @@ int ff_wms_parse_sdp_a_line(AVFormatContext *s, const char
*p)
static int asfrtp_parse_sdp_line(AVFormatContext *s, int stream_index,
PayloadContext *asf, const char *line)
{
+ if (stream_index < 0)
+ return 0;
if (av_strstart(line, "stream:", &line)) {
RTSPState *rt = s->priv_data;
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index c94ef02..902a28e 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -374,6 +374,10 @@ static void sdp_parse_line(AVFormatContext *s,
SDPParseState *s1,
if (!strcmp(ff_rtp_enc_name(rtsp_st->sdp_payload_type), "MP2T")) {
/* no corresponding stream */
+ } else if (rt->server_type == RTSP_SERVER_WMS &&
+ codec_type == AVMEDIA_TYPE_DATA) {
+ /* RTX stream, a stream that carries all the other actual
+ * audio/video streams. Don't expose this to the callers. */
} else {
st = avformat_new_stream(s, NULL);
if (!st)
@@ -430,9 +434,11 @@ static void sdp_parse_line(AVFormatContext *s,
SDPParseState *s1,
/* NOTE: rtpmap is only supported AFTER the 'm=' tag */
get_word(buf1, sizeof(buf1), &p);
payload_type = atoi(buf1);
- st = s->streams[s->nb_streams - 1];
rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
- sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
+ if (rtsp_st->stream_index >= 0) {
+ st = s->streams[rtsp_st->stream_index];
+ sdp_parse_rtpmap(s, st, rtsp_st, payload_type, p);
+ }
} else if (av_strstart(p, "fmtp:", &p) ||
av_strstart(p, "framesize:", &p)) {
/* NOTE: fmtp is only supported AFTER the 'a=rtpmap:xxx' tag */
@@ -467,14 +473,15 @@ static void sdp_parse_line(AVFormatContext *s,
SDPParseState *s1,
if (rt->server_type == RTSP_SERVER_WMS)
ff_wms_parse_sdp_a_line(s, p);
if (s->nb_streams > 0) {
+ rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
+
if (rt->server_type == RTSP_SERVER_REAL)
- ff_real_parse_sdp_a_line(s, s->nb_streams - 1, p);
+ ff_real_parse_sdp_a_line(s, rtsp_st->stream_index, p);
- rtsp_st = rt->rtsp_streams[rt->nb_rtsp_streams - 1];
if (rtsp_st->dynamic_handler &&
rtsp_st->dynamic_handler->parse_sdp_a_line)
rtsp_st->dynamic_handler->parse_sdp_a_line(s,
- s->nb_streams - 1,
+ rtsp_st->stream_index,
rtsp_st->dynamic_protocol_context, buf);
}
}
@@ -1250,8 +1257,9 @@ int ff_rtsp_make_setup_request(AVFormatContext *s, const
char *host, int port,
* UDP. When trying to set it up for TCP streams, the server
* will return an error. Therefore, we skip those streams. */
if (rt->server_type == RTSP_SERVER_WMS &&
- s->streams[rtsp_st->stream_index]->codec->codec_type ==
- AVMEDIA_TYPE_DATA)
+ (rtsp_st->stream_index < 0 ||
+ s->streams[rtsp_st->stream_index]->codec->codec_type ==
+ AVMEDIA_TYPE_DATA))
continue;
snprintf(transport, sizeof(transport) - 1,
"%s/TCP;", trans_pref);
--
1.7.9.4
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