[FFmpeg-devel] [FFmpeg-cvslog] avformat/rtsp: support for listen_timeout option for sdp

[lance.lmwang at gmail.com]

On Mon, Nov 02, 2020 at 11:57:47PM -0500, Andriy Gelman wrote:
> On Thu, 22. Oct 14:21, Limin Wang wrote:
> > ffmpeg | branch: master | Limin Wang <lance.lmwang at gmail.com> | Sat Oct 10 21:30:21 2020 +0800| [92c40ef882be115e72d2aa02f9032b7ce88f8537] | committer: Limin Wang
> > 
> > avformat/rtsp: support for listen_timeout option for sdp
> > 
> > Now the listen timeout is hardcoded(10s).
> > How to test(30s timeout):
> > ./ffprobe  -listen_timeout 30 -protocol_whitelist rtp,udp,file -i test.sdp
> > 
> > Signed-off-by: Limin Wang <lance.lmwang at gmail.com>
> > 
> > > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=92c40ef882be115e72d2aa02f9032b7ce88f8537
> > ---
> > 
> >  libavformat/rtsp.c | 10 ++++++----
> >  1 file changed, 6 insertions(+), 4 deletions(-)
> > 
> > diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
> > index e9fca034b4..4ecd702780 100644
> > --- a/libavformat/rtsp.c
> > +++ b/libavformat/rtsp.c
> > @@ -55,7 +55,6 @@
> >   * and read_packet(), in seconds  */
> >  #define POLL_TIMEOUT_MS 100
> >  #define READ_PACKET_TIMEOUT_S 10
> > -#define MAX_TIMEOUTS READ_PACKET_TIMEOUT_S * 1000 / POLL_TIMEOUT_MS
> >  #define SDP_MAX_SIZE 16384
> >  #define RECVBUF_SIZE 10 * RTP_MAX_PACKET_LENGTH
> >  #define DEFAULT_REORDERING_DELAY 100000
> > @@ -115,6 +114,7 @@ static const AVOption sdp_options[] = {
> >      RTSP_FLAG_OPTS("sdp_flags", "SDP flags"),
> >      { "custom_io", "use custom I/O", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_CUSTOM_IO}, 0, 0, DEC, "rtsp_flags" },
> >      { "rtcp_to_source", "send RTCP packets to the source address of received packets", 0, AV_OPT_TYPE_CONST, {.i64 = RTSP_FLAG_RTCP_TO_SOURCE}, 0, 0, DEC, "rtsp_flags" },
> > +    { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = READ_PACKET_TIMEOUT_S}, INT_MIN, INT_MAX, DEC },
> >      RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
> >      COMMON_OPTS(),
> >      { NULL },
> > @@ -122,6 +122,8 @@ static const AVOption sdp_options[] = {
> >  
> >  static const AVOption rtp_options[] = {
> >      RTSP_FLAG_OPTS("rtp_flags", "set RTP flags"),
> > +    { "listen_timeout", "set maximum timeout (in seconds) to wait for incoming connections", OFFSET(initial_timeout), AV_OPT_TYPE_INT, {.i64 = READ_PACKET_TIMEOUT_S}, INT_MIN, INT_MAX, DEC },
> > +    RTSP_MEDIATYPE_OPTS("allowed_media_types", "set media types to accept from the server"),
> >      COMMON_OPTS(),
> >      { NULL },
> >  };
> > @@ -1985,9 +1987,10 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
> >  {
> >      RTSPState *rt = s->priv_data;
> >      RTSPStream *rtsp_st;
> > -    int n, i, ret, timeout_cnt = 0;
> > +    int n, i, ret;
> >      struct pollfd *p = rt->p;
> >      int *fds = NULL, fdsnum, fdsidx;
> 
> > +    int runs = rt->initial_timeout * 1000LL / POLL_TIMEOUT_MS;
> 
> This is not correct for rtsp. Default value for for rt->initial_timeout in rtsp is -1.

will submit a patch soon, please help to test and review.

> 
> Setting rt->intial_timeout in rtsp is also used to infer listen mode in
> rtsp_read_header().

What's your suggestion? for rtsp, if listen_timeout is set, it'll used for both listen timeout
and  rtp stream poll tiemout.  

I only tested with below command:
s:
./ffmpeg -y  -listen_timeout 4  -f rtsp -i rtsp://127.0.0.1:8000/ok -f null -
c:
./ffmpeg   -f lavfi -i testsrc=duration=2000:size=vga:rate=25  -c:v libx264  -f rtsp rtsp://127.0.0.1:8000/ok


> 
> >  
> >      if (!p) {
> >          p = rt->p = av_malloc_array(2 * rt->nb_rtsp_streams + 1, sizeof(struct pollfd));
> > @@ -2028,7 +2031,6 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
> >          n = poll(p, rt->max_p, POLL_TIMEOUT_MS);
> >          if (n > 0) {
> >              int j = rt->rtsp_hd ? 1 : 0;
> > -            timeout_cnt = 0;
> >              for (i = 0; i < rt->nb_rtsp_streams; i++) {
> >                  rtsp_st = rt->rtsp_streams[i];
> >                  if (rtsp_st->rtp_handle) {
> > @@ -2049,7 +2051,7 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st,
> >                  }
> >              }
> >  #endif
> > -        } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) {
> > +        } else if (n == 0 && --runs <= 0) {
> >              return AVERROR(ETIMEDOUT);
> >          } else if (n < 0 && errno != EINTR)
> >              return AVERROR(errno);
> > 
> > _______________________________________________
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> > 
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> 
> -- 
> Andriy
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-- 
Thanks,
Limin Wang