On Thu, 22. Oct 14:21, Limin Wang wrote:
> ffmpeg | branch: master | Limin Wang <lance.lmw...@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.lmw...@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.
Setting rt->intial_timeout in rtsp is also used to infer listen mode in
rtsp_read_header().
>
> 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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--
Andriy
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