Re: [FFmpeg-devel] [PATCH v3 1/2] avformat/rtsp: fix infinite loop with udp transport
Hi, On Thu, 1 Oct 2020, Andriy Gelman wrote: On Wed, 30. Sep 12:41, Martin Storsjö wrote: Hi, On Sun, 27 Sep 2020, Zhao Zhili wrote: > Fix #8840. > > Steps to reproduce: > 1. sender: > ./ffmpeg -i test.mp4 -c copy -f rtsp -rtsp_transport udp rtsp://localhost:12345/live.sdp > 2. receiver: > ./ffmpeg_g -y -rtsp_flags listen -timeout 100 -i rtsp://localhost:12345/live.sdp -c copy test.mp4 > --- > v3: mention the ticket. > > libavformat/rtsp.c| 2 ++ > libavformat/rtsp.h| 1 + > libavformat/rtspdec.c | 2 +- > 3 files changed, 4 insertions(+), 1 deletion(-) > > diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c > index 5d8491b74b..597413803f 100644 > --- a/libavformat/rtsp.c > +++ b/libavformat/rtsp.c > @@ -2051,6 +2051,8 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, > if ((ret = parse_rtsp_message(s)) < 0) { > return ret; > } > +if (rt->state == RTSP_STATE_TEARDOWN) > +return AVERROR_EOF; > } > #endif > } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) { > diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h > index 54a9a30c16..481cc0c3ce 100644 > --- a/libavformat/rtsp.h > +++ b/libavformat/rtsp.h > @@ -198,6 +198,7 @@ enum RTSPClientState { > RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */ > RTSP_STATE_PAUSED, /**< initialized, but not receiving data */ > RTSP_STATE_SEEKING, /**< initialized, requesting a seek */ > +RTSP_STATE_TEARDOWN,/**< initialized, in teardown state */ > }; > > /** > diff --git a/libavformat/rtspdec.c b/libavformat/rtspdec.c > index dfa29913bf..ec786a469a 100644 > --- a/libavformat/rtspdec.c > +++ b/libavformat/rtspdec.c > @@ -494,7 +494,7 @@ int ff_rtsp_parse_streaming_commands(AVFormatContext *s) > "Public: ANNOUNCE, PAUSE, SETUP, TEARDOWN, " > "RECORD\r\n", request.seq); > } else if (methodcode == TEARDOWN) { > -rt->state = RTSP_STATE_IDLE; > +rt->state = RTSP_STATE_TEARDOWN; > ret = rtsp_send_reply(s, RTSP_STATUS_OK, NULL , request.seq); > } > return ret; > -- > 2.17.1 Martin, thanks for looking over the patch. I think this approach to fixing it, adding a new state, is a bit of overkill. This usecase actually used to work originally, but I bisected it and it broke in f6161fccf8c5720ceac1ed1df8ba60ff8fed69f5. So with that in mind, it's pretty straightforward to fix it by retaining the original behaviour from before that commit. I'll send an alternative patch that does that. I made the suggestion to add TEARDOWN state because I thought it's safer than relying on the idle state, and it made the code more readable. Yeah, it's not a bad idea per se - but adding more states is generally more risky (more codepaths that might need to take it into account, etc). And in this case, as it's a regression, it's easier to fix it as a specific fix for that breakage. Looking at your patch, I think it's a clean/elegant solution, and also looks good to me. Great, thanks! So if Zhao also is ok with it, I'd push that one, and patch 2/2 from Zhao's set. // Martin ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-devel] [PATCH v3 1/2] avformat/rtsp: fix infinite loop with udp transport
On Wed, 30. Sep 12:41, Martin Storsjö wrote: > Hi, > > On Sun, 27 Sep 2020, Zhao Zhili wrote: > > > Fix #8840. > > > > Steps to reproduce: > > 1. sender: > > ./ffmpeg -i test.mp4 -c copy -f rtsp -rtsp_transport udp > > rtsp://localhost:12345/live.sdp > > 2. receiver: > > ./ffmpeg_g -y -rtsp_flags listen -timeout 100 -i > > rtsp://localhost:12345/live.sdp -c copy test.mp4 > > --- > > v3: mention the ticket. > > > > libavformat/rtsp.c| 2 ++ > > libavformat/rtsp.h| 1 + > > libavformat/rtspdec.c | 2 +- > > 3 files changed, 4 insertions(+), 1 deletion(-) > > > > diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c > > index 5d8491b74b..597413803f 100644 > > --- a/libavformat/rtsp.c > > +++ b/libavformat/rtsp.c > > @@ -2051,6 +2051,8 @@ static int udp_read_packet(AVFormatContext *s, > > RTSPStream **prtsp_st, > > if ((ret = parse_rtsp_message(s)) < 0) { > > return ret; > > } > > +if (rt->state == RTSP_STATE_TEARDOWN) > > +return AVERROR_EOF; > > } > > #endif > > } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) { > > diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h > > index 54a9a30c16..481cc0c3ce 100644 > > --- a/libavformat/rtsp.h > > +++ b/libavformat/rtsp.h > > @@ -198,6 +198,7 @@ enum RTSPClientState { > > RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */ > > RTSP_STATE_PAUSED, /**< initialized, but not receiving data */ > > RTSP_STATE_SEEKING, /**< initialized, requesting a seek */ > > +RTSP_STATE_TEARDOWN,/**< initialized, in teardown state */ > > }; > > > > /** > > diff --git a/libavformat/rtspdec.c b/libavformat/rtspdec.c > > index dfa29913bf..ec786a469a 100644 > > --- a/libavformat/rtspdec.c > > +++ b/libavformat/rtspdec.c > > @@ -494,7 +494,7 @@ int ff_rtsp_parse_streaming_commands(AVFormatContext *s) > > "Public: ANNOUNCE, PAUSE, SETUP, TEARDOWN, " > > "RECORD\r\n", request.seq); > > } else if (methodcode == TEARDOWN) { > > -rt->state = RTSP_STATE_IDLE; > > +rt->state = RTSP_STATE_TEARDOWN; > > ret = rtsp_send_reply(s, RTSP_STATUS_OK, NULL , request.seq); > > } > > return ret; > > -- > > 2.17.1 Martin, thanks for looking over the patch. > > I think this approach to fixing it, adding a new state, is a bit of > overkill. This usecase actually used to work originally, but I bisected it > and it broke in f6161fccf8c5720ceac1ed1df8ba60ff8fed69f5. So with that in > mind, it's pretty straightforward to fix it by retaining the original > behaviour from before that commit. I'll send an alternative patch that does > that. I made the suggestion to add TEARDOWN state because I thought it's safer than relying on the idle state, and it made the code more readable. Looking at your patch, I think it's a clean/elegant solution, and also looks good to me. -- Andriy ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-devel] [PATCH v3 1/2] avformat/rtsp: fix infinite loop with udp transport
Hi, On Sun, 27 Sep 2020, Zhao Zhili wrote: Fix #8840. Steps to reproduce: 1. sender: ./ffmpeg -i test.mp4 -c copy -f rtsp -rtsp_transport udp rtsp://localhost:12345/live.sdp 2. receiver: ./ffmpeg_g -y -rtsp_flags listen -timeout 100 -i rtsp://localhost:12345/live.sdp -c copy test.mp4 --- v3: mention the ticket. libavformat/rtsp.c| 2 ++ libavformat/rtsp.h| 1 + libavformat/rtspdec.c | 2 +- 3 files changed, 4 insertions(+), 1 deletion(-) diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index 5d8491b74b..597413803f 100644 --- a/libavformat/rtsp.c +++ b/libavformat/rtsp.c @@ -2051,6 +2051,8 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, if ((ret = parse_rtsp_message(s)) < 0) { return ret; } +if (rt->state == RTSP_STATE_TEARDOWN) +return AVERROR_EOF; } #endif } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) { diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h index 54a9a30c16..481cc0c3ce 100644 --- a/libavformat/rtsp.h +++ b/libavformat/rtsp.h @@ -198,6 +198,7 @@ enum RTSPClientState { RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */ RTSP_STATE_PAUSED, /**< initialized, but not receiving data */ RTSP_STATE_SEEKING, /**< initialized, requesting a seek */ +RTSP_STATE_TEARDOWN,/**< initialized, in teardown state */ }; /** diff --git a/libavformat/rtspdec.c b/libavformat/rtspdec.c index dfa29913bf..ec786a469a 100644 --- a/libavformat/rtspdec.c +++ b/libavformat/rtspdec.c @@ -494,7 +494,7 @@ int ff_rtsp_parse_streaming_commands(AVFormatContext *s) "Public: ANNOUNCE, PAUSE, SETUP, TEARDOWN, " "RECORD\r\n", request.seq); } else if (methodcode == TEARDOWN) { -rt->state = RTSP_STATE_IDLE; +rt->state = RTSP_STATE_TEARDOWN; ret = rtsp_send_reply(s, RTSP_STATUS_OK, NULL , request.seq); } return ret; -- 2.17.1 I think this approach to fixing it, adding a new state, is a bit of overkill. This usecase actually used to work originally, but I bisected it and it broke in f6161fccf8c5720ceac1ed1df8ba60ff8fed69f5. So with that in mind, it's pretty straightforward to fix it by retaining the original behaviour from before that commit. I'll send an alternative patch that does that. // Martin ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-devel] [PATCH v3 1/2] avformat/rtsp: fix infinite loop with udp transport
Fix #8840. Steps to reproduce: 1. sender: ./ffmpeg -i test.mp4 -c copy -f rtsp -rtsp_transport udp rtsp://localhost:12345/live.sdp 2. receiver: ./ffmpeg_g -y -rtsp_flags listen -timeout 100 -i rtsp://localhost:12345/live.sdp -c copy test.mp4 --- v3: mention the ticket. libavformat/rtsp.c| 2 ++ libavformat/rtsp.h| 1 + libavformat/rtspdec.c | 2 +- 3 files changed, 4 insertions(+), 1 deletion(-) diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c index 5d8491b74b..597413803f 100644 --- a/libavformat/rtsp.c +++ b/libavformat/rtsp.c @@ -2051,6 +2051,8 @@ static int udp_read_packet(AVFormatContext *s, RTSPStream **prtsp_st, if ((ret = parse_rtsp_message(s)) < 0) { return ret; } +if (rt->state == RTSP_STATE_TEARDOWN) +return AVERROR_EOF; } #endif } else if (n == 0 && ++timeout_cnt >= MAX_TIMEOUTS) { diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h index 54a9a30c16..481cc0c3ce 100644 --- a/libavformat/rtsp.h +++ b/libavformat/rtsp.h @@ -198,6 +198,7 @@ enum RTSPClientState { RTSP_STATE_STREAMING, /**< initialized and sending/receiving data */ RTSP_STATE_PAUSED, /**< initialized, but not receiving data */ RTSP_STATE_SEEKING, /**< initialized, requesting a seek */ +RTSP_STATE_TEARDOWN,/**< initialized, in teardown state */ }; /** diff --git a/libavformat/rtspdec.c b/libavformat/rtspdec.c index dfa29913bf..ec786a469a 100644 --- a/libavformat/rtspdec.c +++ b/libavformat/rtspdec.c @@ -494,7 +494,7 @@ int ff_rtsp_parse_streaming_commands(AVFormatContext *s) "Public: ANNOUNCE, PAUSE, SETUP, TEARDOWN, " "RECORD\r\n", request.seq); } else if (methodcode == TEARDOWN) { -rt->state = RTSP_STATE_IDLE; +rt->state = RTSP_STATE_TEARDOWN; ret = rtsp_send_reply(s, RTSP_STATUS_OK, NULL , request.seq); } return ret; -- 2.17.1 ___ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-devel To unsubscribe, visit link above, or email ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".