> On Sep 30, 2021, at 9:14 AM, lance.lmw...@gmail.com wrote: > > From: Limin Wang <lance.lmw...@gmail.com> > > Signed-off-by: Limin Wang <lance.lmw...@gmail.com> > --- > doc/protocols.texi | 10 ++++++++++ > libavformat/libsrt.c | 7 +++++++ > 2 files changed, 17 insertions(+) > > diff --git a/doc/protocols.texi b/doc/protocols.texi > index 726e5f1..9c246f8 100644 > --- a/doc/protocols.texi > +++ b/doc/protocols.texi > @@ -1496,6 +1496,16 @@ when the old encryption key is decommissioned. Default > is -1. > -1 means auto (0x1000 in srt library). The range for > this option is integers in the 0 - @code{INT_MAX}. > > +@item snddropdelay=@var{microseconds} > +The sender's delay before dropping packets. This delay is > +added to the default drop delay time interval value. > +Keep in mind that the longer the delay, the more probable it > +becomes that packets would be retransmitted uselessly because > +they will be dropped by the receiver anyway. > + > +Default is -1 means auto which typically means do not drop > +packets on the sender at all.
There are two issues here: Firstly, -1 means auto for our libsrt wrapper. "typically means do not drop packets on the sender at all” is incorrect. The default value of libsrt is 0 for live mode, -1 for file mode. Live mode is the default and the file mode doesn’t matter much for FFmpeg, TBH. Secondly, since our wrapper and libsrt defined different sematic for -1, now it’s impossible to pass -1 to libsrt. And the description from libsrt doc > Keep in mind that the longer the delay, the more probable it becomes that > packets would be > retransmitted can mislead the user to set a higher value for it, which is not a good idea. I don’t know how to improve that. > + > @item payload_size=@var{bytes} > Sets the maximum declared size of a packet transferred > during the single call to the sending function in Live > diff --git a/libavformat/libsrt.c b/libavformat/libsrt.c > index c6308d1..13697d2 100644 > --- a/libavformat/libsrt.c > +++ b/libavformat/libsrt.c > @@ -65,6 +65,7 @@ typedef struct SRTContext { > int enforced_encryption; > int kmrefreshrate; > int kmpreannounce; > + int64_t snddropdelay; > #endif > int mss; > int ffs; > @@ -111,6 +112,7 @@ static const AVOption libsrt_options[] = { > { "enforced_encryption", "Enforces that both connection parties have the > same passphrase set", > OFFSET(enforced_encryption), AV_OPT_TYPE_BOOL, { .i64 = -1 }, -1, 1, > .flags = D|E }, > { "kmrefreshrate", "The number of packets to be transmitted after > which the encryption key is switched to a new key", OFFSET(kmrefreshrate), > AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E }, > { "kmpreannounce", "The interval between when a new encryption key > is sent and when switchover occurs", OFFSET(kmpreannounce), > AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E }, > + { "snddropdelay", "The sender's delay(in microseconds) before > dropping packets", OFFSET(snddropdelay), > AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E }, > #endif > { "mss", "The Maximum Segment Size", > OFFSET(mss), AV_OPT_TYPE_INT, { .i64 = -1 > }, -1, 1500, .flags = D|E }, > { "ffs", "Flight flag size (window size) (in bytes)", > OFFSET(ffs), AV_OPT_TYPE_INT, { .i64 = -1 > }, -1, INT_MAX, .flags = D|E }, > @@ -318,6 +320,7 @@ static int libsrt_set_options_pre(URLContext *h, int fd) > int latency = s->latency / 1000; > int rcvlatency = s->rcvlatency / 1000; > int peerlatency = s->peerlatency / 1000; > + int snddropdelay = s->snddropdelay / 1000; > int connect_timeout = s->connect_timeout; > > if ((s->mode == SRT_MODE_RENDEZVOUS && libsrt_setsockopt(h, fd, > SRTO_RENDEZVOUS, "SRTO_RENDEZVOUS", &yes, sizeof(yes)) < 0) || > @@ -334,6 +337,7 @@ static int libsrt_set_options_pre(URLContext *h, int fd) > #endif > (s->kmrefreshrate >= 0 && libsrt_setsockopt(h, fd, > SRTO_KMREFRESHRATE, "SRTO_KMREFRESHRATE", &s->kmrefreshrate, > sizeof(s->kmrefreshrate)) < 0) || > (s->kmpreannounce >= 0 && libsrt_setsockopt(h, fd, > SRTO_KMPREANNOUNCE, "SRTO_KMPREANNOUNCE", &s->kmpreannounce, > sizeof(s->kmpreannounce)) < 0) || > + (s->snddropdelay >= 0 && libsrt_setsockopt(h, fd, > SRTO_SNDDROPDELAY, "SRTO_SNDDROPDELAY", &snddropdelay, > sizeof(snddropdelay)) < 0) || > #endif > (s->mss >= 0 && libsrt_setsockopt(h, fd, SRTO_MSS, "SRTO_MSS", > &s->mss, sizeof(s->mss)) < 0) || > (s->ffs >= 0 && libsrt_setsockopt(h, fd, SRTO_FC, "SRTO_FC", &s->ffs, > sizeof(s->ffs)) < 0) || > @@ -549,6 +553,9 @@ static int libsrt_open(URLContext *h, const char *uri, > int flags) > if (av_find_info_tag(buf, sizeof(buf), "kmpreannounce", p)) { > s->kmpreannounce = strtol(buf, NULL, 10); > } > + if (av_find_info_tag(buf, sizeof(buf), "snddropdelay", p)) { > + s->snddropdelay = strtoll(buf, NULL, 10); > + } > #endif > if (av_find_info_tag(buf, sizeof(buf), "mss", p)) { > s->mss = strtol(buf, NULL, 10); > -- > 1.8.3.1 > > _______________________________________________ > 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". > _______________________________________________ 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".