ffmpeg | branch: master | James Almer <jamr...@gmail.com> | Sun Sep 2 19:11:45 2018 -0300| [de33b3e457a656230fc6d544a1889218d77a5b3c] | committer: James Almer
Merge commit 'a507af97eef468238d545ff954a39d7432832e54' * commit 'a507af97eef468238d545ff954a39d7432832e54': avformat/libsrt: add latency options and deprecate tspbdelay Merged-by: James Almer <jamr...@gmail.com> > http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=de33b3e457a656230fc6d544a1889218d77a5b3c --- doc/protocols.texi | 31 +++++++++++++++++++++++++++---- libavformat/libsrt.c | 28 +++++++++++++++++++++++----- libavformat/version.h | 2 +- 3 files changed, 51 insertions(+), 10 deletions(-) diff --git a/doc/protocols.texi b/doc/protocols.texi index 6322581c86..fad6c44c24 100644 --- a/doc/protocols.texi +++ b/doc/protocols.texi @@ -1210,6 +1210,17 @@ IP Type of Service. Applies to sender only. Default value is 0xB8. @item ipttl=@var{ttl} IP Time To Live. Applies to sender only. Default value is 64. +@item latency +Timestamp-based Packet Delivery Delay. +Used to absorb bursts of missed packet retransmissions. +This flag sets both @option{rcvlatency} and @option{peerlatency} +to the same value. Note that prior to version 1.3.0 +this is the only flag to set the latency, however +this is effectively equivalent to setting @option{peerlatency}, +when side is sender and @option{rcvlatency} +when side is receiver, and the bidirectional stream +sending is not supported. + @item listen_timeout Set socket listen timeout. @@ -1270,6 +1281,10 @@ use a bigger maximum frame size, though not greater than @item pkt_size=@var{bytes} Alias for @samp{payload_size}. +@item peerlatency +The latency value (as described in @option{rcvlatency}) that is +set by the sender side as a minimum value for the receiver. + @item pbkeylen=@var{bytes} Sender encryption key length, in bytes. Only can be set to 0, 16, 24 and 32. @@ -1278,6 +1293,18 @@ Not required on receiver (set to 0), key size obtained from sender in HaiCrypt handshake. Default value is 0. +@item rcvlatency +The time that should elapse since the moment when the +packet was sent and the moment when it's delivered to +the receiver application in the receiving function. +This time should be a buffer time large enough to cover +the time spent for sending, unexpectedly extended RTT +time, and the time needed to retransmit the lost UDP +packet. The effective latency value will be the maximum +of this options' value and the value of @option{peerlatency} +set by the peer side. Before version 1.3.0 this option +is only available as @option{latency}. + @item recv_buffer_size=@var{bytes} Set receive buffer size, expressed in bytes. @@ -1302,10 +1329,6 @@ have no chance of being delivered in time. It was automatically enabled in the sender if the receiver supports it. -@item tsbpddelay -Timestamp-based Packet Delivery Delay. -Used to absorb burst of missed packet retransmission. - @end table For more information see: @url{https://github.com/Haivision/srt}. diff --git a/libavformat/libsrt.c b/libavformat/libsrt.c index cb30f23eb3..c19d85192f 100644 --- a/libavformat/libsrt.c +++ b/libavformat/libsrt.c @@ -68,11 +68,13 @@ typedef struct SRTContext { int iptos; int64_t inputbw; int oheadbw; - int64_t tsbpddelay; + int64_t latency; int tlpktdrop; int nakreport; int64_t connect_timeout; int payload_size; + int64_t rcvlatency; + int64_t peerlatency; enum SRTMode mode; } SRTContext; @@ -97,7 +99,10 @@ static const AVOption libsrt_options[] = { { "iptos", "IP Type of Service", OFFSET(iptos), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 255, .flags = D|E }, { "inputbw", "Estimated input stream rate", OFFSET(inputbw), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E }, { "oheadbw", "MaxBW ceiling based on % over input stream rate", OFFSET(oheadbw), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 100, .flags = D|E }, - { "tsbpddelay", "TsbPd receiver delay to absorb burst of missed packet retransmission", OFFSET(tsbpddelay), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E }, + { "latency", "receiver delay to absorb bursts of missed packet retransmissions", OFFSET(latency), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E }, + { "tsbpddelay", "deprecated, same effect as latency option", OFFSET(latency), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E }, + { "rcvlatency", "receive latency", OFFSET(rcvlatency), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E }, + { "peerlatency", "peer latency", OFFSET(peerlatency), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E }, { "tlpktdrop", "Enable receiver pkt drop", OFFSET(tlpktdrop), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 1, .flags = D|E }, { "nakreport", "Enable receiver to send periodic NAK reports", OFFSET(nakreport), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 1, .flags = D|E }, { "connect_timeout", "Connect timeout. Caller default: 3000, rendezvous (x 10)", OFFSET(connect_timeout), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E }, @@ -286,7 +291,9 @@ static int libsrt_set_options_pre(URLContext *h, int fd) { SRTContext *s = h->priv_data; int yes = 1; - int tsbpddelay = s->tsbpddelay / 1000; + int latency = s->latency / 1000; + int rcvlatency = s->rcvlatency / 1000; + int peerlatency = s->peerlatency / 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) || @@ -297,7 +304,9 @@ static int libsrt_set_options_pre(URLContext *h, int fd) (s->ffs >= 0 && libsrt_setsockopt(h, fd, SRTO_FC, "SRTO_FC", &s->ffs, sizeof(s->ffs)) < 0) || (s->ipttl >= 0 && libsrt_setsockopt(h, fd, SRTO_IPTTL, "SRTO_UPTTL", &s->ipttl, sizeof(s->ipttl)) < 0) || (s->iptos >= 0 && libsrt_setsockopt(h, fd, SRTO_IPTOS, "SRTO_IPTOS", &s->iptos, sizeof(s->iptos)) < 0) || - (tsbpddelay >= 0 && libsrt_setsockopt(h, fd, SRTO_TSBPDDELAY, "SRTO_TSBPDELAY", &tsbpddelay, sizeof(tsbpddelay)) < 0) || + (s->latency >= 0 && libsrt_setsockopt(h, fd, SRTO_LATENCY, "SRTO_LATENCY", &latency, sizeof(latency)) < 0) || + (s->rcvlatency >= 0 && libsrt_setsockopt(h, fd, SRTO_RCVLATENCY, "SRTO_RCVLATENCY", &rcvlatency, sizeof(rcvlatency)) < 0) || + (s->peerlatency >= 0 && libsrt_setsockopt(h, fd, SRTO_PEERLATENCY, "SRTO_PEERLATENCY", &peerlatency, sizeof(peerlatency)) < 0) || (s->tlpktdrop >= 0 && libsrt_setsockopt(h, fd, SRTO_TLPKTDROP, "SRTO_TLPKDROP", &s->tlpktdrop, sizeof(s->tlpktdrop)) < 0) || (s->nakreport >= 0 && libsrt_setsockopt(h, fd, SRTO_NAKREPORT, "SRTO_NAKREPORT", &s->nakreport, sizeof(s->nakreport)) < 0) || (connect_timeout >= 0 && libsrt_setsockopt(h, fd, SRTO_CONNTIMEO, "SRTO_CONNTIMEO", &connect_timeout, sizeof(connect_timeout)) <0 )) || @@ -477,8 +486,17 @@ static int libsrt_open(URLContext *h, const char *uri, int flags) if (av_find_info_tag(buf, sizeof(buf), "oheadbw", p)) { s->oheadbw = strtoll(buf, NULL, 10); } + if (av_find_info_tag(buf, sizeof(buf), "latency", p)) { + s->latency = strtol(buf, NULL, 10); + } if (av_find_info_tag(buf, sizeof(buf), "tsbpddelay", p)) { - s->tsbpddelay = strtol(buf, NULL, 10); + s->latency = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "rcvlatency", p)) { + s->rcvlatency = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "peerlatency", p)) { + s->peerlatency = strtol(buf, NULL, 10); } if (av_find_info_tag(buf, sizeof(buf), "tlpktdrop", p)) { s->tlpktdrop = strtol(buf, NULL, 10); diff --git a/libavformat/version.h b/libavformat/version.h index b65ba19f3c..9496b72200 100644 --- a/libavformat/version.h +++ b/libavformat/version.h @@ -33,7 +33,7 @@ // Also please add any ticket numbers that you believe might be affected here #define LIBAVFORMAT_VERSION_MAJOR 58 #define LIBAVFORMAT_VERSION_MINOR 17 -#define LIBAVFORMAT_VERSION_MICRO 105 +#define LIBAVFORMAT_VERSION_MICRO 106 #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \ LIBAVFORMAT_VERSION_MINOR, \ ====================================================================== diff --cc doc/protocols.texi index 6322581c86,c3d6e150e0..fad6c44c24 --- a/doc/protocols.texi +++ b/doc/protocols.texi @@@ -1267,9 -777,10 +1278,13 @@@ wrapping a live stream in very small fr use a bigger maximum frame size, though not greater than 1456 bytes. +@item pkt_size=@var{bytes} +Alias for @samp{payload_size}. + + @item peerlatency + The latency value (as described in @option{rcvlatency}) that is + set by the sender side as a minimum value for the receiver. + @item pbkeylen=@var{bytes} Sender encryption key length, in bytes. Only can be set to 0, 16, 24 and 32. diff --cc libavformat/libsrt.c index cb30f23eb3,8e44ce6b80..c19d85192f --- a/libavformat/libsrt.c +++ b/libavformat/libsrt.c @@@ -297,11 -294,13 +304,13 @@@ static int libsrt_set_options_pre(URLCo (s->ffs >= 0 && libsrt_setsockopt(h, fd, SRTO_FC, "SRTO_FC", &s->ffs, sizeof(s->ffs)) < 0) || (s->ipttl >= 0 && libsrt_setsockopt(h, fd, SRTO_IPTTL, "SRTO_UPTTL", &s->ipttl, sizeof(s->ipttl)) < 0) || (s->iptos >= 0 && libsrt_setsockopt(h, fd, SRTO_IPTOS, "SRTO_IPTOS", &s->iptos, sizeof(s->iptos)) < 0) || - (tsbpddelay >= 0 && libsrt_setsockopt(h, fd, SRTO_TSBPDDELAY, "SRTO_TSBPDELAY", &tsbpddelay, sizeof(tsbpddelay)) < 0) || + (s->latency >= 0 && libsrt_setsockopt(h, fd, SRTO_LATENCY, "SRTO_LATENCY", &latency, sizeof(latency)) < 0) || + (s->rcvlatency >= 0 && libsrt_setsockopt(h, fd, SRTO_RCVLATENCY, "SRTO_RCVLATENCY", &rcvlatency, sizeof(rcvlatency)) < 0) || + (s->peerlatency >= 0 && libsrt_setsockopt(h, fd, SRTO_PEERLATENCY, "SRTO_PEERLATENCY", &peerlatency, sizeof(peerlatency)) < 0) || (s->tlpktdrop >= 0 && libsrt_setsockopt(h, fd, SRTO_TLPKTDROP, "SRTO_TLPKDROP", &s->tlpktdrop, sizeof(s->tlpktdrop)) < 0) || (s->nakreport >= 0 && libsrt_setsockopt(h, fd, SRTO_NAKREPORT, "SRTO_NAKREPORT", &s->nakreport, sizeof(s->nakreport)) < 0) || - (connect_timeout >= 0 && libsrt_setsockopt(h, fd, SRTO_CONNTIMEO, "SRTO_CONNTIMEO", &connect_timeout, sizeof(connect_timeout)) < 0) || - (s->payload_size >= 0 && libsrt_setsockopt(h, fd, SRTO_PAYLOADSIZE, "SRTO_PAYLOADSIZE", &s->payload_size, sizeof(s->payload_size)) < 0)) { + (connect_timeout >= 0 && libsrt_setsockopt(h, fd, SRTO_CONNTIMEO, "SRTO_CONNTIMEO", &connect_timeout, sizeof(connect_timeout)) <0 )) || + (s->payload_size >= 0 && libsrt_setsockopt(h, fd, SRTO_PAYLOADSIZE, "SRTO_PAYLOADSIZE", &s->payload_size, sizeof(s->payload_size)) < 0) { return AVERROR(EIO); } return 0; diff --cc libavformat/version.h index b65ba19f3c,016f266f1f..9496b72200 --- a/libavformat/version.h +++ b/libavformat/version.h @@@ -29,11 -29,9 +29,11 @@@ #include "libavutil/version.h" -#define LIBAVFORMAT_VERSION_MAJOR 58 -#define LIBAVFORMAT_VERSION_MINOR 2 -#define LIBAVFORMAT_VERSION_MICRO 0 +// Major bumping may affect Ticket5467, 5421, 5451(compatibility with Chromium) +// Also please add any ticket numbers that you believe might be affected here +#define LIBAVFORMAT_VERSION_MAJOR 58 +#define LIBAVFORMAT_VERSION_MINOR 17 - #define LIBAVFORMAT_VERSION_MICRO 105 ++#define LIBAVFORMAT_VERSION_MICRO 106 #define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \ LIBAVFORMAT_VERSION_MINOR, \ _______________________________________________ ffmpeg-cvslog mailing list ffmpeg-cvslog@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-cvslog
