Several SRT options are missing. Since pkg_config requires libsrt v1.3.0 and above, it should be able to support options added in libsrt v1.3.0 and below. This commit adds 8 SRT options. sndbuf, rcvbuf, lossmaxttl, minversion, streamid, smoother, messageapi and transtype The keys of option are equivalent to stransmit. https://github.com/Haivision/srt/blob/v1.3.0/apps/socketoptions.hpp#L196-L223 --- doc/protocols.texi | 85 ++++++++++++++++++++++++++++++++++++++++++-- libavformat/libsrt.c | 56 +++++++++++++++++++++++++++++ 2 files changed, 139 insertions(+), 2 deletions(-)
diff --git a/doc/protocols.texi b/doc/protocols.texi index b34f29eebf..5bd0985b0e 100644 --- a/doc/protocols.texi +++ b/doc/protocols.texi @@ -1306,10 +1306,10 @@ 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. +Set UDP receive buffer size, expressed in bytes. @item send_buffer_size=@var{bytes} -Set send buffer size, expressed in bytes. +Set UDP send buffer size, expressed in bytes. @item rw_timeout Set raise error timeout for read/write optations. @@ -1329,6 +1329,87 @@ have no chance of being delivered in time. It was automatically enabled in the sender if the receiver supports it. +@item sndbuf=@var{bytes} +Set send buffer size, expressed in bytes. + +@item rcvbuf=@var{bytes} +Set receive buffer size, expressed in bytes. + +Receive buffer must not be greater than @option{ffs}. + +@item lossmaxttl=@var{packets} +The value up to which the Reorder Tolerance may grow. When +Reorder Tolerance is > 0, then packet loss report is delayed +until that number of packets come in. Reorder Tolerance +increases every time a "belated" packet has come, but it +wasn't due to retransmission (that is, when UDP packets tend +to come out of order), with the difference between the latest +sequence and this packet's sequence, and not more than the +value of this option. By default it's 0, which means that this +mechanism is turned off, and the loss report is always sent +immediately upon experiencing a "gap" in sequences. + +@item minversion +The minimum SRT version that is required from the peer. A connection +to a peer that does not satisfy the minimum version requirement +will be rejected. + +The version format in hex is 0xXXYYZZ for x.y.z in human readable +form, where x = ("%d", (version>>16) & 0xff), etc. + +@item streamid=@var{string} +A string limited to 512 characters that can be set on the socket prior +to connecting. This stream ID will be able to be retrieved by the +listener side from the socket that is returned from srt_accept and +was connected by a socket with that set stream ID. SRT does not enforce +any special interpretation of the contents of this string. +This option doesn’t make sense in Rendezvous connection; the result +might be that simply one side will override the value from the other +side and it’s the matter of luck which one would win + +@item smoother=@var{live|file} +The type of Smoother used for the transmission for that socket, which +is responsible for the transmission and congestion control. The Smoother +type must be exactly the same on both connecting parties, otherwise +the connection is rejected. + +@item messageapi=@var{1|0} +When set, this socket uses the Message API, otherwise it uses Buffer +API. Note that in live mode (see @option{transtype}) there’s only +message API available. In File mode you can chose to use one of two modes: + +Stream API (default, when this option is false). In this mode you may +send as many data as you wish with one sending instruction, or even use +dedicated functions that read directly from a file. The internal facility +will take care of any speed and congestion control. When receiving, you +can also receive as many data as desired, the data not extracted will be +waiting for the next call. There is no boundary between data portions in +the Stream mode. + +Message API. In this mode your single sending instruction passes exactly +one piece of data that has boundaries (a message). Contrary to Live mode, +this message may span across multiple UDP packets and the only size +limitation is that it shall fit as a whole in the sending buffer. The +receiver shall use as large buffer as necessary to receive the message, +otherwise the message will not be given up. When the message is not +complete (not all packets received or there was a packet loss) it will +not be given up. + +@item transtype=@var{live|file} +Sets the transmission type for the socket, in particular, setting this +option sets multiple other parameters to their default values as required +for a particular transmission type. + +live: Set options as for live transmission. In this mode, you should +send by one sending instruction only so many data that fit in one UDP packet, +and limited to the value defined first in @option{payload_size} (1316 is +default in this mode). There is no speed control in this mode, only the +bandwidth control, if configured, in order to not exceed the bandwidth with +the overhead transmission (retransmitted and control packets). + +file: Set options as for non-live transmission. See @option{messageapi} +for further explanations + @end table For more information see: @url{https://github.com/Haivision/srt}. diff --git a/libavformat/libsrt.c b/libavformat/libsrt.c index fbfd6ace83..087d62d3c1 100644 --- a/libavformat/libsrt.c +++ b/libavformat/libsrt.c @@ -76,6 +76,14 @@ typedef struct SRTContext { int64_t rcvlatency; int64_t peerlatency; enum SRTMode mode; + int sndbuf; + int rcvbuf; + int lossmaxttl; + int minversion; + char *streamid; + char *smoother; + int messageapi; + SRT_TRANSTYPE transtype; } SRTContext; #define D AV_OPT_FLAG_DECODING_PARAM @@ -110,6 +118,16 @@ static const AVOption libsrt_options[] = { { "caller", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRT_MODE_CALLER }, INT_MIN, INT_MAX, .flags = D|E, "mode" }, { "listener", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRT_MODE_LISTENER }, INT_MIN, INT_MAX, .flags = D|E, "mode" }, { "rendezvous", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRT_MODE_RENDEZVOUS }, INT_MIN, INT_MAX, .flags = D|E, "mode" }, + { "sndbuf", "Send buffer size (in bytes)", OFFSET(sndbuf), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E }, + { "rcvbuf", "Receive buffer size (in bytes)", OFFSET(rcvbuf), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E }, + { "lossmaxttl", "Maximum possible packet reorder tolerance", OFFSET(lossmaxttl), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E }, + { "minversion", "The minimum SRT version that is required from the peer", OFFSET(minversion), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E }, + { "streamid", "A string of up to 512 characters that an Initiator can pass to a Responder", OFFSET(streamid), AV_OPT_TYPE_STRING, { .str = NULL }, .flags = D|E }, + { "smoother", "The type of Smoother used for the transmission for that socket", OFFSET(smoother), AV_OPT_TYPE_STRING, { .str = NULL }, .flags = D|E }, + { "messageapi", "Enable message API", OFFSET(messageapi), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 1, .flags = D|E }, + { "transtype", "The transmission type for the socket", OFFSET(transtype), AV_OPT_TYPE_INT, { .i64 = SRTT_INVALID }, SRTT_LIVE, SRTT_INVALID, .flags = D|E, "transtype" }, + { "live", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRTT_LIVE }, INT_MIN, INT_MAX, .flags = D|E, "transtype" }, + { "file", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRTT_FILE }, INT_MIN, INT_MAX, .flags = D|E, "transtype" }, { NULL } }; @@ -310,6 +328,14 @@ static int libsrt_set_options_pre(URLContext *h, int fd) (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->sndbuf >= 0 && libsrt_setsockopt(h, fd, SRTO_SNDBUF, "SRTO_SNDBUF", &s->sndbuf, sizeof(s->sndbuf)) < 0) || + (s->rcvbuf >= 0 && libsrt_setsockopt(h, fd, SRTO_RCVBUF, "SRTO_RCVBUF", &s->rcvbuf, sizeof(s->rcvbuf)) < 0) || + (s->lossmaxttl >= 0 && libsrt_setsockopt(h, fd, SRTO_LOSSMAXTTL, "SRTO_LOSSMAXTTL", &s->lossmaxttl, sizeof(s->lossmaxttl)) < 0) || + (s->minversion >= 0 && libsrt_setsockopt(h, fd, SRTO_MINVERSION, "SRTO_MINVERSION", &s->minversion, sizeof(s->minversion)) < 0) || + (s->streamid && libsrt_setsockopt(h, fd, SRTO_STREAMID, "SRTO_STREAMID", s->streamid, strlen(s->streamid)) < 0) || + (s->smoother && libsrt_setsockopt(h, fd, SRTO_SMOOTHER, "SRTO_SMOOTHER", s->smoother, strlen(s->smoother)) < 0) || + (s->messageapi >= 0 && libsrt_setsockopt(h, fd, SRTO_MESSAGEAPI, "SRTO_MESSAGEAPI", &s->messageapi, sizeof(s->messageapi)) < 0) || + (s->transtype != SRTT_INVALID && libsrt_setsockopt(h, fd, SRTO_TRANSTYPE, "SRTO_TRANSTYPE", &s->transtype, sizeof(s->transtype)) < 0) || (s->payload_size >= 0 && libsrt_setsockopt(h, fd, SRTO_PAYLOADSIZE, "SRTO_PAYLOADSIZE", &s->payload_size, sizeof(s->payload_size)) < 0)) { return AVERROR(EIO); } @@ -522,6 +548,36 @@ static int libsrt_open(URLContext *h, const char *uri, int flags) return AVERROR(EIO); } } + if (av_find_info_tag(buf, sizeof(buf), "sndbuf", p)) { + s->sndbuf = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "rcvbuf", p)) { + s->rcvbuf = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "lossmaxttl", p)) { + s->lossmaxttl = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "minversion", p)) { + s->minversion = strtol(buf, NULL, 0); + } + if (av_find_info_tag(buf, sizeof(buf), "streamid", p)) { + s->streamid = av_strndup(buf, strlen(buf)); + } + if (av_find_info_tag(buf, sizeof(buf), "smoother", p)) { + s->smoother = av_strndup(buf, strlen(buf)); + } + if (av_find_info_tag(buf, sizeof(buf), "messageapi", p)) { + s->messageapi = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "transtype", p)) { + if (!strcmp(buf, "live")) { + s->transtype = SRTT_LIVE; + } else if (!strcmp(buf, "file")) { + s->transtype = SRTT_FILE; + } else { + return AVERROR(EIO); + } + } } return libsrt_setup(h, uri, flags); } -- 2.17.1 (Apple Git-112) _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel