Message ID | CY4PR06MB33354050426F508F2CB30984C7F40@CY4PR06MB3335.namprd06.prod.outlook.com |
---|---|
State | Superseded |
Headers | show |
On Mon, 22 Oct 2018, Matsuzawa Tomohiro wrote: > 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 | 62 ++++++++++++++++++++++++++++++++ > 2 files changed, 145 insertions(+), 2 deletions(-) > > diff --git a/doc/protocols.texi b/doc/protocols.texi > index b34f29eebf..fb7725e058 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. > + > +@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..e7a60dbc40 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 } > }; > > @@ -297,6 +315,7 @@ static int libsrt_set_options_pre(URLContext *h, int fd) > int connect_timeout = s->connect_timeout; > > if ((s->mode == SRT_MODE_RENDEZVOUS && libsrt_setsockopt(h, fd, SRTO_RENDEZVOUS, "SRTO_RENDEZVOUS", &yes, sizeof(yes)) < 0) || > + (s->transtype != SRTT_INVALID && libsrt_setsockopt(h, fd, SRTO_TRANSTYPE, "SRTO_TRANSTYPE", &s->transtype, sizeof(s->transtype)) < 0) || Technically this is the second, but it does not conflict with rendezvous, so it does not matter much. > (s->maxbw >= 0 && libsrt_setsockopt(h, fd, SRTO_MAXBW, "SRTO_MAXBW", &s->maxbw, sizeof(s->maxbw)) < 0) || > (s->pbkeylen >= 0 && libsrt_setsockopt(h, fd, SRTO_PBKEYLEN, "SRTO_PBKEYLEN", &s->pbkeylen, sizeof(s->pbkeylen)) < 0) || > (s->passphrase && libsrt_setsockopt(h, fd, SRTO_PASSPHRASE, "SRTO_PASSPHRASE", s->passphrase, strlen(s->passphrase)) < 0) || > @@ -310,6 +329,13 @@ 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->payload_size >= 0 && libsrt_setsockopt(h, fd, SRTO_PAYLOADSIZE, "SRTO_PAYLOADSIZE", &s->payload_size, sizeof(s->payload_size)) < 0)) { > return AVERROR(EIO); > } > @@ -522,6 +548,42 @@ 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)) { > + if (s->streamid) { > + av_freep(s->streamid); > + } av_freep needs a pointer to a pointer. Also the NULL check is unneeded, av_freep handles it just fine. > + s->streamid = av_strdup(buf); > + } > + if (av_find_info_tag(buf, sizeof(buf), "smoother", p)) { > + if (s->smoother) { > + av_freep(s->smoother); > + } Same here. Regards, Marton
diff --git a/doc/protocols.texi b/doc/protocols.texi index b34f29eebf..fb7725e058 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. + +@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..e7a60dbc40 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 } }; @@ -297,6 +315,7 @@ static int libsrt_set_options_pre(URLContext *h, int fd) int connect_timeout = s->connect_timeout; if ((s->mode == SRT_MODE_RENDEZVOUS && libsrt_setsockopt(h, fd, SRTO_RENDEZVOUS, "SRTO_RENDEZVOUS", &yes, sizeof(yes)) < 0) || + (s->transtype != SRTT_INVALID && libsrt_setsockopt(h, fd, SRTO_TRANSTYPE, "SRTO_TRANSTYPE", &s->transtype, sizeof(s->transtype)) < 0) || (s->maxbw >= 0 && libsrt_setsockopt(h, fd, SRTO_MAXBW, "SRTO_MAXBW", &s->maxbw, sizeof(s->maxbw)) < 0) || (s->pbkeylen >= 0 && libsrt_setsockopt(h, fd, SRTO_PBKEYLEN, "SRTO_PBKEYLEN", &s->pbkeylen, sizeof(s->pbkeylen)) < 0) || (s->passphrase && libsrt_setsockopt(h, fd, SRTO_PASSPHRASE, "SRTO_PASSPHRASE", s->passphrase, strlen(s->passphrase)) < 0) || @@ -310,6 +329,13 @@ 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->payload_size >= 0 && libsrt_setsockopt(h, fd, SRTO_PAYLOADSIZE, "SRTO_PAYLOADSIZE", &s->payload_size, sizeof(s->payload_size)) < 0)) { return AVERROR(EIO); } @@ -522,6 +548,42 @@ 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)) { + if (s->streamid) { + av_freep(s->streamid); + } + s->streamid = av_strdup(buf); + } + if (av_find_info_tag(buf, sizeof(buf), "smoother", p)) { + if (s->smoother) { + av_freep(s->smoother); + } + s->smoother = av_strdup(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(EINVAL); + } + } } return libsrt_setup(h, uri, flags); }