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avformat/libsrt: add several options supported in srt 1.3.0

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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

Signed-off-by: Marton Balint <cus@passwd.hu>
This commit is contained in:
Matsuzawa Tomohiro 2018-10-23 04:34:29 +00:00 committed by Marton Balint
parent 110b4a4918
commit c2ac3b8e6a
3 changed files with 142 additions and 3 deletions
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85
doc/protocols.texi
View File

@ -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}.

58
libavformat/libsrt.c
View File

@ -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 +548,38 @@ 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)) {
av_freep(&s->streamid);
s->streamid = av_strdup(buf);
}
if (av_find_info_tag(buf, sizeof(buf), "smoother", p)) {
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);
}

2
libavformat/version.h
View File

@ -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 19
#define LIBAVFORMAT_VERSION_MICRO 101
#define LIBAVFORMAT_VERSION_MICRO 102
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
LIBAVFORMAT_VERSION_MINOR, \