[FFmpeg-cvslog] avformat/libsrt: add several options supported in srt 1.3.0
Matsuzawa Tomohiro
git at videolan.org
Tue Oct 23 20:59:09 EEST 2018
ffmpeg | branch: master | Matsuzawa Tomohiro <thmatuza75 at hotmail.com> | Tue Oct 23 04:34:29 2018 +0000| [c2ac3b8e6a040e33d53fa13548848c8ba981a8e4] | committer: Marton Balint
avformat/libsrt: add several options supported in srt 1.3.0
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 at passwd.hu>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=c2ac3b8e6a040e33d53fa13548848c8ba981a8e4
---
doc/protocols.texi | 85 +++++++++++++++++++++++++++++++++++++++++++++++++--
libavformat/libsrt.c | 58 +++++++++++++++++++++++++++++++++++
libavformat/version.h | 2 +-
3 files changed, 142 insertions(+), 3 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.
+ at item sndbuf=@var{bytes}
+Set send buffer size, expressed in bytes.
+
+ at item rcvbuf=@var{bytes}
+Set receive buffer size, expressed in bytes.
+
+Receive buffer must not be greater than @option{ffs}.
+
+ at 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.
+
+ at 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.
+
+ at 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
+
+ at 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.
+
+ at 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.
+
+ at 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..fe3b312151 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 +548,38 @@ 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);
}
diff --git a/libavformat/version.h b/libavformat/version.h
index e2d0cfd414..855c3dba2b 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 19
-#define LIBAVFORMAT_VERSION_MICRO 101
+#define LIBAVFORMAT_VERSION_MICRO 102
#define LIBAVFORMAT_VERSION_INT AV_VERSION_INT(LIBAVFORMAT_VERSION_MAJOR, \
LIBAVFORMAT_VERSION_MINOR, \
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