[FFmpeg-devel] [PATCH] avformat/libsrt: add several options supported in srt 1.3.0 v2
Marton Balint
cus at passwd.hu
Mon Oct 22 12:35:24 EEST 2018
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.
>
> + 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..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
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