[FFmpeg-cvslog] Add Haivision SRT protocol
Sven Dueking
git at videolan.org
Fri Mar 30 03:59:39 EEST 2018
ffmpeg | branch: master | Sven Dueking <sven.dueking at nablet.com> | Mon Mar 26 11:37:49 2018 -0400| [a2fc8dbae85339d1b418d296f2982b6c04c53c57] | committer: Luca Barbato
Add Haivision SRT protocol
The protocol requires libsrt (https://github.com/Haivision/srt) to be
installed
Signed-off-by: Sven Dueking <sven.dueking at nablet.com>
Signed-off-by: Luca Barbato <lu_zero at gentoo.org>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=a2fc8dbae85339d1b418d296f2982b6c04c53c57
---
Changelog | 1 +
configure | 5 +
doc/protocols.texi | 140 +++++++++++++
libavformat/Makefile | 3 +
libavformat/libsrt.c | 546 ++++++++++++++++++++++++++++++++++++++++++++++++
libavformat/protocols.c | 1 +
6 files changed, 696 insertions(+)
diff --git a/Changelog b/Changelog
index 0d20cd47df..53562a1aa0 100644
--- a/Changelog
+++ b/Changelog
@@ -21,6 +21,7 @@ version <next>:
- NVIDIA CUVID-accelerated H.264 and HEVC decoding
- Intel QSV-accelerated overlay filter
- AV1 Support through libaom
+- Haivision SRT protocol via libsrt
version 12:
diff --git a/configure b/configure
index 90fb6f07ca..7612a6052c 100755
--- a/configure
+++ b/configure
@@ -213,6 +213,7 @@ External library support:
--enable-libschroedinger Dirac video encoding/decoding
--enable-libsnappy snappy compression
--enable-libspeex Speex audio encoding/decoding
+ --enable-libsrt Haivision SRT protocol
--enable-libtheora Theora video encoding/decoding
--enable-libtwolame MP2 audio encoding
--enable-libvo-aacenc AAC audio encoding
@@ -1374,6 +1375,7 @@ EXTERNAL_LIBRARY_LIST="
libschroedinger
libsnappy
libspeex
+ libsrt
libtheora
libtwolame
libvorbis
@@ -2525,6 +2527,8 @@ librtmpt_protocol_deps="librtmp"
librtmpte_protocol_deps="librtmp"
mmsh_protocol_select="http_protocol"
mmst_protocol_select="network"
+libsrt_protocol_deps="libsrt"
+libsrt_protocol_select="network"
rtmp_protocol_conflict="librtmp_protocol"
rtmp_protocol_select="tcp_protocol"
rtmp_protocol_suggest="zlib"
@@ -4674,6 +4678,7 @@ enabled librtmp && require_pkg_config librtmp librtmp librtmp/rtmp.h R
enabled libschroedinger && require_pkg_config libschroedinger schroedinger-1.0 schroedinger/schro.h schro_init
enabled libsnappy && require libsnappy snappy-c.h snappy_compress -lsnappy
enabled libspeex && require_pkg_config libspeex speex speex/speex.h speex_decoder_init
+enabled libsrt && require_pkg_config libsrt "srt >= 1.2.0" srt/srt.h srt_socket
enabled libtheora && require libtheora theora/theoraenc.h th_info_init -ltheoraenc -ltheoradec -logg
enabled libtwolame && require libtwolame twolame.h twolame_init -ltwolame
enabled libvo_aacenc && require libvo_aacenc vo-aacenc/voAAC.h voGetAACEncAPI -lvo-aacenc
diff --git a/doc/protocols.texi b/doc/protocols.texi
index c136c74e41..e2d06a0675 100644
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -655,6 +655,146 @@ To play back the first stream announced on one the default IPv6 SAP multicast ad
avplay sap://[ff0e::2:7ffe]
@end example
+ at section srt
+
+Haivision Secure Reliable Transport Protocol via libsrt.
+
+The supported syntax for a SRT URL is:
+ at example
+srt://@var{hostname}:@var{port}[?@var{options}]
+ at end example
+
+ at var{options} contains a list of &-separated options of the form
+ at var{key}=@var{val}.
+
+or
+
+ at example
+ at var{options} srt://@var{hostname}:@var{port}
+ at end example
+
+ at var{options} contains a list of '- at var{key} @var{val}'
+options.
+
+This protocol accepts the following options.
+
+ at table @option
+ at item connect_timeout
+Connection timeout; SRT cannot connect for RTT > 1500 msec
+(2 handshake exchanges) with the default connect timeout of
+3 seconds. This option applies to the caller and rendezvous
+connection modes. The connect timeout is 10 times the value
+set for the rendezvous mode (which can be used as a
+workaround for this connection problem with earlier versions).
+
+ at item ffs=@var{bytes}
+Flight Flag Size (Window Size), in bytes. FFS is actually an
+internal parameter and you should set it to not less than
+ at option{recv_buffer_size} and @option{mss}. The default value
+is relatively large, therefore unless you set a very large receiver buffer,
+you do not need to change this option. Default value is 25600.
+
+ at item inputbw=@var{bytes/seconds}
+Sender nominal input rate, in bytes per seconds. Used along with
+ at option{oheadbw}, when @option{maxbw} is set to relative (0), to
+calculate maximum sending rate when recovery packets are sent
+along with the main media stream:
+ at option{inputbw} * (100 + @option{oheadbw}) / 100
+if @option{inputbw} is not set while @option{maxbw} is set to
+relative (0), the actual input rate is evaluated inside
+the library. Default value is 0.
+
+ at item iptos=@var{tos}
+IP Type of Service. Applies to sender only. Default value is 0xB8.
+
+ at item ipttl=@var{ttl}
+IP Time To Live. Applies to sender only. Default value is 64.
+
+ at item listen_timeout
+Set socket listen timeout.
+
+ at item maxbw=@var{bytes/seconds}
+Maximum sending bandwidth, in bytes per seconds.
+-1 infinite (CSRTCC limit is 30mbps)
+0 relative to input rate (see @option{inputbw})
+>0 absolute limit value
+Default value is 0 (relative)
+
+ at item mode=@var{caller|listener|rendezvous}
+Connection mode.
+ at option{caller} opens client connection.
+ at option{listener} starts server to listen for incoming connections.
+ at option{rendezvous} use Rendez-Vous connection mode.
+Default value is caller.
+
+ at item mss=@var{bytes}
+Maximum Segment Size, in bytes. Used for buffer allocation
+and rate calculation using a packet counter assuming fully
+filled packets. The smallest MSS between the peers is
+used. This is 1500 by default in the overall internet.
+This is the maximum size of the UDP packet and can be
+only decreased, unless you have some unusual dedicated
+network settings. Default value is 1500.
+
+ at item nakreport=@var{1|0}
+If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
+periodically until a lost packet is retransmitted or
+intentionally dropped. Default value is 1.
+
+ at item oheadbw=@var{percents}
+Recovery bandwidth overhead above input rate, in percents.
+See @option{inputbw}. Default value is 25%.
+
+ at item passphrase=@var{string}
+HaiCrypt Encryption/Decryption Passphrase string, length
+from 10 to 79 characters. The passphrase is the shared
+secret between the sender and the receiver. It is used
+to generate the Key Encrypting Key using PBKDF2
+(Password-Based Key Derivation Function). It is used
+only if @option{pbkeylen} is non-zero. It is used on
+the receiver only if the received data is encrypted.
+The configured passphrase cannot be recovered (write-only).
+
+ at item pbkeylen=@var{bytes}
+Sender encryption key length, in bytes.
+Only can be set to 0, 16, 24 and 32.
+Enable sender encryption if not 0.
+Not required on receiver (set to 0),
+key size obtained from sender in HaiCrypt handshake.
+Default value is 0.
+
+ at item recv_buffer_size=@var{bytes}
+Set receive buffer size, expressed in bytes.
+
+ at item send_buffer_size=@var{bytes}
+Set send buffer size, expressed in bytes.
+
+ at item rw_timeout
+Set raise error timeout for read/write optations.
+
+This option is only relevant in read mode:
+if no data arrived in more than this time
+interval, raise error.
+
+ at item tlpktdrop=@var{1|0}
+Too-late Packet Drop. When enabled on receiver, it skips
+missing packets that have not been delivered in time and
+delivers the following packets to the application when
+their time-to-play has come. It also sends a fake ACK to
+the sender. When enabled on sender and enabled on the
+receiving peer, the sender drops the older packets that
+have no chance of being delivered in time. It was
+automatically enabled in the sender if the receiver
+supports it.
+
+ at item tsbpddelay
+Timestamp-based Packet Delivery Delay.
+Used to absorb burst of missed packet retransmission.
+
+ at end table
+
+For more information see: @url{https://github.com/Haivision/srt}.
+
@section tcp
Transmission Control Protocol.
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 2c1c0f6d7f..96085d20c6 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -414,6 +414,9 @@ OBJS-$(CONFIG_TLS_PROTOCOL) += tls.o $(TLS-OBJS-yes)
OBJS-$(CONFIG_UDP_PROTOCOL) += udp.o
OBJS-$(CONFIG_UNIX_PROTOCOL) += unix.o
+# external libraries
+OBJS-$(CONFIG_LIBSRT_PROTOCOL) += libsrt.o
+
SKIPHEADERS-$(CONFIG_FFRTMPCRYPT_PROTOCOL) += rtmpdh.h
SKIPHEADERS-$(CONFIG_NETWORK) += network.h rtsp.h
diff --git a/libavformat/libsrt.c b/libavformat/libsrt.c
new file mode 100644
index 0000000000..3e50dab64f
--- /dev/null
+++ b/libavformat/libsrt.c
@@ -0,0 +1,546 @@
+/*
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Haivision Open SRT (Secure Reliable Transport) protocol
+ */
+
+#include <srt/srt.h>
+
+#include "libavutil/avassert.h"
+#include "libavutil/opt.h"
+#include "libavutil/parseutils.h"
+#include "libavutil/time.h"
+
+#include "avformat.h"
+#include "internal.h"
+#include "network.h"
+#include "os_support.h"
+#include "url.h"
+
+enum SRTMode {
+ SRT_MODE_CALLER = 0,
+ SRT_MODE_LISTENER = 1,
+ SRT_MODE_RENDEZVOUS = 2
+};
+
+typedef struct SRTContext {
+ const AVClass *class;
+ int fd;
+ int eid;
+ int64_t rw_timeout;
+ int64_t listen_timeout;
+ int recv_buffer_size;
+ int send_buffer_size;
+
+ int64_t maxbw;
+ int pbkeylen;
+ char *passphrase;
+ int mss;
+ int ffs;
+ int ipttl;
+ int iptos;
+ int64_t inputbw;
+ int oheadbw;
+ int64_t tsbpddelay;
+ int tlpktdrop;
+ int nakreport;
+ int64_t connect_timeout;
+ enum SRTMode mode;
+} SRTContext;
+
+#define D AV_OPT_FLAG_DECODING_PARAM
+#define E AV_OPT_FLAG_ENCODING_PARAM
+#define OFFSET(x) offsetof(SRTContext, x)
+static const AVOption libsrt_options[] = {
+ { "rw_timeout", "Timeout of socket I/O operations", OFFSET(rw_timeout), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
+ { "listen_timeout", "Connection awaiting timeout", OFFSET(listen_timeout), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
+ { "send_buffer_size", "Socket send buffer size (in bytes)", OFFSET(send_buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E },
+ { "recv_buffer_size", "Socket receive buffer size (in bytes)", OFFSET(recv_buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E },
+ { "maxbw", "Maximum bandwidth (bytes per second) that the connection can use", OFFSET(maxbw), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
+ { "pbkeylen", "Crypto key len in bytes {16,24,32} Default: 16 (128-bit)", OFFSET(pbkeylen), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 32, .flags = D|E },
+ { "passphrase", "Crypto PBKDF2 Passphrase size[0,10..64] 0:disable crypto", OFFSET(passphrase), AV_OPT_TYPE_STRING, { .str = NULL }, .flags = D|E },
+ { "mss", "The Maximum Segment Size", OFFSET(mss), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 1500, .flags = D|E },
+ { "ffs", "Flight flag size (window size) (in bytes)", OFFSET(ffs), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E },
+ { "ipttl", "IP Time To Live", OFFSET(ipttl), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 255, .flags = D|E },
+ { "iptos", "IP Type of Service", OFFSET(iptos), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 255, .flags = D|E },
+ { "inputbw", "Estimated input stream rate", OFFSET(inputbw), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
+ { "oheadbw", "MaxBW ceiling based on % over input stream rate", OFFSET(oheadbw), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 100, .flags = D|E },
+ { "tsbpddelay", "TsbPd receiver delay to absorb burst of missed packet retransmission", OFFSET(tsbpddelay), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
+ { "tlpktdrop", "Enable receiver pkt drop", OFFSET(tlpktdrop), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 1, .flags = D|E },
+ { "nakreport", "Enable receiver to send periodic NAK reports", OFFSET(nakreport), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 1, .flags = D|E },
+ { "connect_timeout", "Connect timeout. Caller default: 3000, rendezvous (x 10)", OFFSET(connect_timeout), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
+ { "mode", "Connection mode (caller, listener, rendezvous)", OFFSET(mode), AV_OPT_TYPE_INT, { .i64 = SRT_MODE_CALLER }, SRT_MODE_CALLER, SRT_MODE_RENDEZVOUS, .flags = D|E, "mode" },
+ { "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" },
+ { NULL }
+};
+
+static int libsrt_neterrno(URLContext *h)
+{
+ int err = srt_getlasterror(NULL);
+ av_log(h, AV_LOG_ERROR, "%s\n", srt_getlasterror_str());
+ if (err == SRT_EASYNCRCV)
+ return AVERROR(EAGAIN);
+ return AVERROR_UNKNOWN;
+}
+
+static int libsrt_socket_nonblock(int socket, int enable)
+{
+ int ret = srt_setsockopt(socket, 0, SRTO_SNDSYN, &enable, sizeof(enable));
+ if (ret < 0)
+ return ret;
+ return srt_setsockopt(socket, 0, SRTO_RCVSYN, &enable, sizeof(enable));
+}
+
+static int libsrt_network_wait_fd(URLContext *h, int eid, int fd, int write)
+{
+ int ret, len = 1;
+ int modes = write ? SRT_EPOLL_OUT : SRT_EPOLL_IN;
+ SRTSOCKET ready[1];
+
+ if (srt_epoll_add_usock(eid, fd, &modes) < 0)
+ return libsrt_neterrno(h);
+ if (write) {
+ ret = srt_epoll_wait(eid, 0, 0, ready, &len, POLLING_TIME, 0, 0, 0, 0);
+ } else {
+ ret = srt_epoll_wait(eid, ready, &len, 0, 0, POLLING_TIME, 0, 0, 0, 0);
+ }
+ if (ret < 0) {
+ if (srt_getlasterror(NULL) == SRT_ETIMEOUT)
+ ret = AVERROR(EAGAIN);
+ else
+ ret = libsrt_neterrno(h);
+ } else {
+ ret = 0;
+ }
+ if (srt_epoll_remove_usock(eid, fd) < 0)
+ return libsrt_neterrno(h);
+ return ret;
+}
+
+/* TODO de-duplicate code from ff_network_wait_fd_timeout() */
+
+static int libsrt_network_wait_fd_timeout(URLContext *h, int eid, int fd, int write, int64_t timeout, AVIOInterruptCB *int_cb)
+{
+ int ret;
+ int64_t wait_start = 0;
+
+ while (1) {
+ if (ff_check_interrupt(int_cb))
+ return AVERROR_EXIT;
+ ret = libsrt_network_wait_fd(h, eid, fd, write);
+ if (ret != AVERROR(EAGAIN))
+ return ret;
+ if (timeout > 0) {
+ if (!wait_start)
+ wait_start = av_gettime_relative();
+ else if (av_gettime_relative() - wait_start > timeout)
+ return AVERROR(ETIMEDOUT);
+ }
+ }
+}
+
+static int libsrt_listen(int eid, int fd, const struct sockaddr *addr, socklen_t addrlen, URLContext *h, int timeout)
+{
+ int ret;
+ int reuse = 1;
+ if (srt_setsockopt(fd, SOL_SOCKET, SRTO_REUSEADDR, &reuse, sizeof(reuse))) {
+ av_log(h, AV_LOG_WARNING, "setsockopt(SRTO_REUSEADDR) failed\n");
+ }
+ ret = srt_bind(fd, addr, addrlen);
+ if (ret)
+ return libsrt_neterrno(h);
+
+ ret = srt_listen(fd, 1);
+ if (ret)
+ return libsrt_neterrno(h);
+
+ while ((ret = libsrt_network_wait_fd_timeout(h, eid, fd, 1, timeout, &h->interrupt_callback))) {
+ switch (ret) {
+ case AVERROR(ETIMEDOUT):
+ continue;
+ default:
+ return ret;
+ }
+ }
+
+ ret = srt_accept(fd, NULL, NULL);
+ if (ret < 0)
+ return libsrt_neterrno(h);
+ if (libsrt_socket_nonblock(ret, 1) < 0)
+ av_log(h, AV_LOG_DEBUG, "libsrt_socket_nonblock failed\n");
+
+ return ret;
+}
+
+static int libsrt_listen_connect(int eid, int fd, const struct sockaddr *addr, socklen_t addrlen, int timeout, URLContext *h, int will_try_next)
+{
+ int ret;
+
+ if (libsrt_socket_nonblock(fd, 1) < 0)
+ av_log(h, AV_LOG_DEBUG, "ff_socket_nonblock failed\n");
+
+ while ((ret = srt_connect(fd, addr, addrlen))) {
+ ret = libsrt_neterrno(h);
+ switch (ret) {
+ case AVERROR(EINTR):
+ if (ff_check_interrupt(&h->interrupt_callback))
+ return AVERROR_EXIT;
+ continue;
+ case AVERROR(EINPROGRESS):
+ case AVERROR(EAGAIN):
+ ret = libsrt_network_wait_fd_timeout(h, eid, fd, 1, timeout, &h->interrupt_callback);
+ if (ret < 0)
+ return ret;
+ ret = srt_getlasterror(NULL);
+ srt_clearlasterror();
+ if (ret != 0) {
+ char buf[128];
+ ret = AVERROR(ret);
+ av_strerror(ret, buf, sizeof(buf));
+ if (will_try_next)
+ av_log(h, AV_LOG_WARNING,
+ "Connection to %s failed (%s), trying next address\n",
+ h->filename, buf);
+ else
+ av_log(h, AV_LOG_ERROR, "Connection to %s failed: %s\n",
+ h->filename, buf);
+ }
+ default:
+ return ret;
+ }
+ }
+ return ret;
+}
+
+static int libsrt_setsockopt(URLContext *h, int fd, SRT_SOCKOPT optname, const char * optnamestr, const void * optval, int optlen)
+{
+ if (srt_setsockopt(fd, 0, optname, optval, optlen) < 0) {
+ av_log(h, AV_LOG_ERROR, "failed to set option %s on socket: %s\n", optnamestr, srt_getlasterror_str());
+ return AVERROR(EIO);
+ }
+ return 0;
+}
+
+/* - The "POST" options can be altered any time on a connected socket.
+ They MAY have also some meaning when set prior to connecting; such
+ option is SRTO_RCVSYN, which makes connect/accept call asynchronous.
+ Because of that this option is treated special way in this app. */
+static int libsrt_set_options_post(URLContext *h, int fd)
+{
+ SRTContext *s = h->priv_data;
+
+ if ((s->inputbw >= 0 && libsrt_setsockopt(h, fd, SRTO_INPUTBW, "SRTO_INPUTBW", &s->inputbw, sizeof(s->inputbw)) < 0) ||
+ (s->oheadbw >= 0 && libsrt_setsockopt(h, fd, SRTO_OHEADBW, "SRTO_OHEADBW", &s->oheadbw, sizeof(s->oheadbw)) < 0)) {
+ return AVERROR(EIO);
+ }
+ return 0;
+}
+
+/* - The "PRE" options must be set prior to connecting and can't be altered
+ on a connected socket, however if set on a listening socket, they are
+ derived by accept-ed socket. */
+static int libsrt_set_options_pre(URLContext *h, int fd)
+{
+ SRTContext *s = h->priv_data;
+ int yes = 1;
+ int tsbpddelay = s->tsbpddelay / 1000;
+ 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->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, sizeof(s->passphrase)) < 0) ||
+ (s->mss >= 0 && libsrt_setsockopt(h, fd, SRTO_MSS, "SRTO_MMS", &s->mss, sizeof(s->mss)) < 0) ||
+ (s->ffs >= 0 && libsrt_setsockopt(h, fd, SRTO_FC, "SRTO_FC", &s->ffs, sizeof(s->ffs)) < 0) ||
+ (s->ipttl >= 0 && libsrt_setsockopt(h, fd, SRTO_IPTTL, "SRTO_UPTTL", &s->ipttl, sizeof(s->ipttl)) < 0) ||
+ (s->iptos >= 0 && libsrt_setsockopt(h, fd, SRTO_IPTOS, "SRTO_IPTOS", &s->iptos, sizeof(s->iptos)) < 0) ||
+ (tsbpddelay >= 0 && libsrt_setsockopt(h, fd, SRTO_TSBPDDELAY, "SRTO_TSBPDELAY", &tsbpddelay, sizeof(tsbpddelay)) < 0) ||
+ (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 )) {
+ return AVERROR(EIO);
+ }
+ return 0;
+}
+
+
+static int libsrt_setup(URLContext *h, const char *uri, int flags)
+{
+ struct addrinfo hints = { 0 }, *ai, *cur_ai;
+ int port, fd = -1;
+ SRTContext *s = h->priv_data;
+ const char *p;
+ char buf[256];
+ int ret;
+ char hostname[1024],proto[1024],path[1024];
+ char portstr[10];
+ int open_timeout = 5000000;
+ int eid;
+
+ eid = srt_epoll_create();
+ if (eid < 0)
+ return libsrt_neterrno(h);
+ s->eid = eid;
+
+ av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname),
+ &port, path, sizeof(path), uri);
+ if (strcmp(proto, "srt"))
+ return AVERROR(EINVAL);
+ if (port <= 0 || port >= 65536) {
+ av_log(h, AV_LOG_ERROR, "Port missing in uri\n");
+ return AVERROR(EINVAL);
+ }
+ p = strchr(uri, '?');
+ if (p) {
+ if (av_find_info_tag(buf, sizeof(buf), "timeout", p)) {
+ s->rw_timeout = strtol(buf, NULL, 10);
+ }
+ if (av_find_info_tag(buf, sizeof(buf), "listen_timeout", p)) {
+ s->listen_timeout = strtol(buf, NULL, 10);
+ }
+ }
+ if (s->rw_timeout >= 0) {
+ open_timeout = h->rw_timeout = s->rw_timeout;
+ }
+ hints.ai_family = AF_UNSPEC;
+ hints.ai_socktype = SOCK_DGRAM;
+ snprintf(portstr, sizeof(portstr), "%d", port);
+ if (s->mode == SRT_MODE_LISTENER)
+ hints.ai_flags |= AI_PASSIVE;
+ ret = getaddrinfo(hostname[0] ? hostname : NULL, portstr, &hints, &ai);
+ if (ret) {
+ av_log(h, AV_LOG_ERROR,
+ "Failed to resolve hostname %s: %s\n",
+ hostname, gai_strerror(ret));
+ return AVERROR(EIO);
+ }
+
+ cur_ai = ai;
+
+ restart:
+
+ fd = srt_socket(cur_ai->ai_family, cur_ai->ai_socktype, 0);
+ if (fd < 0) {
+ ret = libsrt_neterrno(h);
+ goto fail;
+ }
+
+ if ((ret = libsrt_set_options_pre(h, fd)) < 0) {
+ goto fail;
+ }
+
+ /* Set the socket's send or receive buffer sizes, if specified.
+ If unspecified or setting fails, system default is used. */
+ if (s->recv_buffer_size > 0) {
+ srt_setsockopt(fd, SOL_SOCKET, SRTO_UDP_RCVBUF, &s->recv_buffer_size, sizeof (s->recv_buffer_size));
+ }
+ if (s->send_buffer_size > 0) {
+ srt_setsockopt(fd, SOL_SOCKET, SRTO_UDP_SNDBUF, &s->send_buffer_size, sizeof (s->send_buffer_size));
+ }
+ if (s->mode == SRT_MODE_LISTENER) {
+ // multi-client
+ if ((ret = libsrt_listen(s->eid, fd, cur_ai->ai_addr, cur_ai->ai_addrlen, h, open_timeout / 1000)) < 0)
+ goto fail1;
+ fd = ret;
+ } else {
+ if (s->mode == SRT_MODE_RENDEZVOUS) {
+ ret = srt_bind(fd, cur_ai->ai_addr, cur_ai->ai_addrlen);
+ if (ret)
+ goto fail1;
+ }
+
+ if ((ret = libsrt_listen_connect(s->eid, fd, cur_ai->ai_addr, cur_ai->ai_addrlen,
+ open_timeout / 1000, h, !!cur_ai->ai_next)) < 0) {
+ if (ret == AVERROR_EXIT)
+ goto fail1;
+ else
+ goto fail;
+ }
+ }
+ if ((ret = libsrt_set_options_post(h, fd)) < 0) {
+ goto fail;
+ }
+
+ h->is_streamed = 1;
+ s->fd = fd;
+
+ freeaddrinfo(ai);
+ return 0;
+
+ fail:
+ if (cur_ai->ai_next) {
+ /* Retry with the next sockaddr */
+ cur_ai = cur_ai->ai_next;
+ if (fd >= 0)
+ srt_close(fd);
+ ret = 0;
+ goto restart;
+ }
+ fail1:
+ if (fd >= 0)
+ srt_close(fd);
+ freeaddrinfo(ai);
+ return ret;
+}
+
+static int libsrt_open(URLContext *h, const char *uri, int flags)
+{
+ SRTContext *s = h->priv_data;
+ const char * p;
+ char buf[256];
+
+ if (srt_startup() < 0) {
+ return AVERROR_UNKNOWN;
+ }
+
+ /* SRT options (srt/srt.h) */
+ p = strchr(uri, '?');
+ if (p) {
+ if (av_find_info_tag(buf, sizeof(buf), "maxbw", p)) {
+ s->maxbw = strtoll(buf, NULL, 0);
+ }
+ if (av_find_info_tag(buf, sizeof(buf), "pbkeylen", p)) {
+ s->pbkeylen = strtol(buf, NULL, 10);
+ }
+ if (av_find_info_tag(buf, sizeof(buf), "passphrase", p)) {
+ s->passphrase = av_strndup(buf, strlen(buf));
+ }
+ if (av_find_info_tag(buf, sizeof(buf), "mss", p)) {
+ s->mss = strtol(buf, NULL, 10);
+ }
+ if (av_find_info_tag(buf, sizeof(buf), "ffs", p)) {
+ s->ffs = strtol(buf, NULL, 10);
+ }
+ if (av_find_info_tag(buf, sizeof(buf), "ipttl", p)) {
+ s->ipttl = strtol(buf, NULL, 10);
+ }
+ if (av_find_info_tag(buf, sizeof(buf), "iptos", p)) {
+ s->iptos = strtol(buf, NULL, 10);
+ }
+ if (av_find_info_tag(buf, sizeof(buf), "inputbw", p)) {
+ s->inputbw = strtoll(buf, NULL, 10);
+ }
+ if (av_find_info_tag(buf, sizeof(buf), "oheadbw", p)) {
+ s->oheadbw = strtoll(buf, NULL, 10);
+ }
+ if (av_find_info_tag(buf, sizeof(buf), "tsbpddelay", p)) {
+ s->tsbpddelay = strtol(buf, NULL, 10);
+ }
+ if (av_find_info_tag(buf, sizeof(buf), "tlpktdrop", p)) {
+ s->tlpktdrop = strtol(buf, NULL, 10);
+ }
+ if (av_find_info_tag(buf, sizeof(buf), "nakreport", p)) {
+ s->nakreport = strtol(buf, NULL, 10);
+ }
+ if (av_find_info_tag(buf, sizeof(buf), "connect_timeout", p)) {
+ s->connect_timeout = strtol(buf, NULL, 10);
+ }
+ if (av_find_info_tag(buf, sizeof(buf), "mode", p)) {
+ if (!strcmp(buf, "caller")) {
+ s->mode = SRT_MODE_CALLER;
+ } else if (!strcmp(buf, "listener")) {
+ s->mode = SRT_MODE_LISTENER;
+ } else if (!strcmp(buf, "rendezvous")) {
+ s->mode = SRT_MODE_RENDEZVOUS;
+ } else {
+ return AVERROR(EIO);
+ }
+ }
+ }
+ return libsrt_setup(h, uri, flags);
+}
+
+static int libsrt_read(URLContext *h, uint8_t *buf, int size)
+{
+ SRTContext *s = h->priv_data;
+ int ret;
+
+ if (!(h->flags & AVIO_FLAG_NONBLOCK)) {
+ ret = libsrt_network_wait_fd_timeout(h, s->eid, s->fd, 0, h->rw_timeout, &h->interrupt_callback);
+ if (ret)
+ return ret;
+ }
+
+ ret = srt_recvmsg(s->fd, buf, size);
+ if (ret < 0) {
+ ret = libsrt_neterrno(h);
+ }
+
+ return ret;
+}
+
+static int libsrt_write(URLContext *h, const uint8_t *buf, int size)
+{
+ SRTContext *s = h->priv_data;
+ int ret;
+
+ if (!(h->flags & AVIO_FLAG_NONBLOCK)) {
+ ret = libsrt_network_wait_fd_timeout(h, s->eid, s->fd, 1, h->rw_timeout, &h->interrupt_callback);
+ if (ret)
+ return ret;
+ }
+
+ ret = srt_sendmsg(s->fd, buf, size, -1, 0);
+ if (ret < 0) {
+ ret = libsrt_neterrno(h);
+ }
+
+ return ret;
+}
+
+static int libsrt_close(URLContext *h)
+{
+ SRTContext *s = h->priv_data;
+
+ srt_close(s->fd);
+
+ srt_epoll_release(s->eid);
+
+ srt_cleanup();
+
+ return 0;
+}
+
+static int libsrt_get_file_handle(URLContext *h)
+{
+ SRTContext *s = h->priv_data;
+ return s->fd;
+}
+
+static const AVClass libsrt_class = {
+ .class_name = "libsrt",
+ .item_name = av_default_item_name,
+ .option = libsrt_options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+const URLProtocol ff_libsrt_protocol = {
+ .name = "srt",
+ .url_open = libsrt_open,
+ .url_read = libsrt_read,
+ .url_write = libsrt_write,
+ .url_close = libsrt_close,
+ .url_get_file_handle = libsrt_get_file_handle,
+ .priv_data_size = sizeof(SRTContext),
+ .flags = URL_PROTOCOL_FLAG_NETWORK,
+ .priv_data_class = &libsrt_class,
+};
diff --git a/libavformat/protocols.c b/libavformat/protocols.c
index 8ea5c0e757..15b9ed736d 100644
--- a/libavformat/protocols.c
+++ b/libavformat/protocols.c
@@ -56,6 +56,7 @@ extern const URLProtocol ff_librtmpe_protocol;
extern const URLProtocol ff_librtmps_protocol;
extern const URLProtocol ff_librtmpt_protocol;
extern const URLProtocol ff_librtmpte_protocol;
+extern const URLProtocol ff_libsrt_protocol;
#include "libavformat/protocol_list.c"
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