protocol requires libsrt (https://github.com/Haivision/srt) to be installed
Signed-off-by: Sven Dueking <sven.duek...@nablet.com>
---
MAINTAINERS | 1 +
configure | 5 +
doc/protocols.texi | 134 ++++++++++-
libavformat/Makefile | 1 +
libavformat/opensrt.c | 589
++++++++++++++++++++++++++++++++++++++++++++++++
libavformat/protocols.c | 1 +
6 files changed, 730 insertions(+), 1 deletion(-) create mode 100644
libavformat/opensrt.c
diff --git a/MAINTAINERS b/MAINTAINERS
index b691bd5..3e0355a 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -499,6 +499,7 @@ Protocols:
http.c Ronald S. Bultje
libssh.c Lukasz Marek
mms*.c Ronald S. Bultje
+ opensrt.c sven Dueking
udp.c Luca Abeni
icecast.c Marvin Scholz
diff --git a/configure b/configure
index 013308c..9a78bae 100755
--- a/configure
+++ b/configure
@@ -294,6 +294,7 @@ External library support:
--enable-opengl enable OpenGL rendering [no]
--enable-openssl enable openssl, needed for https support
if gnutls or libtls is not used [no]
+ --enable-opensrt enable Haivision Open SRT protocol [no]
--disable-sndio disable sndio support [autodetect]
--disable-schannel disable SChannel SSP, needed for TLS support on
Windows if openssl and gnutls are not used
[autodetect] @@ -1648,6 +1649,7 @@ EXTERNAL_LIBRARY_LIST="
mediacodec
openal
opengl
+ opensrt
"
HWACCEL_AUTODETECT_LIBRARY_LIST="
@@ -3157,6 +3159,8 @@ libssh_protocol_deps="libssh"
libtls_conflict="openssl gnutls"
mmsh_protocol_select="http_protocol"
mmst_protocol_select="network"
+opensrt_protocol_select="network"
+opensrt_protocol_deps="opensrt"
rtmp_protocol_conflict="librtmp_protocol"
rtmp_protocol_select="tcp_protocol"
rtmp_protocol_suggest="zlib"
@@ -6028,6 +6032,7 @@ enabled omx && require_header OMX_Core.h
enabled omx_rpi && { check_header OMX_Core.h ||
{ ! enabled cross_compile && add_cflags
-isystem/opt/vc/include/IL && check_header OMX_Core.h ; } ||
die "ERROR: OpenMAX IL headers not found"; }
&& enable omx
+enabled opensrt && require_pkg_config libsrt "srt >= 1.2.0"
srt/srt.h srt_socket
enabled openssl && { check_pkg_config openssl openssl
openssl/ssl.h OPENSSL_init_ssl ||
check_pkg_config openssl openssl
openssl/ssl.h SSL_library_init ||
check_lib openssl openssl/ssl.h
SSL_library_init -lssl -lcrypto || diff --git a/doc/protocols.texi
b/doc/protocols.texi index c24dc74..1d49eaa 100644
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -752,12 +752,144 @@ Truncate existing files on write, if set to 1. A
value of 0 prevents truncating. Default value is 1.
@item workgroup
-Set the workgroup used for making connections. By default workgroup is not
specified.
+Set the workgroup used for making connections. By default workgroup is
+not specified.
@end table
For more information see: @url{http://www.samba.org/}.
+@section srt
+
+Haivision Secure Reliable Transport Protocol via libsrt.
+
+The required syntax for a SRT url is:
+@example
+srt://@var{hostname}:@var{port}[?@var{options}]
+@end example
+
+@var{options} contains a list of &-separated options of the form
+@var{key}=@var{val}.
+
+This protocol accepts the following options.
+
+@table @option
+@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).
+
+@item fc=@var{bytes}
+Flight Flag Size (Window Size), in bytes. FC is actually an internal
+parameter and you should set it to not less than
+@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.
+
+@item inputbw=@var{bytes/seconds}
+Sender nominal input rate, in bytes per seconds. Used along with
+@option{oheadbw}, when @option{maxbw} is set to relative (0), to
+calculate maximum sending rate when recovery packets are sent along
+with main media stream:
+@option{inputbw} * (100 + @option{oheadbw}) / 100 if @option{inputbw}
+is not set while @option{maxbw} is set to relative (0), the actual
+ctual input rate is evaluated inside the library. Default value is 0.
+
+@item iptos=@var{tos}
+IP Type of Service. Applies to sender only. Default value is 0xB8.
+
+@item ipttl=@var{ttl}
+IP Time To Live. Applies to sender only. Default value is 64.
+
+@item listen_timeout
+Set socket listen timeout.
+
+@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)
+
+@item mode=@var{caller|listener|rendezvous}
+Connection mode.
+caller opens client connection.
+listener starts server to listen for incoming connections.
+rendezvous use Rendez-Vous connection mode.
+Default valus is caller.
+
+@item mss=@var{bytes}
+Maximum Segment Size, in bytes. Used for buffer allocation and rate
+calculation using 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.
+
+@item nakreport=@var{1|0}
+If set to 1, Receiver will send `UMSG_LOSSREPORT` messages periodically
+until the lost packet is retransmitted or intentionally dropped.
+Default value is 1.
+
+@item oheadbw=@var{percents}
+Recovery bandwidth overhead above input rate, in percents.
+See @option{inputbw}. Default value is 25%.
+
+@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 Deriviation 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 get back (write-only).
+
+@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.
+
+@item recv_buffer_size=@var{bytes}
+Set receive buffer size, expressed bytes.
+
+@item send_buffer_size=@var{bytes}
+Set send buffer size, expressed bytes.
+
+@item timeout
+Set raise error timeout.
+
+This option is only relevant in read mode:
+if no data arrived in more than this time interval, raise error.
+
+@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 deliver the following
+packets to the application when their time-to-play has come. It also
+send a fake ACK to sender. When enabled on sender and enabled on the
+receiving peer, sender drops the older packets that have no chance to
+be delivered in time. It was automatically enabled in sender if
+receiver supports it.
+
+@item tsbpddelay
+Timestamp-based Packet Delivery Delay.
+Used to absorb burst of missed packet retransmission.
+
+@end table
+
+For more information see: @url{https://github.com/Haivision/srt}.
+
+
@section libssh
Secure File Transfer Protocol via libssh diff --git a/libavformat/Makefile
b/libavformat/Makefile index 7ac1ba9..46ea43f 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -606,6 +606,7 @@ TLS-OBJS-$(CONFIG_SCHANNEL) +=
tls_schannel.o
OBJS-$(CONFIG_TLS_PROTOCOL) += tls.o $(TLS-OBJS-yes)
OBJS-$(CONFIG_UDP_PROTOCOL) += udp.o
OBJS-$(CONFIG_UDPLITE_PROTOCOL) += udp.o
+OBJS-$(CONFIG_OPENSRT_PROTOCOL) += opensrt.o
OBJS-$(CONFIG_UNIX_PROTOCOL) += unix.o
# libavdevice dependencies
diff --git a/libavformat/opensrt.c b/libavformat/opensrt.c new file mode
100644 index 0000000..3836ef7
--- /dev/null
+++ b/libavformat/opensrt.c
@@ -0,0 +1,589 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg 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.
+ *
+ * FFmpeg 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 FFmpeg; 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 "avformat.h"
+#include "libavutil/avassert.h"
+#include "libavutil/parseutils.h"
+#include "libavutil/opt.h"
+#include "libavutil/time.h"
+
+#include "internal.h"
+#include "network.h"
+#include "os_support.h"
+#include "url.h"
+#if HAVE_POLL_H
+#include <poll.h>
+#endif
+
+#if CONFIG_OPENSRT_PROTOCOL
+#include <srt/srt.h>
+#endif
+
+enum SRTMode {
+ SRT_MODE_CALLER = 0,
+ SRT_MODE_LISTENER = 1,
+ SRT_MODE_RENDEZVOUS = 2
+};
+
+typedef struct SRTContext {
+ 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 fc;
+ 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
+opensrt_options[] = {
+ { "timeout", "set timeout of socket I/O operations",
OFFSET(rw_timeout), AV_OPT_TYPE_DURATION, { .i64 = -1 }, -1, INT_MAX,
.flags = D|E },
+ { "listen_timeout", "Connection awaiting timeout",
OFFSET(listen_timeout), AV_OPT_TYPE_DURATION, { .i64 = -1 }, -1, INT_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 Transfer Unit",
OFFSET(mss), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 1500,
.flags = D|E },
+ { "fc", "Flight flag size (window size) (in bytes)",
OFFSET(fc), 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_DURATION, {
.i64 = -1 }, -1, INT_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. Ccaller default: 3000,
rendezvous (x 10)", OFFSET(connect_timeout),
AV_OPT_TYPE_DURATION, { .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 },
+ { "caller", NULL, 0, AV_OPT_TYPE_CONST, { .i64 =
SRT_MODE_CALLER }, INT_MIN, INT_MAX, .flags = D|E },
+ { "listener", NULL, 0, AV_OPT_TYPE_CONST, { .i64 =
SRT_MODE_LISTENER }, INT_MIN, INT_MAX, .flags = D|E },
+ { "rendezvous", NULL, 0, AV_OPT_TYPE_CONST, { .i64 =
SRT_MODE_RENDEZVOUS }, INT_MIN, INT_MAX, .flags = D|E },
+ { NULL }
+};
+
+static const AVClass opensrt_class = {
+ .class_name = "opensrt",
+ .item_name = av_default_item_name,
+ .option = opensrt_options,
+ .version = LIBAVUTIL_VERSION_INT,
+};
+
+static int opensrt_neterrno(void)
+{
+ int err = srt_getlasterror(NULL);
+ if (err == SRT_EASYNCRCV)
+ return AVERROR(EAGAIN);
+ return AVERROR_EXTERNAL;
+}
+
+static int opensrt_socket_nonblock(int socket, int enable) {
+ int ret = srt_setsockopt(socket, 0, SRTO_SNDSYN, &enable,
sizeof(enable));
+ if (ret < 0)
+ return ret;
+ ret = srt_setsockopt(socket, 0, SRTO_RCVSYN, &enable, sizeof(enable));
+ return ret;
+}
+
+static int opensrt_network_wait_fd(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 opensrt_neterrno();
+ 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 = opensrt_neterrno();
+ } else {
+ ret = 0;
+ }
+ if (srt_epoll_remove_usock(eid, fd) < 0)
+ return opensrt_neterrno();
+ return ret;
+}
+
+/* TODO de-duplicate code from ff_network_wait_fd_timeout() */
+
+static int opensrt_network_wait_fd_timeout(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 = opensrt_network_wait_fd(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 opensrt_do_accept(int eid, int fd, int timeout, URLContext
+*h) {
+ int ret;
+
+ ret = opensrt_network_wait_fd_timeout(eid, fd, 0, timeout,
&h->interrupt_callback);
+ if (ret < 0)
+ return ret;
+
+ ret = srt_accept(fd, NULL, NULL);
+ if (ret < 0)
+ return opensrt_neterrno();
+ if (opensrt_socket_nonblock(ret, 1) < 0)
+ av_log(h, AV_LOG_DEBUG, "opensrt_socket_nonblock failed\n");
+
+ return ret;
+}
+
+static int opensrt_listen(int fd, const struct sockaddr *addr,
+socklen_t addrlen, URLContext *h) {
+ 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 opensrt_neterrno();
+
+ ret = srt_listen(fd, 1);
+ if (ret)
+ return opensrt_neterrno();
+ return ret;
+}
+
+static int opensrt_listen_connect(int eid, int fd, const struct
+sockaddr *addr, socklen_t addrlen, int timeout, URLContext *h, int
will_try_next) {
+ int ret;
+
+ if (opensrt_socket_nonblock(fd, 1) < 0)
+ av_log(h, AV_LOG_DEBUG, "ff_socket_nonblock failed\n");
+
+ while ((ret = srt_connect(fd, addr, addrlen))) {
+ ret = opensrt_neterrno();
+ switch (ret) {
+ case AVERROR(EINTR):
+ if (ff_check_interrupt(&h->interrupt_callback))
+ return AVERROR_EXIT;
+ continue;
+ case AVERROR(EINPROGRESS):
+ case AVERROR(EAGAIN):
+ ret = opensrt_network_wait_fd_timeout(eid, fd, 1, timeout,
&h->interrupt_callback);
+ if (ret < 0)
+ return ret;
+ ret = srt_getlasterror(NULL);
+ srt_clearlasterror();
+ if (ret != 0) {
+ ret = AVERROR(ret);
+ if (will_try_next)
+ av_log(h, AV_LOG_WARNING,
+ "Connection to %s failed (%s), trying next
address\n",
+ h->filename, av_err2str(ret));
+ else
+ av_log(h, AV_LOG_ERROR, "Connection to %s failed:
%s\n",
+ h->filename, av_err2str(ret));
+ }
+ default:
+ return ret;
+ }
+ }
+ return ret;
+}
+
+static int opensrt_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 opensrt_set_options_post(URLContext *h, int fd) {
+ SRTContext *s = h->priv_data;
+
+ if (s->inputbw >= 0 && opensrt_setsockopt(h, fd, SRTO_INPUTBW,
"SRTO_INPUTBW", &s->inputbw, sizeof(s->inputbw)) < 0) {
+ return AVERROR(EIO);
+ }
+ if (s->oheadbw >= 0 && opensrt_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 opensrt_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 && opensrt_setsockopt(h, fd,
SRTO_RENDEZVOUS, "SRTO_RENDEZVOUS", &yes, sizeof(yes)) < 0) {
+ return AVERROR(EIO);
+ }
+ if (s->maxbw >= 0 && opensrt_setsockopt(h, fd, SRTO_MAXBW,
"SRTO_MAXBW", &s->maxbw, sizeof(s->maxbw)) < 0) {
+ return AVERROR(EIO);
+ }
+ if (s->pbkeylen >= 0 && opensrt_setsockopt(h, fd, SRTO_PBKEYLEN,
"SRTO_PBKEYLEN", &s->pbkeylen, sizeof(s->pbkeylen)) < 0) {
+ return AVERROR(EIO);
+ }
+ if (s->passphrase[0] && opensrt_setsockopt(h, fd, SRTO_PASSPHRASE,
"SRTO_PASSPHRASE", &s->passphrase, sizeof(s->passphrase)) < 0) {
+ return AVERROR(EIO);
+ }
+ if (s->mss >= 0 && opensrt_setsockopt(h, fd, SRTO_MSS, "SRTO_MMS",
&s->mss, sizeof(s->mss)) < 0) {
+ return AVERROR(EIO);
+ }
+ if (s->fc >= 0 && opensrt_setsockopt(h, fd, SRTO_FC, "SRTO_FC", &s->fc,
sizeof(s->fc)) < 0) {
+ return AVERROR(EIO);
+ }
+ if (s->ipttl >= 0 && opensrt_setsockopt(h, fd, SRTO_IPTTL,
"SRTO_UPTTL", &s->ipttl, sizeof(s->ipttl)) < 0) {
+ return AVERROR(EIO);
+ }
+ if (s->iptos >= 0 && opensrt_setsockopt(h, fd, SRTO_IPTOS,
"SRTO_IPTOS", &s->iptos, sizeof(s->iptos)) < 0) {
+ return AVERROR(EIO);
+ }
+ if (tsbpddelay >= 0 && opensrt_setsockopt(h, fd, SRTO_TSBPDDELAY,
"SRTO_TSBPDELAY", &tsbpddelay, sizeof(tsbpddelay)) < 0) {
+ return AVERROR(EIO);
+ }
+ if (s->tlpktdrop >= 0 && opensrt_setsockopt(h, fd, SRTO_TLPKTDROP,
"SRTO_TLPKDROP", &s->tlpktdrop, sizeof(s->tlpktdrop)) < 0) {
+ return AVERROR(EIO);
+ }
+ if (s->nakreport >= 0 && opensrt_setsockopt(h, fd, SRTO_NAKREPORT,
"SRTO_NAKREPORT", &s->nakreport, sizeof(s->nakreport)) < 0) {
+ return AVERROR(EIO);
+ }
+ if (connect_timeout >= 0 && opensrt_setsockopt(h, fd, SRTO_CONNTIMEO,
"SRTO_CONNTIMEO", &connect_timeout, sizeof(connect_timeout)) < 0) {
+ return AVERROR(EIO);
+ }
+ return 0;
+}
+
+
+static int opensrt_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 opensrt_neterrno();
+ 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_STREAM;
+ 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 = opensrt_neterrno();
+ goto fail;
+ }
+
+ if ((ret = opensrt_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 = opensrt_listen(fd, cur_ai->ai_addr, cur_ai->ai_addrlen,
h)) < 0)
+ goto fail1;
+ } else {
+ if ((ret = opensrt_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 = opensrt_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 opensrt_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_EXTERNAL;
+ }
+
+ /* 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), "fc", p)) {
+ s->fc = 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 opensrt_setup(h, uri, flags); }
+
+
+static int opensrt_accept(URLContext *s, URLContext **c) {
+ SRTContext *sc = s->priv_data;
+ SRTContext *cc;
+ int ret;
+ av_assert0(sc->mode == SRT_MODE_LISTENER);
+ if ((ret = ffurl_alloc(c, s->filename, s->flags,
&s->interrupt_callback)) < 0)
+ return ret;
+ cc = (*c)->priv_data;
+ ret = opensrt_do_accept(sc->eid, sc->fd, sc->listen_timeout / 1000, s);
+ if (ret < 0)
+ return ret;
+ cc->fd = ret;
+ return 0;
+}
+
+static int opensrt_read(URLContext *h, uint8_t *buf, int size) {
+ SRTContext *s = h->priv_data;
+ int ret;
+
+ if (!(h->flags & AVIO_FLAG_NONBLOCK)) {
+ ret = opensrt_network_wait_fd_timeout(s->eid, s->fd, 0,
h->rw_timeout, &h->interrupt_callback);
+ if (ret)
+ return ret;
+ }
+ ret = srt_recvmsg(s->fd, buf, size);
+ return ret < 0 ? opensrt_neterrno() : ret; }
+
+static int opensrt_write(URLContext *h, const uint8_t *buf, int size) {
+ SRTContext *s = h->priv_data;
+ int ret;
+
+ if (!(h->flags & AVIO_FLAG_NONBLOCK)) {
+ ret = opensrt_network_wait_fd_timeout(s->eid, s->fd, 1,
h->rw_timeout, &h->interrupt_callback);
+ if (ret)
+ return ret;
+ }
+ ret = srt_sendmsg(s->fd, buf, size, -1, 0);
+ return ret < 0 ? opensrt_neterrno() : ret; }
+
+static int opensrt_close(URLContext *h) {
+ SRTContext *s = h->priv_data;
+
+ srt_close(s->fd);
+
+ srt_epoll_release(s->eid);
+
+ srt_cleanup();
+
+ return 0;
+}
+
+static int opensrt_get_file_handle(URLContext *h) {
+ SRTContext *s = h->priv_data;
+ return s->fd;
+}
+
+static int opensrt_get_window_size(URLContext *h) {
+ SRTContext *s = h->priv_data;
+ int avail;
+ socklen_t avail_len = sizeof(avail);
+
+ if (srt_getsockopt(s->fd, SOL_SOCKET, SRTO_UDP_RCVBUF, &avail,
&avail_len)) {
+ return opensrt_neterrno();
+ }
+ return avail;
+}
+
+const URLProtocol ff_opensrt_protocol = {
+ .name = "srt",
+ .url_open = opensrt_open,
+ .url_accept = opensrt_accept,
+ .url_read = opensrt_read,
+ .url_write = opensrt_write,
+ .url_close = opensrt_close,
+ .url_get_file_handle = opensrt_get_file_handle,
+ .url_get_short_seek = opensrt_get_window_size,
+ .priv_data_size = sizeof(SRTContext),
+ .flags = URL_PROTOCOL_FLAG_NETWORK,
+ .priv_data_class = &opensrt_class,
+};
diff --git a/libavformat/protocols.c b/libavformat/protocols.c index
669d74d..823349a 100644
--- a/libavformat/protocols.c
+++ b/libavformat/protocols.c
@@ -59,6 +59,7 @@ extern const URLProtocol ff_tcp_protocol; extern const
URLProtocol ff_tls_protocol; extern const URLProtocol ff_udp_protocol;
extern const URLProtocol ff_udplite_protocol;
+extern const URLProtocol ff_opensrt_protocol;
extern const URLProtocol ff_unix_protocol; extern const URLProtocol
ff_librtmp_protocol; extern const URLProtocol ff_librtmpe_protocol;
--
1.8.3.1
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