[FFmpeg-devel] [PATCH] avformat/opensrt: add Haivision Open SRT protocol

Sven Dueking sven at nablet.com
Wed Feb 21 11:16:48 EET 2018


protocol requires libsrt (https://github.com/Haivision/srt) to be installed

Signed-off-by: Sven Dueking <sven.dueking at 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/}.
 
+ at section srt
+
+Haivision Secure Reliable Transport Protocol via libsrt.
+
+The required 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}.
+
+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 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 
+ 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 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 
+ctual 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.
+caller opens client connection.
+listener starts server to listen for incoming connections.
+rendezvous use Rendez-Vous connection mode.
+Default valus is caller.
+
+ at 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.
+
+ at 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.
+
+ 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 Deriviation Function). It is used only if 
+ at 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).
+
+ 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 bytes.
+
+ at item send_buffer_size=@var{bytes}
+Set send buffer size, expressed bytes.
+
+ at 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.
+
+ 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 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.
+
+ 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 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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