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

Nablet Developer sdk at nablet.com
Wed Dec 13 10:31:04 EET 2017


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

Signed-off-by: Nablet Developer <sdk at nablet.com>
---
 configure               |  10 +
 doc/protocols.texi      | 116 +++++++++
 libavformat/Makefile    |   1 +
 libavformat/opensrt.c   | 622 ++++++++++++++++++++++++++++++++++++++++++++++++
 libavformat/protocols.c |   1 +
 5 files changed, 750 insertions(+)
 create mode 100644 libavformat/opensrt.c

diff --git a/configure b/configure
index d5bbb5b..b26c60f 100755
--- a/configure
+++ b/configure
@@ -293,6 +293,7 @@ External library support:
   --enable-opengl          enable OpenGL rendering [no]
   --enable-openssl         enable openssl, needed for https support
                            if gnutls 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]
@@ -1638,6 +1639,7 @@ EXTERNAL_LIBRARY_LIST="
     mediacodec
     openal
     opengl
+    opensrt
 "
 
 HWACCEL_AUTODETECT_LIBRARY_LIST="
@@ -3145,6 +3147,8 @@ libsmbclient_protocol_deps="libsmbclient gplv3"
 libssh_protocol_deps="libssh"
 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"
@@ -5972,6 +5976,8 @@ 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           && check_lib srt srt/srt.h srt_socket -lsrt
+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 ||
@@ -6026,6 +6032,10 @@ if enabled decklink; then
     esac
 fi
 
+if enabled opensrt; then
+    opensrt_protocol_extralibs="$opensrt_protocol_extralibs -lsrt"
+fi
+
 enabled securetransport &&
     check_func SecIdentityCreate "-Wl,-framework,CoreFoundation -Wl,-framework,Security" &&
     check_lib securetransport "Security/SecureTransport.h Security/Security.h" "SSLCreateContext" "-Wl,-framework,CoreFoundation -Wl,-framework,Security" ||
diff --git a/doc/protocols.texi b/doc/protocols.texi
index 8661aea..358366e 100644
--- a/doc/protocols.texi
+++ b/doc/protocols.texi
@@ -752,6 +752,122 @@ Set the workgroup used for making connections. By default workgroup is not speci
 
 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 conntimeo=@var{milliseconds}
+Connection timeout, in milliseconds. 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 @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 @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=@var{milliseconds}
+Set listen timeout, expressed in milliseconds.
+
+ 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{0|1|2}
+Connection mode.
+0 (caller) opens client connection.
+1 (listener) starts server to listen for incoming connections.
+2 (rendezvous) use Rendez-Vous connection mode.
+Default valus is 0 (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 @option{pbkeylen} is non-zero.
+t 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=@var{microseconds}
+Set raise error timeout, expressed in microseconds.
+
+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=@var{milliseconds}
+Timestamp-based Packet Delivery Delay, in milliseconds.
+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 734b703..d78c7d3 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -599,6 +599,7 @@ 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..98742c1
--- /dev/null
+++ b/libavformat/opensrt.c
@@ -0,0 +1,622 @@
+/*
+ * 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 rw_timeout;
+    int 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;
+    int tsbpddelay;
+    int tlpktdrop;
+    int nakreport;
+    int conntimeo;
+    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 (in microseconds) of socket I/O operations",                      OFFSET(rw_timeout),       AV_OPT_TYPE_INT,    { .i64 = -1 }, -1, INT_MAX,   .flags = D|E },
+    { "listen_timeout", "Connection awaiting timeout (in milliseconds)",                               OFFSET(listen_timeout),   AV_OPT_TYPE_INT,    { .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 (mSec) to absorb burst of missed packet retransmission", OFFSET(tsbpddelay),       AV_OPT_TYPE_INT,    { .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 },
+    { "conntimeo",      "Connect timeout in msec. Ccaller default: 3000, rendezvous (x 10)",           OFFSET(conntimeo),        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 },
+    { "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(EINVAL);
+}
+
+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_poll(struct pollfd *fds, nfds_t nfds, int timeout)
+{
+    int eid, ret, len = 1;
+    int modes = fds[0].events;
+    SRTSOCKET ready[1];
+    eid = srt_epoll_create();
+    if (eid < 0)
+        return eid;
+    ret = srt_epoll_add_usock(eid, fds[0].fd, &modes);
+    if (ret < 0) {
+        srt_epoll_release(eid);
+        return ret;
+    }
+    if (fds[0].events & POLLOUT) {
+        ret = srt_epoll_wait(eid, 0, 0, ready, &len, timeout, 0, 0, 0, 0);
+    } else {
+        ret = srt_epoll_wait(eid, ready, &len, 0, 0, timeout, 0, 0, 0, 0);
+    }
+    if (ret > 0) {
+        fds[0].revents = fds[0].events;
+    } else if (ret == 0) {
+        fds[0].revents = POLLERR;
+    } else {
+        if (srt_getlasterror(NULL) == SRT_ETIMEOUT)
+            ret = 0;
+    }
+    srt_epoll_release(eid);
+    return ret;
+}
+
+static int opensrt_network_wait_fd(int fd, int write)
+{
+    int ev = write ? POLLOUT : POLLIN;
+    struct pollfd p = { .fd = fd, .events = ev, .revents = 0 };
+    int ret;
+    ret = opensrt_poll(&p, 1, POLLING_TIME);
+    return ret < 0 ? opensrt_neterrno() : p.revents & (ev | POLLERR | POLLHUP) ? 0 : AVERROR(EAGAIN);
+}
+
+static int opensrt_network_wait_fd_timeout(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(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_poll_interrupt(struct pollfd *p, nfds_t nfds, int timeout, AVIOInterruptCB *cb)
+{
+    int runs = timeout / POLLING_TIME;
+    int ret = 0;
+
+    do {
+        if (ff_check_interrupt(cb))
+            return AVERROR_EXIT;
+        ret = opensrt_poll(p, nfds, POLLING_TIME);
+        if (ret != 0)
+            break;
+    } while (timeout <= 0 || runs-- > 0);
+
+    if (!ret)
+        return AVERROR(ETIMEDOUT);
+    if (ret < 0)
+        return opensrt_neterrno();
+    return ret;
+}
+
+static int opensrt_do_accept(int fd, int timeout, URLContext *h)
+{
+    int ret;
+    struct pollfd lp = { fd, POLLIN, 0 };
+
+    ret = opensrt_poll_interrupt(&lp, 1, 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(NULL, AV_LOG_DEBUG, "opensrt_socket_nonblock failed\n");
+
+    return ret;
+}
+
+static int opensrt_listen(int fd, const struct sockaddr *addr, socklen_t addrlen)
+{
+    int ret;
+    int reuse = 1;
+    if (srt_setsockopt(fd, SOL_SOCKET, SRTO_REUSEADDR, &reuse, sizeof(reuse))) {
+        av_log(NULL, 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 fd, const struct sockaddr *addr, socklen_t addrlen, int timeout, URLContext *h, int will_try_next)
+{
+    struct pollfd p = {fd, POLLOUT, 0};
+    int ret;
+    socklen_t optlen;
+
+    if (opensrt_socket_nonblock(fd, 1) < 0)
+        av_log(NULL, 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_poll_interrupt(&p, 1, timeout, &h->interrupt_callback);
+            if (ret < 0)
+                return ret;
+            optlen = sizeof(ret);
+            ret = srt_getlasterror(NULL);
+            srt_clearlasterror();
+            if (ret != 0) {
+                char errbuf[100];
+                ret = AVERROR(ret);
+                av_strerror(ret, errbuf, sizeof(errbuf));
+                if (will_try_next)
+                    av_log(h, AV_LOG_WARNING,
+                           "Connection to %s failed (%s), trying next address\n",
+                           h->filename, errbuf);
+                else
+                    av_log(h, AV_LOG_ERROR, "Connection to %s failed: %s\n",
+                           h->filename, errbuf);
+            }
+        default:
+            return ret;
+        }
+    }
+    return ret;
+}
+
+/* - 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 && srt_setsockopt(fd, 0, SRTO_INPUTBW, &s->inputbw, sizeof(s->inputbw)) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option SRTO_INPUTBW on socket: %s", srt_getlasterror_str());
+        return AVERROR(EIO);
+    }
+    if (s->oheadbw >= 0 && srt_setsockopt(fd, 0, SRTO_OHEADBW, &s->oheadbw, sizeof(s->oheadbw)) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option SRTO_OHEADBW on socket: %s", srt_getlasterror_str());
+        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;
+
+    if (s->mode == SRT_MODE_RENDEZVOUS && srt_setsockopt(fd, 0, SRTO_RENDEZVOUS, &yes, sizeof(yes)) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option SRTO_RENDEZVOUS on socket: %s", srt_getlasterror_str());
+        return AVERROR(EIO);
+    }
+    if (s->maxbw >= 0 && srt_setsockopt(fd, 0, SRTO_MAXBW, &s->maxbw, sizeof(s->maxbw)) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option SRTO_MAXBW on socket: %s", srt_getlasterror_str());
+        return AVERROR(EIO);
+    }
+    if (s->pbkeylen >= 0 && srt_setsockopt(fd, 0, SRTO_PBKEYLEN, &s->pbkeylen, sizeof(s->pbkeylen)) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option SRTO_PBKEYLEN on socket: %s", srt_getlasterror_str());
+        return AVERROR(EIO);
+    }
+    if (s->passphrase[0] && srt_setsockopt(fd, 0, SRTO_PASSPHRASE, &s->passphrase, sizeof(s->passphrase)) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option SRTO_PASSPHRASE on socket: %s", srt_getlasterror_str());
+        return AVERROR(EIO);
+    }
+    if (s->mss >= 0 && srt_setsockopt(fd, 0, SRTO_MSS, &s->mss, sizeof(s->mss)) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option SRTO_MSS on socket: %s", srt_getlasterror_str());
+        return AVERROR(EIO);
+    }
+    if (s->fc >= 0 && srt_setsockopt(fd, 0, SRTO_FC, &s->fc, sizeof(s->fc)) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option SRTO_FC on socket: %s", srt_getlasterror_str());
+        return AVERROR(EIO);
+    }
+    if (s->ipttl >= 0 && srt_setsockopt(fd, 0, SRTO_IPTTL, &s->ipttl, sizeof(s->ipttl)) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option SRTO_IPTTL on socket: %s", srt_getlasterror_str());
+        return AVERROR(EIO);
+    }
+    if (s->iptos >= 0 && srt_setsockopt(fd, 0, SRTO_IPTOS, &s->iptos, sizeof(s->iptos)) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option SRTO_IPTOS on socket: %s", srt_getlasterror_str());
+        return AVERROR(EIO);
+    }
+    if (s->tsbpddelay >= 0 && srt_setsockopt(fd, 0, SRTO_TSBPDDELAY, &s->tsbpddelay, sizeof(s->tsbpddelay)) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option SRTO_TSBPDDELAY on socket: %s", srt_getlasterror_str());
+        return AVERROR(EIO);
+    }
+    if (s->tlpktdrop >= 0 && srt_setsockopt(fd, 0, SRTO_TLPKTDROP, &s->tlpktdrop, sizeof(s->tlpktdrop)) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option SRTO_TLPKTDROP on socket: %s", srt_getlasterror_str());
+        return AVERROR(EIO);
+    }
+    if (s->nakreport >= 0 && srt_setsockopt(fd, 0, SRTO_NAKREPORT, &s->nakreport, sizeof(s->nakreport)) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option SRTO_NAKREPORT on socket: %s", srt_getlasterror_str());
+        return AVERROR(EIO);
+    }
+    if (s->conntimeo >= 0 && srt_setsockopt(fd, 0, SRTO_CONNTIMEO, &s->conntimeo, sizeof(s->conntimeo)) < 0) {
+        av_log(h, AV_LOG_ERROR, "failed to set option SRTO_CONNTIMEO on socket: %s", srt_getlasterror_str());
+        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;
+
+    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;
+    if (!hostname[0])
+        ret = getaddrinfo(NULL, portstr, &hints, &ai);
+    else
+        ret = getaddrinfo(hostname, 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)) < 0)
+            goto fail1;
+    } else {
+        if ((ret = opensrt_listen_connect(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];
+    int ret;
+
+    if (srt_startup() < 0) {
+        return AVERROR(EIO);
+    }
+
+    /* 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, 10);
+        }
+        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), "conntimeo", p)) {
+            s->conntimeo = 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;
+            }
+        }
+    }
+    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->fd, sc->listen_timeout, 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->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->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_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;
-- 
2.7.4



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