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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00

Merge commit 'a2fc8dbae85339d1b418d296f2982b6c04c53c57'

* commit 'a2fc8dbae85339d1b418d296f2982b6c04c53c57':
  Add Haivision SRT protocol

Merged-by: James Almer <jamrial@gmail.com>
This commit is contained in:
James Almer 2018-03-29 21:56:19 -03:00
commit a123e576a4
6 changed files with 694 additions and 0 deletions

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@ -51,6 +51,7 @@ version <next>:
- AV1 Support through libaom
- E-AC-3 dependent frames support
- bitstream filter for extracting E-AC-3 core
- Haivision SRT protocol via libsrt
version 3.4:

5
configure vendored
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@ -257,6 +257,7 @@ External library support:
--enable-libsnappy enable Snappy compression, needed for hap encoding [no]
--enable-libsoxr enable Include libsoxr resampling [no]
--enable-libspeex enable Speex de/encoding via libspeex [no]
--enable-libsrt enable Haivision SRT protocol via libsrt [no]
--enable-libssh enable SFTP protocol via libssh [no]
--enable-libtesseract enable Tesseract, needed for ocr filter [no]
--enable-libtheora enable Theora encoding via libtheora [no]
@ -1705,6 +1706,7 @@ EXTERNAL_LIBRARY_LIST="
libsnappy
libsoxr
libspeex
libsrt
libssh
libtesseract
libtheora
@ -3246,6 +3248,8 @@ libssh_protocol_deps="libssh"
libtls_conflict="openssl gnutls"
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"
@ -6021,6 +6025,7 @@ enabled libsnappy && require libsnappy snappy-c.h snappy_compress -lsnap
enabled libsoxr && require libsoxr soxr.h soxr_create -lsoxr
enabled libssh && require_pkg_config libssh libssh libssh/sftp.h sftp_init
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 libtesseract && require_pkg_config libtesseract tesseract tesseract/capi.h TessBaseAPICreate
enabled libtheora && require libtheora theora/theoraenc.h th_info_init -ltheoraenc -ltheoradec -logg
enabled libtls && require_pkg_config libtls libtls tls.h tls_configure

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@ -1155,6 +1155,146 @@ If set to any value, listen for an incoming connection. Outgoing connection is d
Set the maximum number of streams. By default no limit is set.
@end table
@section srt
Haivision Secure Reliable Transport Protocol via libsrt.
The supported 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}.
or
@example
@var{options} srt://@var{hostname}:@var{port}
@end example
@var{options} contains a list of '-@var{key} @var{val}'
options.
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 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
@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 the 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 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.
@option{caller} opens client connection.
@option{listener} starts server to listen for incoming connections.
@option{rendezvous} use Rendez-Vous connection mode.
Default value is caller.
@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.
@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.
@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 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).
@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 in bytes.
@item send_buffer_size=@var{bytes}
Set send buffer size, expressed in bytes.
@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.
@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.
@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 srtp
Secure Real-time Transport Protocol.

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@ -566,6 +566,7 @@ OBJS-$(CONFIG_LIBRTMPE_PROTOCOL) += librtmp.o
OBJS-$(CONFIG_LIBRTMPS_PROTOCOL) += librtmp.o
OBJS-$(CONFIG_LIBRTMPT_PROTOCOL) += librtmp.o
OBJS-$(CONFIG_LIBRTMPTE_PROTOCOL) += librtmp.o
OBJS-$(CONFIG_LIBSRT_PROTOCOL) += libsrt.o
OBJS-$(CONFIG_LIBSSH_PROTOCOL) += libssh.o
OBJS-$(CONFIG_LIBSMBCLIENT_PROTOCOL) += libsmbclient.o

546
libavformat/libsrt.c Normal file
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@ -0,0 +1,546 @@
/*
* 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 <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,
};

View File

@ -65,6 +65,7 @@ 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;
extern const URLProtocol ff_libssh_protocol;
extern const URLProtocol ff_libsmbclient_protocol;