diff --git a/Changelog b/Changelog index 2869dce386..200866d873 100644 --- a/Changelog +++ b/Changelog @@ -51,6 +51,7 @@ version : - 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: diff --git a/configure b/configure index a92ac6acb7..99570a1415 100755 --- a/configure +++ b/configure @@ -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 diff --git a/doc/protocols.texi b/doc/protocols.texi index c24dc74505..e19504d073 100644 --- a/doc/protocols.texi +++ b/doc/protocols.texi @@ -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. diff --git a/libavformat/Makefile b/libavformat/Makefile index 39ec68c28b..af0823a7db 100644 --- a/libavformat/Makefile +++ b/libavformat/Makefile @@ -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 diff --git a/libavformat/libsrt.c b/libavformat/libsrt.c new file mode 100644 index 0000000000..0f9529d263 --- /dev/null +++ b/libavformat/libsrt.c @@ -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 + +#include "libavutil/avassert.h" +#include "libavutil/opt.h" +#include "libavutil/parseutils.h" +#include "libavutil/time.h" + +#include "avformat.h" +#include "internal.h" +#include "network.h" +#include "os_support.h" +#include "url.h" + +enum SRTMode { + SRT_MODE_CALLER = 0, + SRT_MODE_LISTENER = 1, + SRT_MODE_RENDEZVOUS = 2 +}; + +typedef struct SRTContext { + const AVClass *class; + int fd; + int eid; + int64_t rw_timeout; + int64_t listen_timeout; + int recv_buffer_size; + int send_buffer_size; + + int64_t maxbw; + int pbkeylen; + char *passphrase; + int mss; + int ffs; + int ipttl; + int iptos; + int64_t inputbw; + int oheadbw; + int64_t tsbpddelay; + int tlpktdrop; + int nakreport; + int64_t connect_timeout; + enum SRTMode mode; +} SRTContext; + +#define D AV_OPT_FLAG_DECODING_PARAM +#define E AV_OPT_FLAG_ENCODING_PARAM +#define OFFSET(x) offsetof(SRTContext, x) +static const AVOption libsrt_options[] = { + { "rw_timeout", "Timeout of socket I/O operations", OFFSET(rw_timeout), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E }, + { "listen_timeout", "Connection awaiting timeout", OFFSET(listen_timeout), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E }, + { "send_buffer_size", "Socket send buffer size (in bytes)", OFFSET(send_buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E }, + { "recv_buffer_size", "Socket receive buffer size (in bytes)", OFFSET(recv_buffer_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E }, + { "maxbw", "Maximum bandwidth (bytes per second) that the connection can use", OFFSET(maxbw), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E }, + { "pbkeylen", "Crypto key len in bytes {16,24,32} Default: 16 (128-bit)", OFFSET(pbkeylen), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 32, .flags = D|E }, + { "passphrase", "Crypto PBKDF2 Passphrase size[0,10..64] 0:disable crypto", OFFSET(passphrase), AV_OPT_TYPE_STRING, { .str = NULL }, .flags = D|E }, + { "mss", "The Maximum Segment Size", OFFSET(mss), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 1500, .flags = D|E }, + { "ffs", "Flight flag size (window size) (in bytes)", OFFSET(ffs), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E }, + { "ipttl", "IP Time To Live", OFFSET(ipttl), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 255, .flags = D|E }, + { "iptos", "IP Type of Service", OFFSET(iptos), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 255, .flags = D|E }, + { "inputbw", "Estimated input stream rate", OFFSET(inputbw), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E }, + { "oheadbw", "MaxBW ceiling based on % over input stream rate", OFFSET(oheadbw), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 100, .flags = D|E }, + { "tsbpddelay", "TsbPd receiver delay to absorb burst of missed packet retransmission", OFFSET(tsbpddelay), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E }, + { "tlpktdrop", "Enable receiver pkt drop", OFFSET(tlpktdrop), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 1, .flags = D|E }, + { "nakreport", "Enable receiver to send periodic NAK reports", OFFSET(nakreport), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 1, .flags = D|E }, + { "connect_timeout", "Connect timeout. Caller default: 3000, rendezvous (x 10)", OFFSET(connect_timeout), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E }, + { "mode", "Connection mode (caller, listener, rendezvous)", OFFSET(mode), AV_OPT_TYPE_INT, { .i64 = SRT_MODE_CALLER }, SRT_MODE_CALLER, SRT_MODE_RENDEZVOUS, .flags = D|E, "mode" }, + { "caller", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRT_MODE_CALLER }, INT_MIN, INT_MAX, .flags = D|E, "mode" }, + { "listener", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRT_MODE_LISTENER }, INT_MIN, INT_MAX, .flags = D|E, "mode" }, + { "rendezvous", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRT_MODE_RENDEZVOUS }, INT_MIN, INT_MAX, .flags = D|E, "mode" }, + { NULL } +}; + +static int libsrt_neterrno(URLContext *h) +{ + int err = srt_getlasterror(NULL); + av_log(h, AV_LOG_ERROR, "%s\n", srt_getlasterror_str()); + if (err == SRT_EASYNCRCV) + return AVERROR(EAGAIN); + return AVERROR_UNKNOWN; +} + +static int libsrt_socket_nonblock(int socket, int enable) +{ + int ret = srt_setsockopt(socket, 0, SRTO_SNDSYN, &enable, sizeof(enable)); + if (ret < 0) + return ret; + return srt_setsockopt(socket, 0, SRTO_RCVSYN, &enable, sizeof(enable)); +} + +static int libsrt_network_wait_fd(URLContext *h, int eid, int fd, int write) +{ + int ret, len = 1; + int modes = write ? SRT_EPOLL_OUT : SRT_EPOLL_IN; + SRTSOCKET ready[1]; + + if (srt_epoll_add_usock(eid, fd, &modes) < 0) + return libsrt_neterrno(h); + if (write) { + ret = srt_epoll_wait(eid, 0, 0, ready, &len, POLLING_TIME, 0, 0, 0, 0); + } else { + ret = srt_epoll_wait(eid, ready, &len, 0, 0, POLLING_TIME, 0, 0, 0, 0); + } + if (ret < 0) { + if (srt_getlasterror(NULL) == SRT_ETIMEOUT) + ret = AVERROR(EAGAIN); + else + ret = libsrt_neterrno(h); + } else { + ret = 0; + } + if (srt_epoll_remove_usock(eid, fd) < 0) + return libsrt_neterrno(h); + return ret; +} + +/* TODO de-duplicate code from ff_network_wait_fd_timeout() */ + +static int libsrt_network_wait_fd_timeout(URLContext *h, int eid, int fd, int write, int64_t timeout, AVIOInterruptCB *int_cb) +{ + int ret; + int64_t wait_start = 0; + + while (1) { + if (ff_check_interrupt(int_cb)) + return AVERROR_EXIT; + ret = libsrt_network_wait_fd(h, eid, fd, write); + if (ret != AVERROR(EAGAIN)) + return ret; + if (timeout > 0) { + if (!wait_start) + wait_start = av_gettime_relative(); + else if (av_gettime_relative() - wait_start > timeout) + return AVERROR(ETIMEDOUT); + } + } +} + +static int libsrt_listen(int eid, int fd, const struct sockaddr *addr, socklen_t addrlen, URLContext *h, int timeout) +{ + int ret; + int reuse = 1; + if (srt_setsockopt(fd, SOL_SOCKET, SRTO_REUSEADDR, &reuse, sizeof(reuse))) { + av_log(h, AV_LOG_WARNING, "setsockopt(SRTO_REUSEADDR) failed\n"); + } + ret = srt_bind(fd, addr, addrlen); + if (ret) + return libsrt_neterrno(h); + + ret = srt_listen(fd, 1); + if (ret) + return libsrt_neterrno(h); + + while ((ret = libsrt_network_wait_fd_timeout(h, eid, fd, 1, timeout, &h->interrupt_callback))) { + switch (ret) { + case AVERROR(ETIMEDOUT): + continue; + default: + return ret; + } + } + + ret = srt_accept(fd, NULL, NULL); + if (ret < 0) + return libsrt_neterrno(h); + if (libsrt_socket_nonblock(ret, 1) < 0) + av_log(h, AV_LOG_DEBUG, "libsrt_socket_nonblock failed\n"); + + return ret; +} + +static int libsrt_listen_connect(int eid, int fd, const struct sockaddr *addr, socklen_t addrlen, int timeout, URLContext *h, int will_try_next) +{ + int ret; + + if (libsrt_socket_nonblock(fd, 1) < 0) + av_log(h, AV_LOG_DEBUG, "ff_socket_nonblock failed\n"); + + while ((ret = srt_connect(fd, addr, addrlen))) { + ret = libsrt_neterrno(h); + switch (ret) { + case AVERROR(EINTR): + if (ff_check_interrupt(&h->interrupt_callback)) + return AVERROR_EXIT; + continue; + case AVERROR(EINPROGRESS): + case AVERROR(EAGAIN): + ret = libsrt_network_wait_fd_timeout(h, eid, fd, 1, timeout, &h->interrupt_callback); + if (ret < 0) + return ret; + ret = srt_getlasterror(NULL); + srt_clearlasterror(); + if (ret != 0) { + char buf[128]; + ret = AVERROR(ret); + av_strerror(ret, buf, sizeof(buf)); + if (will_try_next) + av_log(h, AV_LOG_WARNING, + "Connection to %s failed (%s), trying next address\n", + h->filename, buf); + else + av_log(h, AV_LOG_ERROR, "Connection to %s failed: %s\n", + h->filename, buf); + } + default: + return ret; + } + } + return ret; +} + +static int libsrt_setsockopt(URLContext *h, int fd, SRT_SOCKOPT optname, const char * optnamestr, const void * optval, int optlen) +{ + if (srt_setsockopt(fd, 0, optname, optval, optlen) < 0) { + av_log(h, AV_LOG_ERROR, "failed to set option %s on socket: %s\n", optnamestr, srt_getlasterror_str()); + return AVERROR(EIO); + } + return 0; +} + +/* - The "POST" options can be altered any time on a connected socket. + They MAY have also some meaning when set prior to connecting; such + option is SRTO_RCVSYN, which makes connect/accept call asynchronous. + Because of that this option is treated special way in this app. */ +static int libsrt_set_options_post(URLContext *h, int fd) +{ + SRTContext *s = h->priv_data; + + if ((s->inputbw >= 0 && libsrt_setsockopt(h, fd, SRTO_INPUTBW, "SRTO_INPUTBW", &s->inputbw, sizeof(s->inputbw)) < 0) || + (s->oheadbw >= 0 && libsrt_setsockopt(h, fd, SRTO_OHEADBW, "SRTO_OHEADBW", &s->oheadbw, sizeof(s->oheadbw)) < 0)) { + return AVERROR(EIO); + } + return 0; +} + +/* - The "PRE" options must be set prior to connecting and can't be altered + on a connected socket, however if set on a listening socket, they are + derived by accept-ed socket. */ +static int libsrt_set_options_pre(URLContext *h, int fd) +{ + SRTContext *s = h->priv_data; + int yes = 1; + int tsbpddelay = s->tsbpddelay / 1000; + int connect_timeout = s->connect_timeout; + + if ((s->mode == SRT_MODE_RENDEZVOUS && libsrt_setsockopt(h, fd, SRTO_RENDEZVOUS, "SRTO_RENDEZVOUS", &yes, sizeof(yes)) < 0) || + (s->maxbw >= 0 && libsrt_setsockopt(h, fd, SRTO_MAXBW, "SRTO_MAXBW", &s->maxbw, sizeof(s->maxbw)) < 0) || + (s->pbkeylen >= 0 && libsrt_setsockopt(h, fd, SRTO_PBKEYLEN, "SRTO_PBKEYLEN", &s->pbkeylen, sizeof(s->pbkeylen)) < 0) || + (s->passphrase && libsrt_setsockopt(h, fd, SRTO_PASSPHRASE, "SRTO_PASSPHRASE", &s->passphrase, sizeof(s->passphrase)) < 0) || + (s->mss >= 0 && libsrt_setsockopt(h, fd, SRTO_MSS, "SRTO_MMS", &s->mss, sizeof(s->mss)) < 0) || + (s->ffs >= 0 && libsrt_setsockopt(h, fd, SRTO_FC, "SRTO_FC", &s->ffs, sizeof(s->ffs)) < 0) || + (s->ipttl >= 0 && libsrt_setsockopt(h, fd, SRTO_IPTTL, "SRTO_UPTTL", &s->ipttl, sizeof(s->ipttl)) < 0) || + (s->iptos >= 0 && libsrt_setsockopt(h, fd, SRTO_IPTOS, "SRTO_IPTOS", &s->iptos, sizeof(s->iptos)) < 0) || + (tsbpddelay >= 0 && libsrt_setsockopt(h, fd, SRTO_TSBPDDELAY, "SRTO_TSBPDELAY", &tsbpddelay, sizeof(tsbpddelay)) < 0) || + (s->tlpktdrop >= 0 && libsrt_setsockopt(h, fd, SRTO_TLPKTDROP, "SRTO_TLPKDROP", &s->tlpktdrop, sizeof(s->tlpktdrop)) < 0) || + (s->nakreport >= 0 && libsrt_setsockopt(h, fd, SRTO_NAKREPORT, "SRTO_NAKREPORT", &s->nakreport, sizeof(s->nakreport)) < 0) || + (connect_timeout >= 0 && libsrt_setsockopt(h, fd, SRTO_CONNTIMEO, "SRTO_CONNTIMEO", &connect_timeout, sizeof(connect_timeout)) <0 )) { + return AVERROR(EIO); + } + return 0; +} + + +static int libsrt_setup(URLContext *h, const char *uri, int flags) +{ + struct addrinfo hints = { 0 }, *ai, *cur_ai; + int port, fd = -1; + SRTContext *s = h->priv_data; + const char *p; + char buf[256]; + int ret; + char hostname[1024],proto[1024],path[1024]; + char portstr[10]; + int open_timeout = 5000000; + int eid; + + eid = srt_epoll_create(); + if (eid < 0) + return libsrt_neterrno(h); + s->eid = eid; + + av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname), + &port, path, sizeof(path), uri); + if (strcmp(proto, "srt")) + return AVERROR(EINVAL); + if (port <= 0 || port >= 65536) { + av_log(h, AV_LOG_ERROR, "Port missing in uri\n"); + return AVERROR(EINVAL); + } + p = strchr(uri, '?'); + if (p) { + if (av_find_info_tag(buf, sizeof(buf), "timeout", p)) { + s->rw_timeout = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "listen_timeout", p)) { + s->listen_timeout = strtol(buf, NULL, 10); + } + } + if (s->rw_timeout >= 0) { + open_timeout = h->rw_timeout = s->rw_timeout; + } + hints.ai_family = AF_UNSPEC; + hints.ai_socktype = SOCK_DGRAM; + snprintf(portstr, sizeof(portstr), "%d", port); + if (s->mode == SRT_MODE_LISTENER) + hints.ai_flags |= AI_PASSIVE; + ret = getaddrinfo(hostname[0] ? hostname : NULL, portstr, &hints, &ai); + if (ret) { + av_log(h, AV_LOG_ERROR, + "Failed to resolve hostname %s: %s\n", + hostname, gai_strerror(ret)); + return AVERROR(EIO); + } + + cur_ai = ai; + + restart: + + fd = srt_socket(cur_ai->ai_family, cur_ai->ai_socktype, 0); + if (fd < 0) { + ret = libsrt_neterrno(h); + goto fail; + } + + if ((ret = libsrt_set_options_pre(h, fd)) < 0) { + goto fail; + } + + /* Set the socket's send or receive buffer sizes, if specified. + If unspecified or setting fails, system default is used. */ + if (s->recv_buffer_size > 0) { + srt_setsockopt(fd, SOL_SOCKET, SRTO_UDP_RCVBUF, &s->recv_buffer_size, sizeof (s->recv_buffer_size)); + } + if (s->send_buffer_size > 0) { + srt_setsockopt(fd, SOL_SOCKET, SRTO_UDP_SNDBUF, &s->send_buffer_size, sizeof (s->send_buffer_size)); + } + if (s->mode == SRT_MODE_LISTENER) { + // multi-client + if ((ret = libsrt_listen(s->eid, fd, cur_ai->ai_addr, cur_ai->ai_addrlen, h, open_timeout / 1000)) < 0) + goto fail1; + fd = ret; + } else { + if (s->mode == SRT_MODE_RENDEZVOUS) { + ret = srt_bind(fd, cur_ai->ai_addr, cur_ai->ai_addrlen); + if (ret) + goto fail1; + } + + if ((ret = libsrt_listen_connect(s->eid, fd, cur_ai->ai_addr, cur_ai->ai_addrlen, + open_timeout / 1000, h, !!cur_ai->ai_next)) < 0) { + if (ret == AVERROR_EXIT) + goto fail1; + else + goto fail; + } + } + if ((ret = libsrt_set_options_post(h, fd)) < 0) { + goto fail; + } + + h->is_streamed = 1; + s->fd = fd; + + freeaddrinfo(ai); + return 0; + + fail: + if (cur_ai->ai_next) { + /* Retry with the next sockaddr */ + cur_ai = cur_ai->ai_next; + if (fd >= 0) + srt_close(fd); + ret = 0; + goto restart; + } + fail1: + if (fd >= 0) + srt_close(fd); + freeaddrinfo(ai); + return ret; +} + +static int libsrt_open(URLContext *h, const char *uri, int flags) +{ + SRTContext *s = h->priv_data; + const char * p; + char buf[256]; + + if (srt_startup() < 0) { + return AVERROR_UNKNOWN; + } + + /* SRT options (srt/srt.h) */ + p = strchr(uri, '?'); + if (p) { + if (av_find_info_tag(buf, sizeof(buf), "maxbw", p)) { + s->maxbw = strtoll(buf, NULL, 0); + } + if (av_find_info_tag(buf, sizeof(buf), "pbkeylen", p)) { + s->pbkeylen = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "passphrase", p)) { + s->passphrase = av_strndup(buf, strlen(buf)); + } + if (av_find_info_tag(buf, sizeof(buf), "mss", p)) { + s->mss = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "ffs", p)) { + s->ffs = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "ipttl", p)) { + s->ipttl = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "iptos", p)) { + s->iptos = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "inputbw", p)) { + s->inputbw = strtoll(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "oheadbw", p)) { + s->oheadbw = strtoll(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "tsbpddelay", p)) { + s->tsbpddelay = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "tlpktdrop", p)) { + s->tlpktdrop = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "nakreport", p)) { + s->nakreport = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "connect_timeout", p)) { + s->connect_timeout = strtol(buf, NULL, 10); + } + if (av_find_info_tag(buf, sizeof(buf), "mode", p)) { + if (!strcmp(buf, "caller")) { + s->mode = SRT_MODE_CALLER; + } else if (!strcmp(buf, "listener")) { + s->mode = SRT_MODE_LISTENER; + } else if (!strcmp(buf, "rendezvous")) { + s->mode = SRT_MODE_RENDEZVOUS; + } else { + return AVERROR(EIO); + } + } + } + return libsrt_setup(h, uri, flags); +} + +static int libsrt_read(URLContext *h, uint8_t *buf, int size) +{ + SRTContext *s = h->priv_data; + int ret; + + if (!(h->flags & AVIO_FLAG_NONBLOCK)) { + ret = libsrt_network_wait_fd_timeout(h, s->eid, s->fd, 0, h->rw_timeout, &h->interrupt_callback); + if (ret) + return ret; + } + + ret = srt_recvmsg(s->fd, buf, size); + if (ret < 0) { + ret = libsrt_neterrno(h); + } + + return ret; +} + +static int libsrt_write(URLContext *h, const uint8_t *buf, int size) +{ + SRTContext *s = h->priv_data; + int ret; + + if (!(h->flags & AVIO_FLAG_NONBLOCK)) { + ret = libsrt_network_wait_fd_timeout(h, s->eid, s->fd, 1, h->rw_timeout, &h->interrupt_callback); + if (ret) + return ret; + } + + ret = srt_sendmsg(s->fd, buf, size, -1, 0); + if (ret < 0) { + ret = libsrt_neterrno(h); + } + + return ret; +} + +static int libsrt_close(URLContext *h) +{ + SRTContext *s = h->priv_data; + + srt_close(s->fd); + + srt_epoll_release(s->eid); + + srt_cleanup(); + + return 0; +} + +static int libsrt_get_file_handle(URLContext *h) +{ + SRTContext *s = h->priv_data; + return s->fd; +} + +static const AVClass libsrt_class = { + .class_name = "libsrt", + .item_name = av_default_item_name, + .option = libsrt_options, + .version = LIBAVUTIL_VERSION_INT, +}; + +const URLProtocol ff_libsrt_protocol = { + .name = "srt", + .url_open = libsrt_open, + .url_read = libsrt_read, + .url_write = libsrt_write, + .url_close = libsrt_close, + .url_get_file_handle = libsrt_get_file_handle, + .priv_data_size = sizeof(SRTContext), + .flags = URL_PROTOCOL_FLAG_NETWORK, + .priv_data_class = &libsrt_class, +}; diff --git a/libavformat/protocols.c b/libavformat/protocols.c index 669d74d5a8..ad95659795 100644 --- a/libavformat/protocols.c +++ b/libavformat/protocols.c @@ -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;