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[79.124.17.100]) by mx.google.com with ESMTP id d16si1062899wrb.312.2017.12.13.00.31.59; Wed, 13 Dec 2017 00:31:59 -0800 (PST) Received-SPF: pass (google.com: domain of ffmpeg-devel-bounces@ffmpeg.org designates 79.124.17.100 as permitted sender) client-ip=79.124.17.100; Authentication-Results: mx.google.com; spf=pass (google.com: domain of ffmpeg-devel-bounces@ffmpeg.org designates 79.124.17.100 as permitted sender) smtp.mailfrom=ffmpeg-devel-bounces@ffmpeg.org Received: from [127.0.1.1] (localhost [127.0.0.1]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTP id AD46968A3AB; Wed, 13 Dec 2017 10:31:46 +0200 (EET) X-Original-To: ffmpeg-devel@ffmpeg.org Delivered-To: ffmpeg-devel@ffmpeg.org Received: from mout.kundenserver.de (mout.kundenserver.de [217.72.192.74]) by ffbox0-bg.mplayerhq.hu (Postfix) with ESMTPS id 7127B68A251 for ; Wed, 13 Dec 2017 10:31:39 +0200 (EET) Received: from ubuntu-server.internal.divx.com ([91.216.211.197]) by mrelayeu.kundenserver.de (mreue103 [212.227.15.183]) with ESMTPSA (Nemesis) id 0M9N4I-1eK3pu1V78-00Cg2y; Wed, 13 Dec 2017 09:31:44 +0100 From: Nablet Developer To: ffmpeg-devel@ffmpeg.org Date: Wed, 13 Dec 2017 15:31:04 +0700 Message-Id: <1513153864-31781-2-git-send-email-sdk@nablet.com> X-Mailer: git-send-email 2.7.4 In-Reply-To: <1513153864-31781-1-git-send-email-sdk@nablet.com> References: <1513153864-31781-1-git-send-email-sdk@nablet.com> X-Provags-ID: V03:K0:f9iBwr5dbE4Uy+WDE2fNygP83743WJuRbFBIK5l5Cb0qjyz07lV F1x6eQN2j+WpwbBevd9KcgkIQ3CBBKLI+IGi36RJ5jBul3cy/lwN59MaV3Taj9/J/FGvKan sXGbRoCIgbMJn8jj+OQtR8BOSYyHHGml0SLtl7oEVJW9K5ys/dZU59Yy7B9rrrYSA7MCCRR LKdlDrmZjH9QAG6Kvr9Ag== X-UI-Out-Filterresults: notjunk:1; V01:K0:IyaAbiDJ098=:liDhXF16ssXYuCGMdElKiY dIbPj3VjDYeIi+Bx6Mb1XNBjCTTPu/4JrrKPhVEBdvzYCaPN0GxbKuktRFA6LD+XVJrCsMtd0 08CRZcP8BOIOzWmPYP6xJS0pnsjlEak/tJWhu8UZWeMNVy3VkGA8cOa5uLqpLvvgEmFUQA4Hz tTR90mug11ax0BTKOhUZ3RktlPsRIxKTxaMtv+WZmaeVRDb4gKCsg5iKuwWWFeqD9eEKGHRo8 YAF9hiXMSgNRG8hB9PVIbgSbw7bMAbOtN8jgmGJdeFmjXkWNLDG/H6XKCqun3Q2ayeXAos6xX 321ShkrRE7z0npMeWHXpci3WPzYfhbxIjBW+pkzMh2X/St79HqxmzxxsneTwOAkozzYlCsJtp SZ3BHn2znux0gAhX3jdxDhqog+rmMtoOxmt8qOsSR+yKxUGRGxgF4Q3QvTuoGFlj1ZeuxR/Tc gTGo4QhWKT8RaS3GnNFN/w3+fd9gieZaTyd5TXKQHMjVYsblwi8cdliaUHL2UNSo+ZbJhQ1qI lMZ8Maz4usk7NqQsgqVO+E9drKuGZqj0x9LNJSKGeGPhhAzXTwTohLVU0uybfLrVm8V92WeEm veh1+s88OoN0yHWBTOWwvEd7wEmXDtwIn7r5eNCuLXfZqnr+iRsRXU3dMA5RnEl1pg62K8jyP /DlSUOuwUHFK7Pq8TOkfGOkStsINVPIVIRCGiwv9/kZburEEexl8iKf84udkWtnb/XHB5Li1L lb87qSltXsr19a3p9fN39rWdfn47n7LzbtZQwg== Subject: [FFmpeg-devel] [PATCH] avformat/opensrt: add Haivision Open SRT protocol X-BeenThere: ffmpeg-devel@ffmpeg.org X-Mailman-Version: 2.1.20 Precedence: list List-Id: FFmpeg development discussions and patches List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Reply-To: FFmpeg development discussions and patches Cc: Nablet Developer MIME-Version: 1.0 Errors-To: ffmpeg-devel-bounces@ffmpeg.org Sender: "ffmpeg-devel" protocol requires libsrt (https://github.com/Haivision/srt) to be installed Signed-off-by: Nablet Developer --- 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/}. +@section srt + +Haivision Secure Reliable Transport Protocol via libsrt. + +The required syntax for a SRT url is: +@example +srt://@var{hostname}:@var{port}[?@var{options}] +@end example + +@var{options} contains a list of &-separated options of the form +@var{key}=@var{val}. + +This protocol accepts the following options. + +@table @option +@item 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). + +@item fc=@var{bytes} +Flight Flag Size (Window Size), in bytes. FC is actually an internal parameter and +you should set it to not less than @option{recv_buffer_size} and @option{mss}. +The default value is relatively large, therefore unless you set a very large +receiver buffer, you do not need to change this option. Default value is 25600. + +@item inputbw=@var{bytes/seconds} +Sender nominal input rate, in bytes per seconds. Used along with @option{oheadbw}, +when @option{maxbw} is set to relative (0), to calculate maximum sending rate when +recovery packets are sent along with main media stream: +@option{inputbw} * (100 + @option{oheadbw}) / 100 +if @option{inputbw} is not set while @option{maxbw} is set to relative (0), the actual +ctual input rate is evaluated inside the library. Default value is 0. + +@item iptos=@var{tos} +IP Type of Service. Applies to sender only. Default value is 0xB8. + +@item ipttl=@var{ttl} +IP Time To Live. Applies to sender only. Default value is 64. + +@item listen_timeout=@var{milliseconds} +Set listen timeout, expressed in milliseconds. + +@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{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). + +@item mss=@var{bytes} +Maximum Segment Size, in bytes. Used for buffer allocation and rate calculation using +packet counter assuming fully filled packets. The smallest MSS between the peers is +used. This is 1500 by default in the overall internet. This is the maximum size of the +UDP packet and can be only decreased, unless you have some unusual dedicated network +settings. Default value is 1500. + +@item nakreport=@var{1|0} +If set to 1, Receiver will send `UMSG_LOSSREPORT` messages periodically until the +lost packet is retransmitted or intentionally dropped. Default value is 1. + +@item oheadbw=@var{percents} +Recovery bandwidth overhead above input rate, in percents. See @option{inputbw}. +Default value is 25%. + +@item passphrase=@var{string} +HaiCrypt Encryption/Decryption Passphrase string, length from 10 to 79 characters. +The passphrase is the shared secret between the sender and the receiver. +It is used to generate the Key Encrypting Key using PBKDF2 (Password-Based +Key Deriviation Function). It is used only if @option{pbkeylen} is non-zero. +t is used on the receiver only if the received data is encrypted. +The configured passphrase cannot be get back (write-only). + +@item pbkeylen=@var{bytes} +Sender encryption key length, in bytes. Only can be set to 0, 16, 24 and 32. +Enable sender encryption if not 0. Not required on receiver (set to 0), +key size obtained from sender in HaiCrypt handshake. Default value is 0. + +@item recv_buffer_size=@var{bytes} +Set receive buffer size, expressed bytes. + +@item send_buffer_size=@var{bytes} +Set send buffer size, expressed bytes. + +@item timeout=@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. + +@item tlpktdrop=@var{1|0} +Too-late Packet Drop. When enabled on receiver, it skips missing packets that +have not been delivered in time and deliver the following packets to the application +when their time-to-play has come. It also send a fake ACK to sender. When enabled on +sender and enabled on the receiving peer, sender drops the older packets that have no +chance to be delivered in time. It was automatically enabled in sender if receiver +supports it. + +@item tsbpddelay=@var{milliseconds} +Timestamp-based Packet Delivery Delay, in milliseconds. +Used to absorb burst of missed packet retransmission. + +@end table + +For more information see: @url{https://github.com/Haivision/srt}. + + @section libssh Secure File Transfer Protocol via libssh diff --git a/libavformat/Makefile b/libavformat/Makefile index 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 +#endif + +#if CONFIG_OPENSRT_PROTOCOL +#include +#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;