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https://git.ffmpeg.org/ffmpeg.git
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c2ac3b8e6a
Several SRT options are missing. Since pkg_config requires libsrt v1.3.0 and above, it should be able to support options added in libsrt v1.3.0 and below. This commit adds 8 SRT options. sndbuf, rcvbuf, lossmaxttl, minversion, streamid, smoother, messageapi and transtype The keys of option are equivalent to stransmit. https://github.com/Haivision/srt/blob/v1.3.0/apps/socketoptions.hpp#L196-L223 Signed-off-by: Marton Balint <cus@passwd.hu>
662 lines
28 KiB
C
662 lines
28 KiB
C
/*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Haivision Open SRT (Secure Reliable Transport) protocol
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*/
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#include <srt/srt.h>
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#include "libavutil/avassert.h"
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#include "libavutil/opt.h"
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#include "libavutil/parseutils.h"
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#include "libavutil/time.h"
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#include "avformat.h"
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#include "internal.h"
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#include "network.h"
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#include "os_support.h"
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#include "url.h"
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/* This is for MPEG-TS and it's a default SRTO_PAYLOADSIZE for SRTT_LIVE (8 TS packets) */
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#ifndef SRT_LIVE_DEFAULT_PAYLOAD_SIZE
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#define SRT_LIVE_DEFAULT_PAYLOAD_SIZE 1316
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#endif
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/* This is the maximum payload size for Live mode, should you have a different payload type than MPEG-TS */
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#ifndef SRT_LIVE_MAX_PAYLOAD_SIZE
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#define SRT_LIVE_MAX_PAYLOAD_SIZE 1456
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#endif
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enum SRTMode {
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SRT_MODE_CALLER = 0,
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SRT_MODE_LISTENER = 1,
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SRT_MODE_RENDEZVOUS = 2
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};
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typedef struct SRTContext {
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const AVClass *class;
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int fd;
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int eid;
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int64_t rw_timeout;
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int64_t listen_timeout;
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int recv_buffer_size;
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int send_buffer_size;
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int64_t maxbw;
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int pbkeylen;
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char *passphrase;
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int mss;
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int ffs;
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int ipttl;
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int iptos;
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int64_t inputbw;
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int oheadbw;
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int64_t latency;
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int tlpktdrop;
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int nakreport;
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int64_t connect_timeout;
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int payload_size;
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int64_t rcvlatency;
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int64_t peerlatency;
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enum SRTMode mode;
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int sndbuf;
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int rcvbuf;
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int lossmaxttl;
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int minversion;
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char *streamid;
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char *smoother;
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int messageapi;
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SRT_TRANSTYPE transtype;
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} SRTContext;
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#define D AV_OPT_FLAG_DECODING_PARAM
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#define E AV_OPT_FLAG_ENCODING_PARAM
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#define OFFSET(x) offsetof(SRTContext, x)
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static const AVOption libsrt_options[] = {
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{ "rw_timeout", "Timeout of socket I/O operations", OFFSET(rw_timeout), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
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{ "listen_timeout", "Connection awaiting timeout", OFFSET(listen_timeout), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
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{ "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 },
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{ "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 },
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{ "pkt_size", "Maximum SRT packet size", OFFSET(payload_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, SRT_LIVE_MAX_PAYLOAD_SIZE, .flags = D|E, "payload_size" },
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{ "payload_size", "Maximum SRT packet size", OFFSET(payload_size), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, SRT_LIVE_MAX_PAYLOAD_SIZE, .flags = D|E, "payload_size" },
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{ "ts_size", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRT_LIVE_DEFAULT_PAYLOAD_SIZE }, INT_MIN, INT_MAX, .flags = D|E, "payload_size" },
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{ "max_size", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRT_LIVE_MAX_PAYLOAD_SIZE }, INT_MIN, INT_MAX, .flags = D|E, "payload_size" },
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{ "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 },
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{ "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 },
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{ "passphrase", "Crypto PBKDF2 Passphrase size[0,10..64] 0:disable crypto", OFFSET(passphrase), AV_OPT_TYPE_STRING, { .str = NULL }, .flags = D|E },
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{ "mss", "The Maximum Segment Size", OFFSET(mss), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 1500, .flags = D|E },
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{ "ffs", "Flight flag size (window size) (in bytes)", OFFSET(ffs), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E },
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{ "ipttl", "IP Time To Live", OFFSET(ipttl), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 255, .flags = D|E },
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{ "iptos", "IP Type of Service", OFFSET(iptos), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 255, .flags = D|E },
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{ "inputbw", "Estimated input stream rate", OFFSET(inputbw), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
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{ "oheadbw", "MaxBW ceiling based on % over input stream rate", OFFSET(oheadbw), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 100, .flags = D|E },
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{ "latency", "receiver delay to absorb bursts of missed packet retransmissions", OFFSET(latency), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
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{ "tsbpddelay", "deprecated, same effect as latency option", OFFSET(latency), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
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{ "rcvlatency", "receive latency", OFFSET(rcvlatency), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
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{ "peerlatency", "peer latency", OFFSET(peerlatency), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
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{ "tlpktdrop", "Enable receiver pkt drop", OFFSET(tlpktdrop), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 1, .flags = D|E },
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{ "nakreport", "Enable receiver to send periodic NAK reports", OFFSET(nakreport), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 1, .flags = D|E },
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{ "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 },
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{ "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" },
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{ "caller", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRT_MODE_CALLER }, INT_MIN, INT_MAX, .flags = D|E, "mode" },
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{ "listener", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRT_MODE_LISTENER }, INT_MIN, INT_MAX, .flags = D|E, "mode" },
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{ "rendezvous", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRT_MODE_RENDEZVOUS }, INT_MIN, INT_MAX, .flags = D|E, "mode" },
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{ "sndbuf", "Send buffer size (in bytes)", OFFSET(sndbuf), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E },
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{ "rcvbuf", "Receive buffer size (in bytes)", OFFSET(rcvbuf), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E },
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{ "lossmaxttl", "Maximum possible packet reorder tolerance", OFFSET(lossmaxttl), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E },
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{ "minversion", "The minimum SRT version that is required from the peer", OFFSET(minversion), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, INT_MAX, .flags = D|E },
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{ "streamid", "A string of up to 512 characters that an Initiator can pass to a Responder", OFFSET(streamid), AV_OPT_TYPE_STRING, { .str = NULL }, .flags = D|E },
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{ "smoother", "The type of Smoother used for the transmission for that socket", OFFSET(smoother), AV_OPT_TYPE_STRING, { .str = NULL }, .flags = D|E },
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{ "messageapi", "Enable message API", OFFSET(messageapi), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 1, .flags = D|E },
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{ "transtype", "The transmission type for the socket", OFFSET(transtype), AV_OPT_TYPE_INT, { .i64 = SRTT_INVALID }, SRTT_LIVE, SRTT_INVALID, .flags = D|E, "transtype" },
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{ "live", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRTT_LIVE }, INT_MIN, INT_MAX, .flags = D|E, "transtype" },
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{ "file", NULL, 0, AV_OPT_TYPE_CONST, { .i64 = SRTT_FILE }, INT_MIN, INT_MAX, .flags = D|E, "transtype" },
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{ NULL }
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};
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static int libsrt_neterrno(URLContext *h)
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{
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int err = srt_getlasterror(NULL);
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av_log(h, AV_LOG_ERROR, "%s\n", srt_getlasterror_str());
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if (err == SRT_EASYNCRCV)
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return AVERROR(EAGAIN);
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return AVERROR_UNKNOWN;
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}
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static int libsrt_socket_nonblock(int socket, int enable)
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{
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int ret = srt_setsockopt(socket, 0, SRTO_SNDSYN, &enable, sizeof(enable));
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if (ret < 0)
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return ret;
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return srt_setsockopt(socket, 0, SRTO_RCVSYN, &enable, sizeof(enable));
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}
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static int libsrt_network_wait_fd(URLContext *h, int eid, int fd, int write)
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{
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int ret, len = 1;
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int modes = write ? SRT_EPOLL_OUT : SRT_EPOLL_IN;
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SRTSOCKET ready[1];
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if (srt_epoll_add_usock(eid, fd, &modes) < 0)
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return libsrt_neterrno(h);
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if (write) {
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ret = srt_epoll_wait(eid, 0, 0, ready, &len, POLLING_TIME, 0, 0, 0, 0);
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} else {
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ret = srt_epoll_wait(eid, ready, &len, 0, 0, POLLING_TIME, 0, 0, 0, 0);
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}
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if (ret < 0) {
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if (srt_getlasterror(NULL) == SRT_ETIMEOUT)
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ret = AVERROR(EAGAIN);
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else
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ret = libsrt_neterrno(h);
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} else {
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ret = 0;
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}
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if (srt_epoll_remove_usock(eid, fd) < 0)
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return libsrt_neterrno(h);
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return ret;
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}
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/* TODO de-duplicate code from ff_network_wait_fd_timeout() */
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static int libsrt_network_wait_fd_timeout(URLContext *h, int eid, int fd, int write, int64_t timeout, AVIOInterruptCB *int_cb)
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{
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int ret;
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int64_t wait_start = 0;
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while (1) {
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if (ff_check_interrupt(int_cb))
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return AVERROR_EXIT;
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ret = libsrt_network_wait_fd(h, eid, fd, write);
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if (ret != AVERROR(EAGAIN))
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return ret;
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if (timeout > 0) {
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if (!wait_start)
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wait_start = av_gettime_relative();
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else if (av_gettime_relative() - wait_start > timeout)
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return AVERROR(ETIMEDOUT);
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}
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}
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}
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static int libsrt_listen(int eid, int fd, const struct sockaddr *addr, socklen_t addrlen, URLContext *h, int timeout)
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{
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int ret;
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int reuse = 1;
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if (srt_setsockopt(fd, SOL_SOCKET, SRTO_REUSEADDR, &reuse, sizeof(reuse))) {
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av_log(h, AV_LOG_WARNING, "setsockopt(SRTO_REUSEADDR) failed\n");
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}
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ret = srt_bind(fd, addr, addrlen);
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if (ret)
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return libsrt_neterrno(h);
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ret = srt_listen(fd, 1);
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if (ret)
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return libsrt_neterrno(h);
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while ((ret = libsrt_network_wait_fd_timeout(h, eid, fd, 1, timeout, &h->interrupt_callback))) {
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switch (ret) {
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case AVERROR(ETIMEDOUT):
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continue;
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default:
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return ret;
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}
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}
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ret = srt_accept(fd, NULL, NULL);
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if (ret < 0)
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return libsrt_neterrno(h);
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if (libsrt_socket_nonblock(ret, 1) < 0)
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av_log(h, AV_LOG_DEBUG, "libsrt_socket_nonblock failed\n");
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return ret;
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}
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static int libsrt_listen_connect(int eid, int fd, const struct sockaddr *addr, socklen_t addrlen, int timeout, URLContext *h, int will_try_next)
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{
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int ret;
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if (libsrt_socket_nonblock(fd, 1) < 0)
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av_log(h, AV_LOG_DEBUG, "ff_socket_nonblock failed\n");
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while ((ret = srt_connect(fd, addr, addrlen))) {
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ret = libsrt_neterrno(h);
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switch (ret) {
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case AVERROR(EINTR):
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if (ff_check_interrupt(&h->interrupt_callback))
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return AVERROR_EXIT;
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continue;
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case AVERROR(EINPROGRESS):
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case AVERROR(EAGAIN):
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ret = libsrt_network_wait_fd_timeout(h, eid, fd, 1, timeout, &h->interrupt_callback);
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if (ret < 0)
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return ret;
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ret = srt_getlasterror(NULL);
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srt_clearlasterror();
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if (ret != 0) {
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char buf[128];
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ret = AVERROR(ret);
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av_strerror(ret, buf, sizeof(buf));
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if (will_try_next)
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av_log(h, AV_LOG_WARNING,
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"Connection to %s failed (%s), trying next address\n",
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h->filename, buf);
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else
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av_log(h, AV_LOG_ERROR, "Connection to %s failed: %s\n",
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h->filename, buf);
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}
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default:
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return ret;
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}
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}
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return ret;
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}
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static int libsrt_setsockopt(URLContext *h, int fd, SRT_SOCKOPT optname, const char * optnamestr, const void * optval, int optlen)
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{
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if (srt_setsockopt(fd, 0, optname, optval, optlen) < 0) {
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av_log(h, AV_LOG_ERROR, "failed to set option %s on socket: %s\n", optnamestr, srt_getlasterror_str());
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return AVERROR(EIO);
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}
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return 0;
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}
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static int libsrt_getsockopt(URLContext *h, int fd, SRT_SOCKOPT optname, const char * optnamestr, void * optval, int * optlen)
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{
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if (srt_getsockopt(fd, 0, optname, optval, optlen) < 0) {
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av_log(h, AV_LOG_ERROR, "failed to get option %s on socket: %s\n", optnamestr, srt_getlasterror_str());
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return AVERROR(EIO);
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}
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return 0;
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}
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/* - The "POST" options can be altered any time on a connected socket.
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They MAY have also some meaning when set prior to connecting; such
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option is SRTO_RCVSYN, which makes connect/accept call asynchronous.
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Because of that this option is treated special way in this app. */
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static int libsrt_set_options_post(URLContext *h, int fd)
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{
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SRTContext *s = h->priv_data;
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if ((s->inputbw >= 0 && libsrt_setsockopt(h, fd, SRTO_INPUTBW, "SRTO_INPUTBW", &s->inputbw, sizeof(s->inputbw)) < 0) ||
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(s->oheadbw >= 0 && libsrt_setsockopt(h, fd, SRTO_OHEADBW, "SRTO_OHEADBW", &s->oheadbw, sizeof(s->oheadbw)) < 0)) {
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return AVERROR(EIO);
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}
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return 0;
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}
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/* - The "PRE" options must be set prior to connecting and can't be altered
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on a connected socket, however if set on a listening socket, they are
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derived by accept-ed socket. */
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static int libsrt_set_options_pre(URLContext *h, int fd)
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{
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SRTContext *s = h->priv_data;
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int yes = 1;
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int latency = s->latency / 1000;
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int rcvlatency = s->rcvlatency / 1000;
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int peerlatency = s->peerlatency / 1000;
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int connect_timeout = s->connect_timeout;
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if ((s->mode == SRT_MODE_RENDEZVOUS && libsrt_setsockopt(h, fd, SRTO_RENDEZVOUS, "SRTO_RENDEZVOUS", &yes, sizeof(yes)) < 0) ||
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(s->transtype != SRTT_INVALID && libsrt_setsockopt(h, fd, SRTO_TRANSTYPE, "SRTO_TRANSTYPE", &s->transtype, sizeof(s->transtype)) < 0) ||
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(s->maxbw >= 0 && libsrt_setsockopt(h, fd, SRTO_MAXBW, "SRTO_MAXBW", &s->maxbw, sizeof(s->maxbw)) < 0) ||
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(s->pbkeylen >= 0 && libsrt_setsockopt(h, fd, SRTO_PBKEYLEN, "SRTO_PBKEYLEN", &s->pbkeylen, sizeof(s->pbkeylen)) < 0) ||
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(s->passphrase && libsrt_setsockopt(h, fd, SRTO_PASSPHRASE, "SRTO_PASSPHRASE", s->passphrase, strlen(s->passphrase)) < 0) ||
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(s->mss >= 0 && libsrt_setsockopt(h, fd, SRTO_MSS, "SRTO_MMS", &s->mss, sizeof(s->mss)) < 0) ||
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(s->ffs >= 0 && libsrt_setsockopt(h, fd, SRTO_FC, "SRTO_FC", &s->ffs, sizeof(s->ffs)) < 0) ||
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(s->ipttl >= 0 && libsrt_setsockopt(h, fd, SRTO_IPTTL, "SRTO_UPTTL", &s->ipttl, sizeof(s->ipttl)) < 0) ||
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(s->iptos >= 0 && libsrt_setsockopt(h, fd, SRTO_IPTOS, "SRTO_IPTOS", &s->iptos, sizeof(s->iptos)) < 0) ||
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(s->latency >= 0 && libsrt_setsockopt(h, fd, SRTO_LATENCY, "SRTO_LATENCY", &latency, sizeof(latency)) < 0) ||
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(s->rcvlatency >= 0 && libsrt_setsockopt(h, fd, SRTO_RCVLATENCY, "SRTO_RCVLATENCY", &rcvlatency, sizeof(rcvlatency)) < 0) ||
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(s->peerlatency >= 0 && libsrt_setsockopt(h, fd, SRTO_PEERLATENCY, "SRTO_PEERLATENCY", &peerlatency, sizeof(peerlatency)) < 0) ||
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(s->tlpktdrop >= 0 && libsrt_setsockopt(h, fd, SRTO_TLPKTDROP, "SRTO_TLPKDROP", &s->tlpktdrop, sizeof(s->tlpktdrop)) < 0) ||
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(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 ) ||
|
|
(s->sndbuf >= 0 && libsrt_setsockopt(h, fd, SRTO_SNDBUF, "SRTO_SNDBUF", &s->sndbuf, sizeof(s->sndbuf)) < 0) ||
|
|
(s->rcvbuf >= 0 && libsrt_setsockopt(h, fd, SRTO_RCVBUF, "SRTO_RCVBUF", &s->rcvbuf, sizeof(s->rcvbuf)) < 0) ||
|
|
(s->lossmaxttl >= 0 && libsrt_setsockopt(h, fd, SRTO_LOSSMAXTTL, "SRTO_LOSSMAXTTL", &s->lossmaxttl, sizeof(s->lossmaxttl)) < 0) ||
|
|
(s->minversion >= 0 && libsrt_setsockopt(h, fd, SRTO_MINVERSION, "SRTO_MINVERSION", &s->minversion, sizeof(s->minversion)) < 0) ||
|
|
(s->streamid && libsrt_setsockopt(h, fd, SRTO_STREAMID, "SRTO_STREAMID", s->streamid, strlen(s->streamid)) < 0) ||
|
|
(s->smoother && libsrt_setsockopt(h, fd, SRTO_SMOOTHER, "SRTO_SMOOTHER", s->smoother, strlen(s->smoother)) < 0) ||
|
|
(s->messageapi >= 0 && libsrt_setsockopt(h, fd, SRTO_MESSAGEAPI, "SRTO_MESSAGEAPI", &s->messageapi, sizeof(s->messageapi)) < 0) ||
|
|
(s->payload_size >= 0 && libsrt_setsockopt(h, fd, SRTO_PAYLOADSIZE, "SRTO_PAYLOADSIZE", &s->payload_size, sizeof(s->payload_size)) < 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;
|
|
}
|
|
|
|
if (flags & AVIO_FLAG_WRITE) {
|
|
int packet_size = 0;
|
|
int optlen = sizeof(packet_size);
|
|
ret = libsrt_getsockopt(h, fd, SRTO_PAYLOADSIZE, "SRTO_PAYLOADSIZE", &packet_size, &optlen);
|
|
if (ret < 0)
|
|
goto fail1;
|
|
if (packet_size > 0)
|
|
h->max_packet_size = packet_size;
|
|
}
|
|
|
|
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), "latency", p)) {
|
|
s->latency = strtol(buf, NULL, 10);
|
|
}
|
|
if (av_find_info_tag(buf, sizeof(buf), "tsbpddelay", p)) {
|
|
s->latency = strtol(buf, NULL, 10);
|
|
}
|
|
if (av_find_info_tag(buf, sizeof(buf), "rcvlatency", p)) {
|
|
s->rcvlatency = strtol(buf, NULL, 10);
|
|
}
|
|
if (av_find_info_tag(buf, sizeof(buf), "peerlatency", p)) {
|
|
s->peerlatency = 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), "payload_size", p) ||
|
|
av_find_info_tag(buf, sizeof(buf), "pkt_size", p)) {
|
|
s->payload_size = 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);
|
|
}
|
|
}
|
|
if (av_find_info_tag(buf, sizeof(buf), "sndbuf", p)) {
|
|
s->sndbuf = strtol(buf, NULL, 10);
|
|
}
|
|
if (av_find_info_tag(buf, sizeof(buf), "rcvbuf", p)) {
|
|
s->rcvbuf = strtol(buf, NULL, 10);
|
|
}
|
|
if (av_find_info_tag(buf, sizeof(buf), "lossmaxttl", p)) {
|
|
s->lossmaxttl = strtol(buf, NULL, 10);
|
|
}
|
|
if (av_find_info_tag(buf, sizeof(buf), "minversion", p)) {
|
|
s->minversion = strtol(buf, NULL, 0);
|
|
}
|
|
if (av_find_info_tag(buf, sizeof(buf), "streamid", p)) {
|
|
av_freep(&s->streamid);
|
|
s->streamid = av_strdup(buf);
|
|
}
|
|
if (av_find_info_tag(buf, sizeof(buf), "smoother", p)) {
|
|
av_freep(&s->smoother);
|
|
s->smoother = av_strdup(buf);
|
|
}
|
|
if (av_find_info_tag(buf, sizeof(buf), "messageapi", p)) {
|
|
s->messageapi = strtol(buf, NULL, 10);
|
|
}
|
|
if (av_find_info_tag(buf, sizeof(buf), "transtype", p)) {
|
|
if (!strcmp(buf, "live")) {
|
|
s->transtype = SRTT_LIVE;
|
|
} else if (!strcmp(buf, "file")) {
|
|
s->transtype = SRTT_FILE;
|
|
} else {
|
|
return AVERROR(EINVAL);
|
|
}
|
|
}
|
|
}
|
|
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,
|
|
};
|