Hi. I had a look at the whole code. There are a few remarks below.
Sorry for the delay, a lot of things on my place these days. Nablet Developer (2018-01-30): > protocol requires libsrt (https://github.com/Haivision/srt) to be > installed > > Signed-off-by: Nablet Developer <s...@nablet.com> > --- > MAINTAINERS | 1 + > configure | 9 + > doc/protocols.texi | 116 +++++++++ > libavformat/Makefile | 1 + > libavformat/opensrt.c | 621 > ++++++++++++++++++++++++++++++++++++++++++++++++ > libavformat/protocols.c | 1 + > 6 files changed, 749 insertions(+) > create mode 100644 libavformat/opensrt.c > > diff --git a/MAINTAINERS b/MAINTAINERS > index ba7a728..0317f24 100644 > --- a/MAINTAINERS > +++ b/MAINTAINERS > @@ -498,6 +498,7 @@ Protocols: > http.c Ronald S. Bultje > libssh.c Lukasz Marek > mms*.c Ronald S. Bultje > + opensrt.c Nablet Developer > udp.c Luca Abeni > icecast.c Marvin Scholz > > diff --git a/configure b/configure > index fcfa7aa..57705ee 100755 > --- a/configure > +++ b/configure > @@ -294,6 +294,7 @@ External library support: > --enable-opengl enable OpenGL rendering [no] > --enable-openssl enable openssl, needed for https support > if gnutls or libtls 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] > @@ -1641,6 +1642,7 @@ EXTERNAL_LIBRARY_LIST=" > mediacodec > openal > opengl > + opensrt > " > > HWACCEL_AUTODETECT_LIBRARY_LIST=" > @@ -3148,6 +3150,8 @@ libssh_protocol_deps="libssh" > libtls_conflict="openssl gnutls" > 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" > @@ -5986,6 +5990,7 @@ 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 && 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 || > @@ -6036,6 +6041,10 @@ if enabled decklink; then > esac > fi > > +if enabled opensrt; then > + opensrt_protocol_extralibs="$opensrt_protocol_extralibs -lsrt" > +fi This looks suspicious: pkg-config should have added -lsrt automatically. > + > 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 98deb73..2e5e630 100644 > --- a/doc/protocols.texi > +++ b/doc/protocols.texi > @@ -755,6 +755,122 @@ 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 Please do not truncate the name. > +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). Nit: maybe wrap the lines shorter, longer lines are more tiring to read. > + > +@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 > +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. > +caller opens client connection. > +listener starts server to listen for incoming connections. > +rendezvous use Rendez-Vous connection mode. > +Default valus is 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 > +Set raise error timeout. > + > +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 > +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 libssh > > Secure File Transfer Protocol via libssh > diff --git a/libavformat/Makefile b/libavformat/Makefile > index de0de92..bd92071 100644 > --- a/libavformat/Makefile > +++ b/libavformat/Makefile > @@ -598,6 +598,7 @@ 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..0b16391 > --- /dev/null > +++ b/libavformat/opensrt.c > @@ -0,0 +1,621 @@ > +/* > + * This file is part of FFmpeg. > + * > + * FFmpeg is free software; you can redistribute it and/or > + * modify it under the terms of the GNU Lesser General Public > + * License as published by the Free Software Foundation; either > + * version 2.1 of the License, or (at your option) any later version. > + * > + * FFmpeg is distributed in the hope that it will be useful, > + * but WITHOUT ANY WARRANTY; without even the implied warranty of > + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU > + * Lesser General Public License for more details. > + * > + * You should have received a copy of the GNU Lesser General Public > + * License along with FFmpeg; if not, write to the Free Software > + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 > USA > + */ > + > +/** > + * @file > + * Haivision Open SRT (Secure Reliable Transport) protocol > + */ > + > +#include "avformat.h" > +#include "libavutil/avassert.h" > +#include "libavutil/parseutils.h" > +#include "libavutil/opt.h" > +#include "libavutil/time.h" > + > +#include "internal.h" > +#include "network.h" > +#include "os_support.h" > +#include "url.h" > +#if HAVE_POLL_H > +#include <poll.h> > +#endif > + > +#if CONFIG_OPENSRT_PROTOCOL > +#include <srt/srt.h> > +#endif > + > +enum SRTMode { > + SRT_MODE_CALLER = 0, > + SRT_MODE_LISTENER = 1, > + SRT_MODE_RENDEZVOUS = 2 > +}; > + > +typedef struct SRTContext { > + int fd; > + int rw_timeout; All AV_OPT_TYPE_DURATION fields need to be int64_t. > + 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 of socket I/O operations", > OFFSET(rw_timeout), AV_OPT_TYPE_DURATION, { .i64 = > -1 }, -1, INT_MAX, .flags = D|E }, > + { "listen_timeout", "Connection awaiting timeout", > OFFSET(listen_timeout), AV_OPT_TYPE_DURATION, { .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 to absorb burst of missed > packet retransmission", OFFSET(tsbpddelay), AV_OPT_TYPE_DURATION, { > .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. Ccaller default: 3000, rendezvous > (x 10)", OFFSET(conntimeo), AV_OPT_TYPE_DURATION, { .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); AVERROR_EXTERNAL; or even better, map all the error code that can be mapped. > +} > + > +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; > + } It looks like it will make quite a few system calls. Maybe create eid at the beginning and reuse it? > + 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); > +} You are wrapping the arguments in a pollfd structure, and then unwrapping them to pass them to the libsrt API. It looks unnecessary, and only there because you followed the example of TCP too closely. I think you should merge opensrt_poll() and this function to use fd directly with srt_epoll_add_usock(). > + > +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); > + } > + } > +} This block looks like a duplicate of ff_network_wait_fd_timeout() with the function changed. It would probably be better to factor the code, but it is not trivial to do it cleanly. In the meantime, please add a comment, maybe: /* TODO de-duplicate code from ff_network_wait_fd_timeout() */ > + > +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; > +} Ditto for ff_poll_interrupt(). > + > +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(h, AV_LOG_DEBUG, "opensrt_socket_nonblock failed\n"); > + > + return ret; > +} > + > +static int opensrt_listen(int fd, const struct sockaddr *addr, socklen_t > addrlen, URLContext *h) > +{ > + 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 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; > + > + if (opensrt_socket_nonblock(fd, 1) < 0) > + av_log(h, 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; > + ret = srt_getlasterror(NULL); > + srt_clearlasterror(); > + if (ret != 0) { > + char errbuf[100]; > + ret = AVERROR(ret); > + av_strerror(ret, errbuf, sizeof(errbuf)); Use av_err2str(). > + 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()); Missing \n. > + return AVERROR(EIO); Is it really the best error code for this situation? > + } > + 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); Ditto. > + } > + 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; > + int tsbpddelay = s->tsbpddelay / 1000; > + int conntimeo = s->conntimeo; > + > + 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 (tsbpddelay >= 0 && srt_setsockopt(fd, 0, SRTO_TSBPDDELAY, > &tsbpddelay, sizeof(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 (conntimeo >= 0 && srt_setsockopt(fd, 0, SRTO_CONNTIMEO, &conntimeo, > sizeof(conntimeo)) < 0) { > + av_log(h, AV_LOG_ERROR, "failed to set option SRTO_CONNTIMEO on > socket: %s", srt_getlasterror_str()); > + return AVERROR(EIO); > + } Please factor that. > + 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); getaddrinfo(hostname[0] ? hostname : NULL), maybe? > + 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, > h)) < 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]; > + > + if (srt_startup() < 0) { > + return AVERROR(EIO); AVERROR_EXTERNAL or more accurate translation. > + } > + > + /* 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); Maybe use 0 instead of 10 to allow hex. > + } > + 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; > + } Missing final case. > + } > + } > + 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 / 1000, 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; Regards, -- Nicolas George
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