[FFmpeg-devel] [PATCH] avformat/opensrt: add Haivision Open SRT protocol
Nicolas George
george at nsup.org
Sun Feb 11 21:04:12 EET 2018
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 <sdk at 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/}.
>
> + at section srt
> +
> +Haivision Secure Reliable Transport Protocol via libsrt.
> +
> +The required syntax for a SRT url is:
> + at example
> +srt://@var{hostname}:@var{port}[?@var{options}]
> + at end example
> +
> + at var{options} contains a list of &-separated options of the form
> + at var{key}=@var{val}.
> +
> +This protocol accepts the following options.
> +
> + at table @option
> + at 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.
> +
> + at 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.
> +
> + at 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:
> + at 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.
> +
> + at item iptos=@var{tos}
> +IP Type of Service. Applies to sender only. Default value is 0xB8.
> +
> + at item ipttl=@var{ttl}
> +IP Time To Live. Applies to sender only. Default value is 64.
> +
> + at item listen_timeout
> +Set socket listen timeout.
> +
> + at 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)
> +
> + at 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.
> +
> + at 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.
> +
> + at 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.
> +
> + at item oheadbw=@var{percents}
> +Recovery bandwidth overhead above input rate, in percents. See @option{inputbw}.
> +Default value is 25%.
> +
> + at 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).
> +
> + at 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.
> +
> + at item recv_buffer_size=@var{bytes}
> +Set receive buffer size, expressed bytes.
> +
> + at item send_buffer_size=@var{bytes}
> +Set send buffer size, expressed bytes.
> +
> + at 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.
> +
> + at 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.
> +
> + at item tsbpddelay
> +Timestamp-based Packet Delivery Delay.
> +Used to absorb burst of missed packet retransmission.
> +
> + at 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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