mirror of https://git.ffmpeg.org/ffmpeg.git
Merge commit 'a2fc8dbae85339d1b418d296f2982b6c04c53c57'
* commit 'a2fc8dbae85339d1b418d296f2982b6c04c53c57': Add Haivision SRT protocol Merged-by: James Almer <jamrial@gmail.com>
This commit is contained in:
James Almer
commit
a123e576a4
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@ -51,6 +51,7 @@ version <next>:
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- AV1 Support through libaom
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- E-AC-3 dependent frames support
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- bitstream filter for extracting E-AC-3 core
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- Haivision SRT protocol via libsrt
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version 3.4:
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@ -257,6 +257,7 @@ External library support:
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--enable-libsnappy enable Snappy compression, needed for hap encoding [no]
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--enable-libsoxr enable Include libsoxr resampling [no]
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--enable-libspeex enable Speex de/encoding via libspeex [no]
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--enable-libsrt enable Haivision SRT protocol via libsrt [no]
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--enable-libssh enable SFTP protocol via libssh [no]
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--enable-libtesseract enable Tesseract, needed for ocr filter [no]
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--enable-libtheora enable Theora encoding via libtheora [no]
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@ -1705,6 +1706,7 @@ EXTERNAL_LIBRARY_LIST="
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libsnappy
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libsoxr
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libspeex
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libsrt
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libssh
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libtesseract
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libtheora
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@ -3246,6 +3248,8 @@ libssh_protocol_deps="libssh"
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libtls_conflict="openssl gnutls"
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mmsh_protocol_select="http_protocol"
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mmst_protocol_select="network"
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libsrt_protocol_deps="libsrt"
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libsrt_protocol_select="network"
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rtmp_protocol_conflict="librtmp_protocol"
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rtmp_protocol_select="tcp_protocol"
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rtmp_protocol_suggest="zlib"
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@ -6021,6 +6025,7 @@ enabled libsnappy && require libsnappy snappy-c.h snappy_compress -lsnap
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enabled libsoxr && require libsoxr soxr.h soxr_create -lsoxr
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enabled libssh && require_pkg_config libssh libssh libssh/sftp.h sftp_init
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enabled libspeex && require_pkg_config libspeex speex speex/speex.h speex_decoder_init
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enabled libsrt && require_pkg_config libsrt "srt >= 1.2.0" srt/srt.h srt_socket
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enabled libtesseract && require_pkg_config libtesseract tesseract tesseract/capi.h TessBaseAPICreate
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enabled libtheora && require libtheora theora/theoraenc.h th_info_init -ltheoraenc -ltheoradec -logg
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enabled libtls && require_pkg_config libtls libtls tls.h tls_configure
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@ -1155,6 +1155,146 @@ If set to any value, listen for an incoming connection. Outgoing connection is d
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Set the maximum number of streams. By default no limit is set.
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@end table
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@section srt
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Haivision Secure Reliable Transport Protocol via libsrt.
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The supported syntax for a SRT URL is:
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@example
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srt://@var{hostname}:@var{port}[?@var{options}]
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@end example
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@var{options} contains a list of &-separated options of the form
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@var{key}=@var{val}.
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or
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@example
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@var{options} srt://@var{hostname}:@var{port}
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@end example
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@var{options} contains a list of '-@var{key} @var{val}'
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options.
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This protocol accepts the following options.
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@table @option
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@item connect_timeout
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Connection timeout; SRT cannot connect for RTT > 1500 msec
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(2 handshake exchanges) with the default connect timeout of
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3 seconds. This option applies to the caller and rendezvous
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connection modes. The connect timeout is 10 times the value
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set for the rendezvous mode (which can be used as a
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workaround for this connection problem with earlier versions).
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@item ffs=@var{bytes}
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Flight Flag Size (Window Size), in bytes. FFS is actually an
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internal parameter and you should set it to not less than
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@option{recv_buffer_size} and @option{mss}. The default value
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is relatively large, therefore unless you set a very large receiver buffer,
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you do not need to change this option. Default value is 25600.
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@item inputbw=@var{bytes/seconds}
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Sender nominal input rate, in bytes per seconds. Used along with
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@option{oheadbw}, when @option{maxbw} is set to relative (0), to
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calculate maximum sending rate when recovery packets are sent
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along with the main media stream:
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@option{inputbw} * (100 + @option{oheadbw}) / 100
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if @option{inputbw} is not set while @option{maxbw} is set to
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relative (0), the actual input rate is evaluated inside
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the library. Default value is 0.
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@item iptos=@var{tos}
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IP Type of Service. Applies to sender only. Default value is 0xB8.
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@item ipttl=@var{ttl}
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IP Time To Live. Applies to sender only. Default value is 64.
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@item listen_timeout
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Set socket listen timeout.
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@item maxbw=@var{bytes/seconds}
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Maximum sending bandwidth, in bytes per seconds.
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-1 infinite (CSRTCC limit is 30mbps)
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0 relative to input rate (see @option{inputbw})
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>0 absolute limit value
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Default value is 0 (relative)
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@item mode=@var{caller|listener|rendezvous}
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Connection mode.
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@option{caller} opens client connection.
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@option{listener} starts server to listen for incoming connections.
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@option{rendezvous} use Rendez-Vous connection mode.
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Default value is caller.
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@item mss=@var{bytes}
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Maximum Segment Size, in bytes. Used for buffer allocation
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and rate calculation using a packet counter assuming fully
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filled packets. The smallest MSS between the peers is
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used. This is 1500 by default in the overall internet.
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This is the maximum size of the UDP packet and can be
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only decreased, unless you have some unusual dedicated
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network settings. Default value is 1500.
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@item nakreport=@var{1|0}
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If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
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periodically until a lost packet is retransmitted or
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intentionally dropped. Default value is 1.
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@item oheadbw=@var{percents}
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Recovery bandwidth overhead above input rate, in percents.
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See @option{inputbw}. Default value is 25%.
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@item passphrase=@var{string}
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HaiCrypt Encryption/Decryption Passphrase string, length
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from 10 to 79 characters. The passphrase is the shared
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secret between the sender and the receiver. It is used
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to generate the Key Encrypting Key using PBKDF2
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(Password-Based Key Derivation Function). It is used
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only if @option{pbkeylen} is non-zero. It is used on
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the receiver only if the received data is encrypted.
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The configured passphrase cannot be recovered (write-only).
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@item pbkeylen=@var{bytes}
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Sender encryption key length, in bytes.
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Only can be set to 0, 16, 24 and 32.
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Enable sender encryption if not 0.
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Not required on receiver (set to 0),
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key size obtained from sender in HaiCrypt handshake.
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Default value is 0.
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@item recv_buffer_size=@var{bytes}
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Set receive buffer size, expressed in bytes.
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@item send_buffer_size=@var{bytes}
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Set send buffer size, expressed in bytes.
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@item rw_timeout
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Set raise error timeout for read/write optations.
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This option is only relevant in read mode:
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if no data arrived in more than this time
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interval, raise error.
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@item tlpktdrop=@var{1|0}
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Too-late Packet Drop. When enabled on receiver, it skips
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missing packets that have not been delivered in time and
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delivers the following packets to the application when
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their time-to-play has come. It also sends a fake ACK to
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the sender. When enabled on sender and enabled on the
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receiving peer, the sender drops the older packets that
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have no chance of being delivered in time. It was
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automatically enabled in the sender if the receiver
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supports it.
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@item tsbpddelay
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Timestamp-based Packet Delivery Delay.
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Used to absorb burst of missed packet retransmission.
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@end table
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For more information see: @url{https://github.com/Haivision/srt}.
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@section srtp
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Secure Real-time Transport Protocol.
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@ -566,6 +566,7 @@ OBJS-$(CONFIG_LIBRTMPE_PROTOCOL) += librtmp.o
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OBJS-$(CONFIG_LIBRTMPS_PROTOCOL) += librtmp.o
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OBJS-$(CONFIG_LIBRTMPT_PROTOCOL) += librtmp.o
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OBJS-$(CONFIG_LIBRTMPTE_PROTOCOL) += librtmp.o
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OBJS-$(CONFIG_LIBSRT_PROTOCOL) += libsrt.o
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OBJS-$(CONFIG_LIBSSH_PROTOCOL) += libssh.o
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OBJS-$(CONFIG_LIBSMBCLIENT_PROTOCOL) += libsmbclient.o
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@ -0,0 +1,546 @@
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/*
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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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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 tsbpddelay;
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int tlpktdrop;
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int nakreport;
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int64_t connect_timeout;
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enum SRTMode mode;
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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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{ "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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{ "tsbpddelay", "TsbPd receiver delay to absorb burst of missed packet retransmission", OFFSET(tsbpddelay), AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, .flags = D|E },
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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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{ 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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|
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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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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)
|
||||
av_log(h, AV_LOG_DEBUG, "libsrt_socket_nonblock failed\n");
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static int libsrt_listen_connect(int eid, int fd, const struct sockaddr *addr, socklen_t addrlen, int timeout, URLContext *h, int will_try_next)
|
||||
{
|
||||
int ret;
|
||||
|
||||
if (libsrt_socket_nonblock(fd, 1) < 0)
|
||||
av_log(h, AV_LOG_DEBUG, "ff_socket_nonblock failed\n");
|
||||
|
||||
while ((ret = srt_connect(fd, addr, addrlen))) {
|
||||
ret = libsrt_neterrno(h);
|
||||
switch (ret) {
|
||||
case AVERROR(EINTR):
|
||||
if (ff_check_interrupt(&h->interrupt_callback))
|
||||
return AVERROR_EXIT;
|
||||
continue;
|
||||
case AVERROR(EINPROGRESS):
|
||||
case AVERROR(EAGAIN):
|
||||
ret = libsrt_network_wait_fd_timeout(h, eid, fd, 1, timeout, &h->interrupt_callback);
|
||||
if (ret < 0)
|
||||
return ret;
|
||||
ret = srt_getlasterror(NULL);
|
||||
srt_clearlasterror();
|
||||
if (ret != 0) {
|
||||
char buf[128];
|
||||
ret = AVERROR(ret);
|
||||
av_strerror(ret, buf, sizeof(buf));
|
||||
if (will_try_next)
|
||||
av_log(h, AV_LOG_WARNING,
|
||||
"Connection to %s failed (%s), trying next address\n",
|
||||
h->filename, buf);
|
||||
else
|
||||
av_log(h, AV_LOG_ERROR, "Connection to %s failed: %s\n",
|
||||
h->filename, buf);
|
||||
}
|
||||
default:
|
||||
return ret;
|
||||
}
|
||||
}
|
||||
return ret;
|
||||
}
|
||||
|
||||
static int libsrt_setsockopt(URLContext *h, int fd, SRT_SOCKOPT optname, const char * optnamestr, const void * optval, int optlen)
|
||||
{
|
||||
if (srt_setsockopt(fd, 0, optname, optval, optlen) < 0) {
|
||||
av_log(h, AV_LOG_ERROR, "failed to set option %s on socket: %s\n", optnamestr, srt_getlasterror_str());
|
||||
return AVERROR(EIO);
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* - The "POST" options can be altered any time on a connected socket.
|
||||
They MAY have also some meaning when set prior to connecting; such
|
||||
option is SRTO_RCVSYN, which makes connect/accept call asynchronous.
|
||||
Because of that this option is treated special way in this app. */
|
||||
static int libsrt_set_options_post(URLContext *h, int fd)
|
||||
{
|
||||
SRTContext *s = h->priv_data;
|
||||
|
||||
if ((s->inputbw >= 0 && libsrt_setsockopt(h, fd, SRTO_INPUTBW, "SRTO_INPUTBW", &s->inputbw, sizeof(s->inputbw)) < 0) ||
|
||||
(s->oheadbw >= 0 && libsrt_setsockopt(h, fd, SRTO_OHEADBW, "SRTO_OHEADBW", &s->oheadbw, sizeof(s->oheadbw)) < 0)) {
|
||||
return AVERROR(EIO);
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
/* - The "PRE" options must be set prior to connecting and can't be altered
|
||||
on a connected socket, however if set on a listening socket, they are
|
||||
derived by accept-ed socket. */
|
||||
static int libsrt_set_options_pre(URLContext *h, int fd)
|
||||
{
|
||||
SRTContext *s = h->priv_data;
|
||||
int yes = 1;
|
||||
int tsbpddelay = s->tsbpddelay / 1000;
|
||||
int connect_timeout = s->connect_timeout;
|
||||
|
||||
if ((s->mode == SRT_MODE_RENDEZVOUS && libsrt_setsockopt(h, fd, SRTO_RENDEZVOUS, "SRTO_RENDEZVOUS", &yes, sizeof(yes)) < 0) ||
|
||||
(s->maxbw >= 0 && libsrt_setsockopt(h, fd, SRTO_MAXBW, "SRTO_MAXBW", &s->maxbw, sizeof(s->maxbw)) < 0) ||
|
||||
(s->pbkeylen >= 0 && libsrt_setsockopt(h, fd, SRTO_PBKEYLEN, "SRTO_PBKEYLEN", &s->pbkeylen, sizeof(s->pbkeylen)) < 0) ||
|
||||
(s->passphrase && libsrt_setsockopt(h, fd, SRTO_PASSPHRASE, "SRTO_PASSPHRASE", &s->passphrase, sizeof(s->passphrase)) < 0) ||
|
||||
(s->mss >= 0 && libsrt_setsockopt(h, fd, SRTO_MSS, "SRTO_MMS", &s->mss, sizeof(s->mss)) < 0) ||
|
||||
(s->ffs >= 0 && libsrt_setsockopt(h, fd, SRTO_FC, "SRTO_FC", &s->ffs, sizeof(s->ffs)) < 0) ||
|
||||
(s->ipttl >= 0 && libsrt_setsockopt(h, fd, SRTO_IPTTL, "SRTO_UPTTL", &s->ipttl, sizeof(s->ipttl)) < 0) ||
|
||||
(s->iptos >= 0 && libsrt_setsockopt(h, fd, SRTO_IPTOS, "SRTO_IPTOS", &s->iptos, sizeof(s->iptos)) < 0) ||
|
||||
(tsbpddelay >= 0 && libsrt_setsockopt(h, fd, SRTO_TSBPDDELAY, "SRTO_TSBPDELAY", &tsbpddelay, sizeof(tsbpddelay)) < 0) ||
|
||||
(s->tlpktdrop >= 0 && libsrt_setsockopt(h, fd, SRTO_TLPKTDROP, "SRTO_TLPKDROP", &s->tlpktdrop, sizeof(s->tlpktdrop)) < 0) ||
|
||||
(s->nakreport >= 0 && libsrt_setsockopt(h, fd, SRTO_NAKREPORT, "SRTO_NAKREPORT", &s->nakreport, sizeof(s->nakreport)) < 0) ||
|
||||
(connect_timeout >= 0 && libsrt_setsockopt(h, fd, SRTO_CONNTIMEO, "SRTO_CONNTIMEO", &connect_timeout, sizeof(connect_timeout)) <0 )) {
|
||||
return AVERROR(EIO);
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
||||
static int libsrt_setup(URLContext *h, const char *uri, int flags)
|
||||
{
|
||||
struct addrinfo hints = { 0 }, *ai, *cur_ai;
|
||||
int port, fd = -1;
|
||||
SRTContext *s = h->priv_data;
|
||||
const char *p;
|
||||
char buf[256];
|
||||
int ret;
|
||||
char hostname[1024],proto[1024],path[1024];
|
||||
char portstr[10];
|
||||
int open_timeout = 5000000;
|
||||
int eid;
|
||||
|
||||
eid = srt_epoll_create();
|
||||
if (eid < 0)
|
||||
return libsrt_neterrno(h);
|
||||
s->eid = eid;
|
||||
|
||||
av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname),
|
||||
&port, path, sizeof(path), uri);
|
||||
if (strcmp(proto, "srt"))
|
||||
return AVERROR(EINVAL);
|
||||
if (port <= 0 || port >= 65536) {
|
||||
av_log(h, AV_LOG_ERROR, "Port missing in uri\n");
|
||||
return AVERROR(EINVAL);
|
||||
}
|
||||
p = strchr(uri, '?');
|
||||
if (p) {
|
||||
if (av_find_info_tag(buf, sizeof(buf), "timeout", p)) {
|
||||
s->rw_timeout = strtol(buf, NULL, 10);
|
||||
}
|
||||
if (av_find_info_tag(buf, sizeof(buf), "listen_timeout", p)) {
|
||||
s->listen_timeout = strtol(buf, NULL, 10);
|
||||
}
|
||||
}
|
||||
if (s->rw_timeout >= 0) {
|
||||
open_timeout = h->rw_timeout = s->rw_timeout;
|
||||
}
|
||||
hints.ai_family = AF_UNSPEC;
|
||||
hints.ai_socktype = SOCK_DGRAM;
|
||||
snprintf(portstr, sizeof(portstr), "%d", port);
|
||||
if (s->mode == SRT_MODE_LISTENER)
|
||||
hints.ai_flags |= AI_PASSIVE;
|
||||
ret = getaddrinfo(hostname[0] ? hostname : NULL, portstr, &hints, &ai);
|
||||
if (ret) {
|
||||
av_log(h, AV_LOG_ERROR,
|
||||
"Failed to resolve hostname %s: %s\n",
|
||||
hostname, gai_strerror(ret));
|
||||
return AVERROR(EIO);
|
||||
}
|
||||
|
||||
cur_ai = ai;
|
||||
|
||||
restart:
|
||||
|
||||
fd = srt_socket(cur_ai->ai_family, cur_ai->ai_socktype, 0);
|
||||
if (fd < 0) {
|
||||
ret = libsrt_neterrno(h);
|
||||
goto fail;
|
||||
}
|
||||
|
||||
if ((ret = libsrt_set_options_pre(h, fd)) < 0) {
|
||||
goto fail;
|
||||
}
|
||||
|
||||
/* Set the socket's send or receive buffer sizes, if specified.
|
||||
If unspecified or setting fails, system default is used. */
|
||||
if (s->recv_buffer_size > 0) {
|
||||
srt_setsockopt(fd, SOL_SOCKET, SRTO_UDP_RCVBUF, &s->recv_buffer_size, sizeof (s->recv_buffer_size));
|
||||
}
|
||||
if (s->send_buffer_size > 0) {
|
||||
srt_setsockopt(fd, SOL_SOCKET, SRTO_UDP_SNDBUF, &s->send_buffer_size, sizeof (s->send_buffer_size));
|
||||
}
|
||||
if (s->mode == SRT_MODE_LISTENER) {
|
||||
// multi-client
|
||||
if ((ret = libsrt_listen(s->eid, fd, cur_ai->ai_addr, cur_ai->ai_addrlen, h, open_timeout / 1000)) < 0)
|
||||
goto fail1;
|
||||
fd = ret;
|
||||
} else {
|
||||
if (s->mode == SRT_MODE_RENDEZVOUS) {
|
||||
ret = srt_bind(fd, cur_ai->ai_addr, cur_ai->ai_addrlen);
|
||||
if (ret)
|
||||
goto fail1;
|
||||
}
|
||||
|
||||
if ((ret = libsrt_listen_connect(s->eid, fd, cur_ai->ai_addr, cur_ai->ai_addrlen,
|
||||
open_timeout / 1000, h, !!cur_ai->ai_next)) < 0) {
|
||||
if (ret == AVERROR_EXIT)
|
||||
goto fail1;
|
||||
else
|
||||
goto fail;
|
||||
}
|
||||
}
|
||||
if ((ret = libsrt_set_options_post(h, fd)) < 0) {
|
||||
goto fail;
|
||||
}
|
||||
|
||||
h->is_streamed = 1;
|
||||
s->fd = fd;
|
||||
|
||||
freeaddrinfo(ai);
|
||||
return 0;
|
||||
|
||||
fail:
|
||||
if (cur_ai->ai_next) {
|
||||
/* Retry with the next sockaddr */
|
||||
cur_ai = cur_ai->ai_next;
|
||||
if (fd >= 0)
|
||||
srt_close(fd);
|
||||
ret = 0;
|
||||
goto restart;
|
||||
}
|
||||
fail1:
|
||||
if (fd >= 0)
|
||||
srt_close(fd);
|
||||
freeaddrinfo(ai);
|
||||
return ret;
|
||||
}
|
||||
|
||||
static int libsrt_open(URLContext *h, const char *uri, int flags)
|
||||
{
|
||||
SRTContext *s = h->priv_data;
|
||||
const char * p;
|
||||
char buf[256];
|
||||
|
||||
if (srt_startup() < 0) {
|
||||
return AVERROR_UNKNOWN;
|
||||
}
|
||||
|
||||
/* SRT options (srt/srt.h) */
|
||||
p = strchr(uri, '?');
|
||||
if (p) {
|
||||
if (av_find_info_tag(buf, sizeof(buf), "maxbw", p)) {
|
||||
s->maxbw = strtoll(buf, NULL, 0);
|
||||
}
|
||||
if (av_find_info_tag(buf, sizeof(buf), "pbkeylen", p)) {
|
||||
s->pbkeylen = strtol(buf, NULL, 10);
|
||||
}
|
||||
if (av_find_info_tag(buf, sizeof(buf), "passphrase", p)) {
|
||||
s->passphrase = av_strndup(buf, strlen(buf));
|
||||
}
|
||||
if (av_find_info_tag(buf, sizeof(buf), "mss", p)) {
|
||||
s->mss = strtol(buf, NULL, 10);
|
||||
}
|
||||
if (av_find_info_tag(buf, sizeof(buf), "ffs", p)) {
|
||||
s->ffs = strtol(buf, NULL, 10);
|
||||
}
|
||||
if (av_find_info_tag(buf, sizeof(buf), "ipttl", p)) {
|
||||
s->ipttl = strtol(buf, NULL, 10);
|
||||
}
|
||||
if (av_find_info_tag(buf, sizeof(buf), "iptos", p)) {
|
||||
s->iptos = strtol(buf, NULL, 10);
|
||||
}
|
||||
if (av_find_info_tag(buf, sizeof(buf), "inputbw", p)) {
|
||||
s->inputbw = strtoll(buf, NULL, 10);
|
||||
}
|
||||
if (av_find_info_tag(buf, sizeof(buf), "oheadbw", p)) {
|
||||
s->oheadbw = strtoll(buf, NULL, 10);
|
||||
}
|
||||
if (av_find_info_tag(buf, sizeof(buf), "tsbpddelay", p)) {
|
||||
s->tsbpddelay = strtol(buf, NULL, 10);
|
||||
}
|
||||
if (av_find_info_tag(buf, sizeof(buf), "tlpktdrop", p)) {
|
||||
s->tlpktdrop = strtol(buf, NULL, 10);
|
||||
}
|
||||
if (av_find_info_tag(buf, sizeof(buf), "nakreport", p)) {
|
||||
s->nakreport = strtol(buf, NULL, 10);
|
||||
}
|
||||
if (av_find_info_tag(buf, sizeof(buf), "connect_timeout", p)) {
|
||||
s->connect_timeout = strtol(buf, NULL, 10);
|
||||
}
|
||||
if (av_find_info_tag(buf, sizeof(buf), "mode", p)) {
|
||||
if (!strcmp(buf, "caller")) {
|
||||
s->mode = SRT_MODE_CALLER;
|
||||
} else if (!strcmp(buf, "listener")) {
|
||||
s->mode = SRT_MODE_LISTENER;
|
||||
} else if (!strcmp(buf, "rendezvous")) {
|
||||
s->mode = SRT_MODE_RENDEZVOUS;
|
||||
} else {
|
||||
return AVERROR(EIO);
|
||||
}
|
||||
}
|
||||
}
|
||||
return libsrt_setup(h, uri, flags);
|
||||
}
|
||||
|
||||
static int libsrt_read(URLContext *h, uint8_t *buf, int size)
|
||||
{
|
||||
SRTContext *s = h->priv_data;
|
||||
int ret;
|
||||
|
||||
if (!(h->flags & AVIO_FLAG_NONBLOCK)) {
|
||||
ret = libsrt_network_wait_fd_timeout(h, s->eid, s->fd, 0, h->rw_timeout, &h->interrupt_callback);
|
||||
if (ret)
|
||||
return ret;
|
||||
}
|
||||
|
||||
ret = srt_recvmsg(s->fd, buf, size);
|
||||
if (ret < 0) {
|
||||
ret = libsrt_neterrno(h);
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static int libsrt_write(URLContext *h, const uint8_t *buf, int size)
|
||||
{
|
||||
SRTContext *s = h->priv_data;
|
||||
int ret;
|
||||
|
||||
if (!(h->flags & AVIO_FLAG_NONBLOCK)) {
|
||||
ret = libsrt_network_wait_fd_timeout(h, s->eid, s->fd, 1, h->rw_timeout, &h->interrupt_callback);
|
||||
if (ret)
|
||||
return ret;
|
||||
}
|
||||
|
||||
ret = srt_sendmsg(s->fd, buf, size, -1, 0);
|
||||
if (ret < 0) {
|
||||
ret = libsrt_neterrno(h);
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static int libsrt_close(URLContext *h)
|
||||
{
|
||||
SRTContext *s = h->priv_data;
|
||||
|
||||
srt_close(s->fd);
|
||||
|
||||
srt_epoll_release(s->eid);
|
||||
|
||||
srt_cleanup();
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int libsrt_get_file_handle(URLContext *h)
|
||||
{
|
||||
SRTContext *s = h->priv_data;
|
||||
return s->fd;
|
||||
}
|
||||
|
||||
static const AVClass libsrt_class = {
|
||||
.class_name = "libsrt",
|
||||
.item_name = av_default_item_name,
|
||||
.option = libsrt_options,
|
||||
.version = LIBAVUTIL_VERSION_INT,
|
||||
};
|
||||
|
||||
const URLProtocol ff_libsrt_protocol = {
|
||||
.name = "srt",
|
||||
.url_open = libsrt_open,
|
||||
.url_read = libsrt_read,
|
||||
.url_write = libsrt_write,
|
||||
.url_close = libsrt_close,
|
||||
.url_get_file_handle = libsrt_get_file_handle,
|
||||
.priv_data_size = sizeof(SRTContext),
|
||||
.flags = URL_PROTOCOL_FLAG_NETWORK,
|
||||
.priv_data_class = &libsrt_class,
|
||||
};
|
||||
|
|
@ -65,6 +65,7 @@ extern const URLProtocol ff_librtmpe_protocol;
|
|||
extern const URLProtocol ff_librtmps_protocol;
|
||||
extern const URLProtocol ff_librtmpt_protocol;
|
||||
extern const URLProtocol ff_librtmpte_protocol;
|
||||
extern const URLProtocol ff_libsrt_protocol;
|
||||
extern const URLProtocol ff_libssh_protocol;
|
||||
extern const URLProtocol ff_libsmbclient_protocol;
|
||||
|
||||
|
|
|
|||
Loading…
Reference in New Issue