There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.
Keep it for external users in order to not cause breakages.
Also improve the other headers a bit while just at it.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This commit does for AVInputFormat what commit
59c9dc82f4 did for AVOutputFormat:
It adds a new type FFInputFormat, moves all the internals
of AVInputFormat to it and adds a now reduced AVInputFormat
as first member.
This does not affect/improve extensibility of both public
or private fields for demuxers (it is still a mess due to lavd).
This is possible since 50f34172e0
(which removed the last usage of an internal field of AVInputFormat
in fftools).
(Hint: tools/probetest.c accesses the internals of FFInputFormat
as well, but given that it is a testing tool this is not considered
a problem.)
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Fixes server compatibility issues with rtspclientsink GStreamer plugin.
>From specification:
RFC 7826 "Real-Time Streaming Protocol Version 2.0" (https://datatracker.ietf.org/doc/html/rfc7826), section 18.54:
mode: The mode parameter indicates the methods to be supported for
this session. The currently defined valid value is "PLAY". If
not provided, the default is "PLAY". The "RECORD" value was
defined in RFC 2326; in this specification, it is unspecified
but reserved. RECORD and other values may be specified in the
future.
RFC 2326 "Real Time Streaming Protocol (RTSP)" (https://datatracker.ietf.org/doc/html/rfc2326), section 12.39:
mode:
The mode parameter indicates the methods to be supported for
this session. Valid values are PLAY and RECORD. If not
provided, the default is PLAY.
mode=receive was always like this, from the initial commit 'a8ad6ffa rtsp: Add listen mode'.
For comparison, Wowza was used to push RTSP stream to. Both GStreamer and FFmpeg had no issues.
Here is the capture of Wowza responding to SETUP request:
200 OK
CSeq: 3
Server: Wowza Streaming Engine 4.8.26+4 build20231212155517
Cache-Control: no-cache
Expires: Mon, 15 Jan 2024 19:40:31 GMT
Transport: RTP/AVP/UDP;unicast;client_port=11640-11641;mode=record;source=172.17.0.2;server_port=6976-6977
Date: Mon, 15 Jan 2024 19:40:31 GMT
Session: 1401457689;timeout=60
Test setup:
Server: ffmpeg -loglevel trace -y -rtsp_flags listen -i rtsp://0.0.0.0:30800/live.stream t.mp4
FFmpeg client: ffmpeg -re -i "Big Buck Bunny - FULL HD 30FPS.mp4" -c:v libx264 -f rtsp rtsp://127.0.0.1:30800/live.stream
GStreamer client: gst-launch-1.0 videotestsrc is-live=true pattern=smpte ! queue ! videorate ! videoscale ! video/x-raw,width=640,height=360,framerate=60/1 ! timeoverlay font-desc="Sans, 84" halignment=center valignment=center ! queue ! videoconvert ! tee name=t t. ! x264enc bitrate=9000 pass=cbr speed-preset=ultrafast byte-stream=false key-int-max=15 threads=1 ! video/x-h264,profile=baseline ! queue ! rsink. audiotestsrc ! voaacenc ! queue ! rsink. t. ! queue ! autovideosink rtspclientsink name=rsink location=rtsp://localhost:30800/live.stream
Test results:
modified FFmpeg client -> stock server : ok
stock FFmpeg client -> modified server : ok
modified FFmpeg client -> modified server : ok
GStreamer client -> modified server : ok
Signed-off-by: Paul Orlyk <paul.orlyk@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Unnecessary since acf63d5350adeae551d412db699f8ca03f7e76b9;
also avoids relocations.
Reviewed-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
This avoids unnecessary rebuilds of most source files if only the
list of enabled components has changed, but not the other properties
of the build, set in config.h.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is possible now that the next-API is gone.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Currently a repeating setup request (with the same stream id) will
simply overwrite rtp_handle/transport_priv without freeing the
resources first. This is fixed by closing the previous setup request.
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
As per the docs network initialization is required before ff_url_join().
Furthermore, because the ff_network_init() was skipped, this makes
one additional call to ff_network_close() if the stream exits without
errors.
The was forgotten in the origin commit of the listen mode:
a8ad6ffafe
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Andriy Gelman <andriy.gelman@gmail.com>
Since all URLContexts have the same AVOptions, such AVOptions
will be applied on the outermost context only and removed from the
dict, while they probably make sense on all contexts.
This makes sure that rw_timeout gets propagated to the innermost
URLContext (to make sure it gets passed to the tcp protocol, when
opening a http connection for instance).
Alternatively, such matching options would be kept in the dict
and only removed after the ffurl_connect call.
Signed-off-by: Martin Storsjö <martin@martin.st>
Move the OpenSSL and GnuTLS implementations to their own files. Other
than the connection code (including options) and some boilerplate, no
code is actually shared.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Move the OpenSSL and GnuTLS implementations to their own files. Other
than the connection code (including options) and some boilerplate, no
code is actually shared.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'cdcc370293a159c321e41af7f0eef141c62d698d':
rtsp: punch holes again after pause
See: 22bb5bd7a3
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When a client behind a NAT issues a pause command, and stay paused for a
long time, the router may stop the RTP/RTCP port redirection. Resend the
hole punching packets before each PLAY command to cause the router to
restart the port redirection in that case.
Move the existing code for sending the packets from the SETUP phase
to the PLAY phase.
Signed-off-by: Martin Storsjö <martin@martin.st>
When a client behind a NAT issues a pause command, and stay paused for a
long time, the router may stop the RTP/RTCP port redirection. Resend the
hole punching packets after each PLAY command to cause the router to
restart the port redirection in that case.
Signed-off-by: Gilles Chanteperdrix <gilles.chanteperdrix@xenomai.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '9108967513fcaff3d55514a7bca4c9fbba128c71':
rtspdec: Consistently use rtsp_hd_out for writing
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The ones left using av_gettime are NTP timestamps (for RTCP,
which is specified to send the actual current realtime clock
in RTCP SR packets), and the NUT muxer timestamper, which is
documented as using wallclock time.
Signed-off-by: Martin Storsjö <martin@martin.st>
Andreas Rheinhardt