Previously, we produced output with either \r\n or mixed line endings.
This was undesirable unto itself, but also made working with patches affecting
FATE output particularly challenging, especially via the mailing list.
Everything that consumes the SSA/ASS format is line-ending-agnostic,
so \n is selected to simplify git/ML usage in FATE.
Extra \r characters at the end of a packet are dropped. These are always
ignored by the renderer anyway.
The channel designation metadata should not override the number of channels.
Let's warn the user if it is inconsistent, and keep the channel layout
unspecified.
Before the conversion to the channel layout API the code only set the mask, but
never overridden the channel count, so this restores the old behaviour.
Signed-off-by: Marton Balint <cus@passwd.hu>
Existing code could have caused wrong channel order signalling or reduced
channel count if a channel designation appeared multiple times. This is
actually an old bug, but the conversion to the new channel layout API made it
visible, because now the code overrides the proper channel count with the one
calculated from the mask.
Signed-off-by: Marton Balint <cus@passwd.hu>
The demuxer opens an internal parser instance in read_timestamp(), which
requires a codec context. There is no need for it to access the FFStream
one which is used for other purposes, it can allocate its own internal
one.
This check has survived the transition to AVCodecParameters, but is no
longer relevant after it, since the codec context is no longer updated
or accessed at all from the demuxer.
It does not use the AVFormatContext at all.
Reviewed-by: Marth64 <marth64@proxyid.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Resetting the counter of used elements is enough as nothing is
ever read from the currently unused elements.
Reviewed-by: Marth64 <marth64@proxyid.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The rcwt muxer uses several counters for how much data
it has already cached: One byte counter and one counter
for how many complete blocks (of three bytes each).
These counters can become inconsistent when the muxer is
fed incomplete blocks as the muxer presumes that it is
about to write a new block at the start of each write_packet
call. E.g. sending 65535*3+1 1-byte packets (with data[0] e.g. 0x03)
will trigger an out-of-bounds write.
This patch fixes this by processing the data in complete blocks
only. This also allows to simplify the code, e.g. to remove one of
the counters.
Reviewed-by: Marth64 <marth64@proxyid.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The muxer's AVCodecContext is currently used for exactly one thing:
To store a time base in it that has been derived via heuristics
in avformat_transfer_internal_stream_timing_info(); said time base
can then be read back via av_stream_get_codec_timebase().
But one does not need a whole AVCodecContext for that, a simple
AVRational is enough.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
For muxers, the internal AVCodecContext is basically unused
except in avformat_transfer_internal_stream_timing_info()
(which sets time_base and ticks_per_frame) and
av_stream_get_codec_timebase() (a getter for time_base).
This makes ticks_per_frame write-only, so don't set it.
Also remove an always-false check for the AVCodecContext's
codec_tag.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Currently, when writing PCMA or PCMU tracks with FLV or RTMP, the
stereo flag and sample rate flag inside RTMP audio messages are
overridden, making impossible to distinguish between mono and stereo
tracks. This patch fixes the issue by restoring the same flag mechanism
of all other codecs, that takes into consideration the right channel
count and sample rate.
Signed-off-by: Alessandro Ros <aler9.dev@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
There are 6 deprecated ISO language codes that are still valid for DVDs.
This patch allows avlanguage to recognize them correctly. The codes are:
(1) "in" - legacy code for Indonesian, mapped to the modern code
(2) "iw" - legacy code for Hebrew, mapped to the modern code
(3) "ji" - legacy code for Yiddish, mapped to the modern code
(4) "jw" - legacy code for Javanese, published and used as a typoed version of "jv"
(5) "mo" - legacy code for Moldavian, mapped to the inclusive code
(6) "sh" - legacy code for Serbo-Croatian, no modern inclusive code so it is left alone
All of this can be verified from several sources including:
https://en.wikipedia.org/wiki/List_of_ISO_639_language_codes
Signed-off-by: Marth64 <marth64@proxyid.net>
Use avio_get_dyn_buf()+ffio_free_dyn_buf() instead of
avio_close_dyn_buf()+av_free(). This saves an allocation
(and memcpy) in case all the data fits in the AVIOContext's
write buffer.
Reviewed-by: James Almer <jamrial@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Fixes a regression since d9fed9df2a, where the single animated stream would
be exported twice as two independent streams.
Signed-off-by: James Almer <jamrial@gmail.com>
Fixes segfaults when trying to map a group index with a value equal to the
amount of groups in the file.
Signed-off-by: James Almer <jamrial@gmail.com>
Probably an artifact of a rebase, as this check is done below.
Fixes "Conditional jump or move depends on uninitialised value(s)" errors as
reported by Valgrind.
Signed-off-by: James Almer <jamrial@gmail.com>
The AVIAMFAudioElement and AVIAMFMixPresentation that are ultimately used
are allocated by ff_iamfdec_read_descriptors().
Fixes some memory leaks reported by Valgrind.
Signed-off-by: James Almer <jamrial@gmail.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>
Fixes server compatibility issues with rtspclientsink GStreamer plugin
Signed-off-by: Paul Orlyk <paul.orlyk@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
