AVFormatContext::oformat was made const on April 27, 2021 [1]. If we
respect the constness of AVOutputFormat and do not cast results from
FFmpeg functions to non-const, we cannot modify the results after the
fact. Our choices are either to cast them to non-const (and presumably
have them implicitly casted back to const on later function calls), or
only try to modify the results in versions of FFmpeg where these are not
expected to be const.
Instead of relying on casts, we can set the encoder values in the
ffmpeg_cfg struct, which are later passed to new_stream.
Also modifies deps/media-playback. Removes compiler warnings.
Some avformat functions return const AV(In/Out)putFormat per [1], so
ifdef as needed.
[1]: 56450a0ee4
lavf 59.0.100 avformat.h
avformat: Constify the API wrt AV(In|Out)putFormat
Also constify AVProbeData.
When video is disabled, no audio was sent to the container so that the
audio was not saved. Prior to this change, all audio was discarded until
the first video packet arrives. This change limits to discard audio only
if video is available.
Some codecs have no recommended pix_fmts so we must check that pix_fmts
is not null before trying to find a good match. This just prays whatever
OBS is configure for is a valid format when users choose these oddball
codecs. When its not you should get a normal recording fail instead of a
crash now.
fixes#3031
Attempt to simply shutdown sequence to avoid race between vidoe and
write threads. Can no longer join on write thread from write thread.
Also use codecpar instead of deprecated codec member.
Add support for VIDEO_CS_SRGB.
Use SMPTE 170M for 601 instead of undef/BT470BG. This is the
American/Chromium way of handling 601.
Add color metadata to the container file as well via ffmpeg-mux.
Fix an issue in the way AVFrames were handled in the FFmpeg
encoder plugin, which could lead to tearing in the encoded
video due to data races in the AVFrame and AVBuffer.
This is fixed by calling av_frame_make_writable which ensures
the frame and its associated buffer are writable. If its not,
it will copy the AVFrame, create a new AVBuffer for it and
decrease the refcount of the old AVFrame and AVBuffer.
This way OBS always ends up with a usable buffer to write into
which is not still used by the encoder while avoiding a copy
when unnecessary.
The error message in the process_packet function was prefixed with
"receive_audio" because it was previously from code in the encode_audio
function which was called from the receive_audio function. This is just
a string change to avoid being misled to believe that the error is
always audio related.
The built-in format selector failed in certain cases like UtVideo now
using a differently packeg RGB format. FFmpeg has a format selection
functionality built-in that does pick the correct format however
(avcodec_find_best_pix_fmt_of_list), so we can simply use that instead.
Certain ffmpeg parameters such as "bufsize" or "maxrate" have to be
applied to the context rather than to "priv_data". In order to make sure
options are still passed to the encoder setting set AV_OPT_SEARCH_CHILDREN.
Code submissions have continually suffered from formatting
inconsistencies that constantly have to be addressed. Using
clang-format simplifies this by making code formatting more consistent,
and allows automation of the code formatting so that maintainers can
focus more on the code itself instead of code formatting.
After you call av_frame_alloc(), ffmpeg expects you to fill in certain
fields on the frame, depending on whether it's an audio or video frame.
obs-ffmpeg did this in the two places where it allocates video frames,
but not where it allocates audio frames. On my system, using trunk
ffmpeg and the Opus codec, this causes OBS to crash while calling
avcodec_send_frame, ultimately because av_frame_copy fails due to
'dst->format < 0' (as 'format' stays at the default of -1), causing a
null pointer to be added to a buffer queue, which later gets
dereferenced.
Oddly, the fields in question can just be copied directly from
corresponding fields in the AVCodecContext, but I don't see any ffmpeg
API to automatically copy all relevant fields, and all the examples I've
seen do it by hand. So this patch does the same.
(also obs, deps/media-playback, libobs/audio-monitoring, decklink,
linux-alsa, linux-pulseaudio, mac-capture, obs-ffmpeg, win-dshow,
win-wasapi)
Default channel layout for 4 channels is 4.0 in FFmpeg.
Replacing quad with 4.0 will improve compatibility since FFmpeg has
better support of its default channel layouts.
Fixes ticket 1070.
See also
https://obsproject.com/forum/threads/ffmpeg-recording.77378/#post-330473
(related bugs).
The ffmpeg constant AVFMT_RAWPICTURE was deprecated in october 2015
and marked for removal at avformat major bump to version 58
(ffmpeg commit 34ed5c2 , oct 12, 2015).
The bump occured with commit 69b5ce6 (oct21, 2017).
The constant was subsequently removed (commit 693a11b, oct 26 2017).
It was removed from obs-studio with commit d670d7b (from me).
But the code block which was executed with this constant was not
removed, causing issues with ffmpeg output.
The commit fixes the issue for old ffmpeg builds as well as new ones.
The constant is reintegrated for avformat major version < 58 and removed
for version >= 58 (along with its accompanying code).
Thanks to J Lowe for help in solving the bug.
(tested on win 10, macos 10.13, ubuntu 17.10 with ffmpeg head & ffmpeg
3.4.1)
(This commit also modifies the following modules: UI,
deps/media-playback, coreaudio-encoder, decklink, linux-alsa,
linux-pulseaudio, mac-capture, obs-ffmpeg, obs-filters, obs-libfdk,
obs-outputs, win-dshow, and win-wasapi)
Adds surround sound audio support to the core, core plugins, and user
interface.
Compatible streaming services: Twitch, FB 360 live
Compatible protocols: rtmp / mpeg-ts tcp udp
Compatible file formats: mkv mp4 ts (others untested)
Compatible codecs: ffmpeg aac, fdk_aac, CoreAudio aac,
opus, vorbis, pcm (others untested).
Tested streaming servers: wowza, nginx
HLS, mpeg-dash : surround passthrough
Html5 players tested with live surround:
videojs, mediaelement, viblast (hls+dash), hls.js
Decklink: on win32, swap channels order for 5.1 7.1
(due to different channel mapping on wav, mpeg, ffmpeg)
Audio filters: surround working.
Monitoring: surround working (win macOs linux (pulse-audio)).
VST: stereo plugins keep in general only the first two channels.
surround plugins should work (e.g. mcfx does).
OS: win, macOs, linux (alsa, pulse-audio).
Misc: larger audio bitrates unlocked to accommodate more channels
NB: mf-aac only supports mono and stereo + 5.1 on win 10
(not implemented due to lack of usefulness)
Closesjp9000/obs-studio#968
(This commit also modifies the deps/media-playback, obs-ffmpeg, and
win-dshow modules)
More fixes due to ffmpeg renaming some constants and deprecating
AVFMT_RAWPICTURE and AV_PIX_FMT_VDA_VLD.
Latter replaced by AV_PIX_FMT_VIDEOTOOLBOX per ffmpeg dev advice.
Closesjp9000/obs-studio#1061
This reverts commit bd70e73c25b0ec18373c9d2270ebe386c304076c.
Turns out the commit was due to a miscommunication -- the commit it was
fixing actually worked fine, and this fix was unnecessary.
(Note: This commit also modifies obs-ffmpeg and obs-outputs)
API Changed:
obs_output_info::void (*stop)(void *data);
To:
obs_output_info::void (*stop)(void *data, uint64_t ts);
This fixes the long-time design flaw where obs_output_stop and the
output 'stop' callback would just shut down the output without
considering the timing of when obs_output_stop was used, discarding any
possible buffering and causing the output to get cut off at an
unexpected timing.
The 'stop' callback of obs_output_info now takes a timestamp with the
expectation that the output will use that timestamp to stop output data
in accordance to that timing. obs_output_stop now records the timestamp
at the time that the function is called and calls the 'stop' callback
with that timestamp. If needed, obs_output_force_stop will still stop
the output immediately without buffering.
For some reason in the FFmpeg output, this AVCodecContext variable is
being set to 1 by FFmpeg itself somewhere, and it's causing a massive
slowdown when encoding with FFmpeg directly. This should be set to 0 to
specify to use as many threads as necessary.
This also adds the ability to detect whether it stopped due to lack of
space or not -- particularly useful for the FFmpeg output due to
lossless file format support.
For the FFmpeg output, the encoder ids are sort of superfluous. They
really should be optional. If they're not set, it should use the
encoder name string instead to determine the ids automatically.
API changed from:
obs_source_info::get_name(void)
obs_output_info::get_name(void)
obs_encoder_info::get_name(void)
obs_service_info::get_name(void)
API changed to:
obs_source_info::get_name(void *type_data)
obs_output_info::get_name(void *type_data)
obs_encoder_info::get_name(void *type_data)
obs_service_info::get_name(void *type_data)
This allows the type data to be used when getting the name of the
object (useful for plugin wrappers primarily).
NOTE: Though a parameter was added, this is backward-compatible with
older plugins due to calling convention. The new parameter will simply
be ignored by older plugins, and the stack (if used) will be cleaned up
by the caller.
This is used by some muxers that set AVFMT_NOFILE and doesn't seem to
hurt muxers that don't set it; notable this makes the hls muxer output
its m3u8 playlist with the proper filename in the proper directory
This particularly affected audio encoding, audio encoding previously
would count samples and use it to create an encoding timestamp, but
because I was using a standard integer (which is 32bit by default on
x86), it would max out at about 0x7FFFFFFF samples, which is about 12
hours of samples at 48000 sample rate. After that, it would start going
into negative territory (overflowing). By changing it to int64_t, it
will make it so that audio at 48000 samples per second would only be
able to overflow after about.. 6.09 million years. In other words,
this should fix the issue for good.
In the settings if you select default container then the
format becomes null. If null then audio/video codec ids should
not be set on the output format as they will both be
AV_CODEC_ID_NONE causing a context with no streams specified
to be created (error).
