When a pause/unpause occurs, a timestamp is set and the actual
pause/unpause does not occur until the output/encoders reach the
specified timestamps. Do not allow pausing/unpausing unless that point
has been reached with all encoders of an encoded output or the output
itself when using a raw output.
This fixes a bug where pause data could get corrupted if
pausing/unpausing too fast, because the audio/video encoders aren't
necessarily synchronized and although one encoder may have unpaused, the
other encoder(s) may not have yet. Checking all encoders first before
allowing a pause/unpause ensures that doesn't occur.
Audio latency can get really low, and if it's low enough, the timestamp
can be passed by the audio subsystem before it's had a chance to pause
with it. So instead, make the pause have a little bit of extra delay to
ensure that doesn't occur.
This fixes crashes that could occur while processing Qt events during a
source's destruction.
Originally the processEvents code was added to make sure that any window
messages being sent to the browsers were being processed by Qt.
However, that was only necessary for browser panels as they are actually
associated with windows/views. Browser sources are OCR, thus they do
not need this.
Cuda appears to be a bit buggy and can result in corruption where as
it's not being seen with other hardware decoders. Remove cuda for now
as a hardware decoder to prevent the possibility of corruption.
Fixes a bug where hardware frames would not transfer to RAM for the last
few frames of a video. The 'ret' variable can be 0 even though there
are still frames ready.
Fixes same issue as 5c04876de1, but reverts that and changes it so that
external_begin_frame_enabled is not used instead, which seems to fix
that issue.
This fixes a race condition where the audio/video backends/threads may
start using sources before their obs_source_info::create function has
been called.
Also increase weight precision by premultiplying UV in VS.
Intel HD Graphics 530, Intel GPA, SetStablePowerState
256x224 -> 1323x1080: 1221 us -> 1020 us
The hardware accelerated decoder context needs to be explicitly unrefed
when it's no longer in use, otherwise it and many resources associated
with it will leak.
When hardware accelerated decoding is enabled, sometimes it can't
initialize for whatever reason, so it will fall back to software on its
own. When this occurs, it will not use the hardware pixel format on the
frame; instead it will defer to a standard format on the frame. So if
the frame format does not match the expected format, assume software
decoding. (This was also what the hw-decode.c FFmpeg example did if the
format did not match the expected format)
When the streaming audio track was separated from the recording tracks
in advanced output mode in be8c06334, it mistakenly removed the opus
audio encoder code when FTL is used. This restores that code.
CEF outputs multiple audio streams at once, and OBS was only able to
handle one at a time. This fixes it by using audio lines for each CEF
audio stream, and mixes them together itself.
Adds the "audio_line" internal source type as a bare source type for the
sole purpose of outputting audio, and the obs_source_info::audio_mix
callback which allows mixing of those audio lines, which is then treated
as normal audio for the source. Audio line objects should be added as
sub-sources when multiple audio lines from a single source are needed,
then mixed together with the audio_mix callback.
The difference between the new obs_source_info::audio_mix callback and
obs_source_info::audio_render is that obs_source_info::audio_mix (along
with the audio_line source) are only one track, and it outputs audio to
the source automatically via obs_source_output_audio() when the call
completes. This allows the mixed audio to be treated like a normal
source's audio, in that you can filter it, change its volume, or monitor
it.
This change was necessary because the CEF (used with the browser source)
outputs multiple audio streams at once to a single browser source, so
it's the program's responsibility to mix those streams together itself.
The dynamic bitrate operates based upon estimating the current bitrate
output, and then adjusting the bitrate on the fly as necessary when
congestion is detected as a replacement for dropping frames.
This may still need adjustment, as it is difficult to accurately emulate
real-world frame drop scenarios. This does not currently drop frames at
all, and because of that, very high congestion may cause additional
stream delay to viewers (because data will be buffered), but from
limited testing, most congestion will not cause that and it can safely
recover pretty quickly without adding significant delay.
When doing the bitrate limit test, it can be useful to have the ability
to change the current maximum bitrate limit. This adds the ability to
press keys on windows (numpad 0-6) to change between bitrates. Numpad 0
being no limit, 1 being 1000, 2 being 2000, etc.
