This reverts commit 22aa66a6eb520444c492f6a7c595fd7a5de98bdc.
Apparently, starting audio on the fly like this can introduce latency in
to the audio playback, so for now revert it. It was a bit of a
precautionary thing rather than an actual fix anyway, so it probably
wasn't all that necessary to begin with.
We really shouldn't be resetting duplicator state as part of gs_flush.
gs_begin_scene is not ideal because it is called twice per frame, and
only after duplicators have been ticked. Even though it makes no
user-facing difference, it makes more logical sense to reset at the top
of the frame than the bottom.
This prevents audio monitoring from actually initializing unless audio
is actually played back through the source. This prevents many browser
sources from initializing audio monitoring all at once needlessly if
audio is not being rerouted to OBS.
Adds functions to allow sources to inform the UI whether the audio is
currently active or not. Allows the ability to turn on/off the items in
the mixer.
If for whatever reason the format is swapped between BGRA/BGRX in an
async filter, swap the texture to compensate for that. This allows
plugins to change the format if necessary.
Originally when the audio_submix function was created, it used all mixer
tracks, but at a certain point that was removed because it can only use
the first track, so some older code was unintentionally left over,
causing the same code to be executed 6 times mistakenly. This cleans
that up by removing the unnecessary function parameter and for loop.
When an unpause occurs, it takes an audio segment and splits it at the
exact point corresponding to the pause timestamp, and then it's supposed
to only send the ending part of the split. However, the audio pointers
were not being incremented, therefore it was sending the front of the
audio segment to instead of the back of the audio segment by mistake.
When pause has been activated, the video_pause_check() function is used
when receiving raw frames in order to filter out frames that are in the
pause window, that way they aren't sent to the encoder or output.
However, when pause was enabled, it was unintentionally filtering out
some frames before the specified starting timestamp as well, causing
extra video data to get cut out prematurely. This fixes that issue.
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 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
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.
Fixes an issue where the browser source settings will continually reset
pre-24. Note that this is not 23.2.2, but the version is being
temporarily updated in order to fix the issue for the release candidate
build.
