Instead of just updating the async texture variables directly in the
source, allow the ability to pass the async texture variables via
function parameters to allow the ability to parse more than one frame to
more than one texture.
This code is primarily intended to be used to upload/convert the
"previous" async frame for the deinterlacer (if necessary).
Just creates an effect to the target variable only if its current value
is null. This will be used for deinterlacing effects to prevent having
to compile the shaders unless they're actually being used.
(Note: This commit also modifies obs-filters and text-freetype2)
This simplifies writing of effects. DrawMatrix is no longer necessary
because there are no sources that require drawing with a color matrix
other than async sources, and async sources are automatically processed
and don't defer their initial render stage to filters.
When the #include directive in in the C lexer preprocessor is
encountered, the files being included need to be relative to the
directory of the file that the include was used in.
(Note: This commit also changes the UI)
Changed:
-------------------
void obs_load_sources(obs_data_array_t *sources_list);
To:
-------------------
void obs_load_sources(obs_data_array_t *sources_list,
obs_source_load_cb callback, void *private_data);
Signals should really never be required to use to make some function
work properly. The "source_load" signal was required for the
obs_load_sources function, but it's meant more for loading private data
in the settings, not for general loading of sources.
This changes it so that a callback is explicitly required to load the
sources.
The default buffering time for audio was always 1 second before the
audio subsystem was changed, and it was always more than sufficient for
max audio buffering time
Under certain circumstances, the timing_adjust variable would cause line
1161 to continually trigger over and over again. The "loop detection"
code incorrectly made it so that any timestamp that was just simply
below the expected value would be seen as a jump. After that, the
timing_adjust variable would be set for the frame again, and then the
audio would see it as a jump again after that, and those two things
would continue endlessly. This would cause stuttering particularly with
certain devices (particularly elgato/lgp/hdpvr) where the audio/video
data are decoded and sent at varying/different/unpredictable times.
To fix this issue, it should not detect values below as jumps, but
instead should only do it for values that exceed the MAX_TS_VAR (maximum
timestamp variance) value.
If obs_source::audio_ts is set to 0 (such as by discard_if_stopped in
obs-audio.c), but the push_back variable in the source_output_audio_data
function in obs-source.c was being set to true (meaning it's within the
seamless audio smoothing threshold), it would cause it to never reset
the obs_source::audio_ts value, and thus all audio data from the source
would become perpetually ignored by the audio subsystem until there was
finally some sort of timestamp jump that caused it to call
source_output_audio_place, and thus reset obs_source::audio_ts.
obs_source::audio_ts is only reset in source_output_audio_place, not in
source_output_audio_push_back, so the most simple solution is to just
call source_output_audio_push_back is obs_source::audio_ts is 0.
This code causes audio data in general to be reset (and subsequently
deleted). It should just be marked as pending and ignored until the
data is ready. The discard_if_stopped function will serve the same
purpose if the source's audio has actually stopped.
There's technically no need to clear the audio data here, nor is there
any need to try to trick the timestamp in to a different position. It
can simple just reset the audio timing.
Prevents a possible case where audio data might be deleted when it's not
necessary to delete any.
This variable is used to detect whether audio has stopped -- if audio
stops, it detects that no new data is coming in, and resets the audio
position so that it eliminates the chance of causing the audio buffering
to go haywire if audio starts up again. However, this variable was not
being reset every time the value changes, which it should.
Sometimes the A and B sources of a transition would a large difference
in their timestamps, and the calculation of where to start the audio
data for one of the sources could be above the tick size, which could
cause a crash.
If the circular audio buffer of the source has data remaining that's
less than the audio frame tick count (1024 frames), it would just leave
that audio data on the source without discarding it. However, this
could cause audio buffering to increase unnecessarily under certain
circumstances (when the next audio timestamp is within the timestamp
jump window), so it would append data to that circular buffer despite
the audio stopping that long ago, causing audio buffering to have to
increase to compensate.
Instead, just discard pending audio if it hasn't been written to. In
other words, if the audio has stopped and there's insufficient audio
left to continue processing.
With the new audio subsystem, audio buffering is minimal at all times.
However, when the audio buffering is too small or non-existent, it would
cause the audio encoders to start with a timestamp that was actually
higher than the first video frame timestamp. Video would have some
inherent buffering/delay, but then audio could return and encode almost
immediately. This created a possible window of empty time between the
first encoded video packet and the first encoded audio packet, where as
audio buffering would cause the first audio packet's timestamp to always
be way before the first video packet's timestamp. It would then
incorrectly assume the two starting points were in sync.
So instead of assuming the audio data is always first, this patch makes
video wait for audio data comes in, and conversely buffers audio data
until video comes in, and tries to find a starting point within that
video data instead, ensuring a synced starting point whether audio
buffering is active or not.
When starting a multi-track output, attempt to pair the video encoder
with one of the audio encoders to ensure that the video and audio
encoders start as close together in time as possible. This ensures the
best possible audio/video syncing point when using multi-track audio
output.
When using multi-track audio, encoders cannot be paired like they can
when only using a single audio track with video, so it has to choose the
best point in the interleaved buffer as the "starting point", and if the
encoders start up at different times, it has to prune that data and wait
to start the output on the next video keyframe. When the audio encoders
started up, there was the case where the encoders would take some time
to load, and it would cause the pruning code to wait for the next
keyframe to ensure startup syncing.
Starting the audio encoders before starting the video encoder should
reduce the possibility of that happening in a multi-track scenario.
In a multi-track scenario it was not taking in to consideration the
possibility of secondary audio tracks, which could have caused desync on
some of the audio tracks.
The seamless audio looping code would erroneously trigger for things
that weren't loops, causing the audio data to continually push back and
ignore timestamps, thus going out of sync.
There does need to be loop handling code, but due to the fact that other
things may need to trigger this code, it's best just to clear the audio
data and start from a fresh sync point. Unfortunately for the case of
loops, this means the window in which audio data loops and video frames
loop need to be muted.
Fixes an issue where audio data would not be popped if they were not
activated/presenting. This would cause the audio subsystem to
needlessly buffer when they were reactivated again. Rendering all audio
sources (excuding composite/filter sources) helps ensure that audio data
is always popped and not left to pile up.
A comment that serves as a reminder to anyone who might need to edit the
scene code. If the graphics mutex must be locked, it must be locked
first before entering the scene mutexes, or outside of the scene
mutexes.
This fixes an age-old issue where audio samples could be lost or audio
could temporarily go out of sync in the case of looping videos. When
audio/video data is looping, there's a window between when the audio
data resets its timestamp value and when the video data resets its
timestamp value. This method simply pushes back the audio data while in
that window and does not modify sync, and when it detects that its out
of the loop window it simply forces a resync of the audio data in the
circular buffer.
This ensures that minimal audio data is lost in the loop process, and
minimizes the likelihood of any sort of sync issues associated with
looping.
Instead of applying the resampler offset right away (to each audio
packet), apply the resampler offset when the timestamps are converted to
system timestamps. This fixes an issue where if audio timestamps reset
to 0 (for whatever reason), the offset would cause the timestamp to go
in to the negative.
(Note: This commit also modifies the UI)
Allows the ability to duplicate sources fully copied, and/or have the
scene and its duplicates be private sources
