Adds functions to turn on audio monitoring to allow the user to hear
playback of an audio source over the user's speaker. It can be set to
turn off monitoring and only output to stream, or it can be set to
output only to monitoring, or it can be set to both.
On windows, audio monitoring uses WASAPI. Windows also is capable of
syncing the audio to the video according to when the video frame itself
was played.
On mac, it uses AudioQueue.
On linux, it's not currently implemented and won't do anything (to be
implemented).
Fixes a bug that would allow possible infinite recursion within a source
tree. To fix this, inactive sources must be enumerated as well in order
to prevent infinite recursion.
Commit 53955301a2 introduced a async source texture copy bug due to
creating a new case in a switch without adding a break to the one above
it, causing it to execute both cases by mistake.
Because D3D11 specifically does not support an L8 texture format (you
have to use a shader swizzle), manually convert Y800 signals to RGBX
instead. This also fixes a bug where Y800 signals will render red.
Closesjp9000/obs-studio#718
If an async source is cropped on one side, then when the program is
restarted and the source is loaded from file, the async source will
start out with a width/height of zero. This will cause the async source
to not be drawn if cropping or scale filtering is added to the scene
item, because it has to be rendered to a texture first. However, the
source cannot reset its size until it's drawn, so it leaves it in
perpetual state of having a 0x0 size.
This fixes that problem by ensuring that the async source size is always
reset even when not being rendered.
Closejp9000/obs-studio#686
When a scene is duplicated the filters on the scene were not copied to
the new scene. This causes that a temporary copy of a scene renders
differently in the program than in the preview when using studio mode.
(Note: This commit also modifies coreaudio-encoder, win-capture, and
win-mf modules)
This reduces logging to the user's log file. Most of the things
specified are not useful for examining log files, and make reading log
files more painful.
The things that are useful to log should be up to the front-end to
implement. The core and core plugins should have minimal mandatory
logging.
The active_refs and show_refs variable would only increment/decrement
their children if their values were 1 and 0, which means that in the
case of scenes within scenes, sub-sources of scenes within scenes would
end up having the wrong ref values.
When using GPU conversion for 4:2:0 frames on async video sources, it
would create a texture bigger than necessary and try to copy too much
data from the frame, resulting in a crash.
When a transition is a sub-source of another source, it would not call
the transition's active source enum function, meaning that any sources
the transition had would not increment their active/showing refs (it
would only be called when activating the transition directly before).
That would result in negative/invalid active/showing refs on its
sub-sources, causing them to become permanently active/inactive and/or
permanently showing/hidden.
(Note: this commit also modifies the obs-filters and test-input modules)
Changes the obs_source_process_filter_begin return type so that it
returns true/false to indicate that filter processing should or should
not continue (for example if the filter is bypassed or if there's some
other sort of issue that causes the filtering to fail)
The source shouldn't be inserted into obs->data.first_audio_source until it's
fully initialized, or other threads will access source->control and
dereference an uninitialized pointer.
This patch fixes a specific crash where if the user named a filter the
same name as an input source that already existed in the system, scene
item loading code could find the filter with the same name instead of
the source, and mistakenly use it as the scene item's source directly.
This would cause a crash when trying to render that filter as a regular
source.
Marking filters as private is a temporary and simple workaround to the
solution. Filters are currently not meant to be found via the main
enumeration/search functions, which is a design flaw (lack of
consistency). In future major API revisions of libobs, filters should
be reworked to act as sources, with the sources they filter as
sub-sources ideally.
Additionally, the concept of "private context objects" and "primary
lists of context objects" in the back-end should probably also be
removed, allowing the font-end (or optional separate API layers) to
control all primary lists of obs context objects. These minor issues
that occur ultimately stem from API design flaws which need to be
corrected.
This crash happened when a filter was mistakenly used as a regular
source due to an unrelated bug in filter code and scene loading code.
The filter and the source it belongs to both had the same names, and the
source loading code found the filter and mistakenly used it as the
source instead of the actual source with the same name.
(Note: Also modified the obs-ffmpeg plugin module)
Allows the ability for frame data to pass 8-bit grayscale images (Y800
color format).
Closesjp9000/obs-studio#515
Adds deinterlacing API functions. Both standard and 2x variants are
supported. Deinterlacing is set via obs_source_set_deinterlace_mode and
obs_source_set_deinterlace_field_order.
This was implemented in to the core itself because deinterlacing should
happen before effect filters are processed, but after async filters are
processed. If this were added as a filter, there is the possibility
that a different filter is processed before deinterlacing, which could
mess with the result. It was also a bit easier to implement this way
due to the fact that that deinterlacing may need to have access to the
previous async frame.
Effects were split in to separate files to reduce load time (especially
for yadif shaders which take a significant amount of time to compile).
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).
(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.
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.
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.
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.
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.
Mostly only used for transitions with the intention of automatically
creating transitions which don't require configuration, returns whether
the source has any properties or not (whether it's configurable)
(Note: This commit also modifies UI)
Instead of using signals, use designated callback lists for audio
capture and audio control helpers. Signals aren't suitable here due to
the fact that signals aren't meant for things that happen every frame or
things that happen every time audio/video is received. Also prevents
audio from being allocated every time these functions are called due to
the calldata structure.
