When OBS is shutting down, if for some reason the filter is destroyed
before the parent source is destroyed, it would try to remove itself
from the source, but it would decrement the reference and try to destroy
itself again while already in the process of destroying itself.
So, the solution was simply to make sure that if it's removing itself
from the source that it doesn't decrement its own reference.
The "last" filter that's rendered is technically at filter index 0, so
enumeration needs to be from the last index in the list to the first
index in the list.
When rendering a filter to a texture, the target is empty and unused, so
there's no reason for blending to be on when rendering the filter to a
render target.
obs_source_process_filter tried to do everything in a single function,
but the problem is that effect parameters would not properly be
accounted for due to the way it internally draws, therefore it was
necessary to split the functions in to two, you first call
obs_source_process_filter_begin, then you set your effect parameters,
then you finally call obs_source_process_filter_end. This ensures that
when the filter is drawn, that the effect parameters are set.
When the filter chain finally reaches the source and the last filter in
the chain is set to not render directly (meaning it has to render to
texture), it would not render the source with any effect due to the fact
that it expects a filter to be present.
Adds a source callback function that is used when a filter is removed
from a source. This ensures that any data that might be associated with
that source can be cleaned up if necessary.
There are a few more conditions which need to be checked to ensure
whether we can actually bypass or not; in this particular case we are
using the YUV texture shaders, plus the image can also be flipped, so we
can't really use the bypass optimization in those situations.
These functions are primarily for use with filters, filters need to be
able to get the width/height of a target source without it necessarily
getting the post-filtered dimensions.
Previously I had it set so that it would set the async_width and
async_height variables to 0,0 if a null frame occurs, but the problem
with this is that it was inadvertently cause the frame cache to clear
when it starts receiving textures again because it detects it as a size
change.
So instead of changing async_width and async_height to 0 when a null
frame occurs, change it so that if the source is set to an inactive
state, make obs_source_get_width/_get_height return 0 for their
dimensions instead.
When a frame is processed by a filter, it comes directly from the
source's video frame cache. However, if a filter is using or processing
those frames for whatever reason, there would be no guarantee that the
frames would persist during processing, and frames could eventually be
deallocated unexpected, for example when the resolution or format
changes.
So the solution is to implement simple reference counting for the frames
so that the frames will exist until they have been released by any
source or filter that's using them.
Fix a bug where when a source first starts up its async cache, it
unintentionally resets its cache, which means that the first few frames
would be lost.
The wrong function was being used to recurse through the filter chain in
obs_source_process_filter, obs_source_get_[width/height] would get the
post-effect dimensions rather than the pre-effect dimensions.
If the audio data had the same format/samplerate as the obs audio
subsystem, it would fail to simply destroy the resampler and set it to
NULL, and then any audio data going through would use the resampler that
was being used before that, causing audio to become garbage.
This bug only started appearing when I recently changed the libobs
internal audio subsystem format to non-interleaved floating point, which
is a common format, and thus caused this bug to actually occur more
often.
I when a source has both async audio/video capability, it would ignore
audio until the video has started. There's really no need to do this,
when the video starts it'll just fix up the timing automatically.
This should fix the case where things like the media source would not be
able to play audio-only files.
Core API functions changed:
-----------------------------
EXPORT bool obs_reset_audio(struct audio_output_info *aoi);
EXPORT bool obs_get_audio_info(struct audio_output_info *aoi);
To:
-----------------------------
EXPORT bool obs_reset_audio(const struct obs_audio_info *oai);
EXPORT bool obs_get_audio_info(struct obs_audio_info *oai);
Core structure added:
-----------------------------
struct obs_audio_info {
uint32_t samples_per_sec;
enum speaker_layout speakers;
uint64_t buffer_ms;
};
Non-interleaved (planar) floating point output is standard with audio
filtering, so to prevent audio filters from having to worry about
different audio format implementations and for the sake consistency
between user interfaces, make it so that audio is always set to
non-interleaved floating point output.
This code was originally meant to skip some checks as an optimization,
but it did not account for async video sources and would call
obs_source_default_render (which is only for synchronous video sources),
this fixes the issue by just calling obs_source_video_render.
The obs_source_get_width and obs_source_get_height functions need to
account for filters that may change their width/height such as a crop
filter or something similar.
As a side effect of this commit, because these functions need to lock
the filter mutex, these functions can no longer be used with a
const-qualified obs_source_t pointer.
This function is somewhat big and is a function called conditionally,
therefore having it be inline is somewhat of a waste if it's not always
being called.
Graphics data has to be freed inside of an active graphics context,
otherwise the resources will not be freed; in the future, we should
probably make sure that the graphics subsystem automatically
asserts/warns about this scenario.
I was iterating through the obs_source::filters array to remove filters,
but every time I removed a filter, it would remove itself from the
array, therefore it would end up skipping items in the array, therefore
leaving filters unreleased. This just removes the first filter until
all filters have been removed.
For the show/hide and activate/deactivate callbacks, schedule these
callbacks to only be called from within the video thread rather than in
a separate thread. This ensures that any potential graphics activity
that occurs within them is kept in the same thread.
obs_source_inc_showing and obs_source_dec_showing are used to indicate
that a source is showing or no longer being shown in a particular area
of the program outside from the main render view.
One could use an obs_view, but I felt like it was unnecessary because
using an obs_view just to display a single source feels somewhat
superfluous.
API changed:
--------------------------
void obs_output_set_audio_encoder(
obs_output_t *output,
obs_encoder_t *encoder);
obs_encoder_t *obs_output_get_audio_encoder(
const obs_output_t *output);
obs_encoder_t *obs_audio_encoder_create(
const char *id,
const char *name,
obs_data_t *settings);
Changed to:
--------------------------
/* 'idx' specifies the track index of the output */
void obs_output_set_audio_encoder(
obs_output_t *output,
obs_encoder_t *encoder,
size_t idx);
/* 'idx' specifies the track index of the output */
obs_encoder_t *obs_output_get_audio_encoder(
const obs_output_t *output,
size_t idx);
/* 'mixer_idx' specifies the mixer index to capture audio from */
obs_encoder_t *obs_audio_encoder_create(
const char *id,
const char *name,
obs_data_t *settings,
size_t mixer_idx);
Overview
--------------------------
This feature allows multiple audio mixers to be used at a time. This
capability was able to be added with surprisingly very little extra
overhead. Audio will not be mixed unless it's assigned to a specific
mixer, and mixers will not mix unless they have an active mix
connection.
Mostly this will be useful for being able to separate out specific audio
for recording versus streaming, but will also be useful for certain
streaming services that support multiple audio streams via RTMP.
I didn't want to use a variable amount of mixers due to the desire to
reduce heap allocations, so currently I set the limit to 4 simultaneous
mixers; this number can be increased later if needed, but honestly I
feel like it's just the right number to use.
Sources:
Sources can now specify which audio mixers their audio is mixed to; this
can be a single mixer or multiple mixers at a time. The
obs_source_set_audio_mixers function sets the audio mixer which an audio
source applies to. For example, 0xF would mean that the source applies
to all four mixers.
Audio Encoders:
Audio encoders now must specify which specific audio mixer they use when
they encode audio data.
Outputs:
Outputs that use encoders can now support multiple audio tracks at once
if they have the OBS_OUTPUT_MULTI_TRACK capability flag set. This is
mostly only useful for certain types of RTMP transmissions, though may
be useful for file formats that support multiple audio tracks as well
later on.
The temporary unoptimized code we were using before just completely
allocated a new copy of each frame every single time a new async frame
was output by the source plugin. This just creates a cache of frames as
needed for the current format/width/height to minimize the allocation
and deallocation. If new frames come in that are of a different
format/width/height, it'll just clear the cache. This is a fairly
important optimization.
all the async video related stuff usually started with async_*, and
there were two that didn't. So I just renamed them so they have the
same naming convention
If an async video source stops video for whatever reason, it would get
stuck on the last frame that was played. This was particularly awkward
when I wanted to give the user the ability to deactivate a source such
as a webcam because it would get stuck on the last frame.
I unintentionally made it use obs_source::sample_info instead of using
the actual target channel count, which is designated by the OBS output
sampler info. obs_source::sample_info is actually used to indicate the
currently set sampler information for incoming samples. So if a source
is outputting 5.1 channel 48khz audio, and OBS is running at stereo
44.1khz, then the obs_source::sample_info value would be set to
5.1/48khz, not the other way around. It indicates what the source
itself is running at, not what OBS is running at.
I suppose the variable needs a better name because even I used it
incorrectly despite actually having been the one who wrote it.
