Adds an additional search path for UI-independent and
installation-independent plugins for windows/mac.
Windows:
%appdata%/obs-plugins/
Mac:
~/Library/Application Support/obs-plugins/
Plugin directory format is [module]/bin and [module]/data.
On windows, for 32bit binaries:
[module]/bin/32bit
and 64bit binaries:
[module]/bin/64bit
Before: After:
obs_service_gettype obs_service_get_type
It seems there was an API function that was missed when we were doing
our big API consistency update. Unsquishes obs_service_gettype to
obs_service_get_type.
I didn't think it would ever need to be exported, but this function is
actually useful for applying settings to properties (to call all of
their update callbacks based upon the settings) without necessarily
having to have an object associated with it.
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 comment says "these are different", but doesn't state why.
Actually, I should really rename the output flags so they're not flags,
but instead just "caps", because that's really all that they are.
obs_data_apply is used to apply the changes of a source object in to a
destination object. Problem with this however is that if sub-objects
are in use, it currently just copies the pointer of the sub-object,
meaning that the source and destination will both share the same
sub-object via reference. If anything modifies that sub-object data,
it'll modify it for both objects, which was not intended.
Instead of copying the object pointer, create a new copy and then
recursively repeat the process to ensure the data is always completely
separate.
Changed:
char *os_get_config_path(const char *name);
To:
int os_get_config_path(char *dst, size_t size, const char *name);
Also added:
char *os_get_config_path_ptr(const char *name);
I don't like this function returning an allocation by default.
Similarly to what was done with the wide character conversion functions,
this function now operates on an array argument, and if you really want
to just get a pointer for convenience, you use the *_ptr version of the
function that clearly indicates that it's returning an allocation.
The gs_enum_adapters function is an optional implementation to allow
enumeration of available graphics adapters that can be used with the
program. The ID associated with the adapter can be an index or a hash
depending on the implementation.
Direct3D textures are usually aligned to a specific pitch, so their
internal width is often not equal to the expected output width; this
means that if we want to use it on our texture output, that we must
de-align the texture while copying the texture data.
However, I unintentionally messed up the calculation at some point with
RGBA textures, so the variable size I was supposed to be using was
supposed to be multiplied by 4 (for RGBA), while I was still expecting
single channel data. So, if the texture width was something like 1332,
the source (directx) texture line size would be somewhere at or above
5328 (because it's RGBA), then destination is at 1332 (YUV luma plane),
and it would unintentionally treat 3996 (or 5328 - 1332) bytes as the
unused alignment data. So this fixes that miscalculation.
Refer to https://www.opengl.org/registry/doc/GLSLangSpec.4.10.6.clean.pdf
for a list of current (reserved) keywords.
In the future the shader compiler in libobs-opengl should probably take
care of avoiding those name conflicts (bonus points for transparently
remapping the names of effect parameters)
Instead of returning a valid string value when there are no more strings
available in the list, return NULL to indicate failure. An empty string
should really be allowed to be a valid value for the list.
The return value of os_sleepto_ns is true if it waited to the specified
time, and false if the current time is past the specified time. So it
basically returns true if it successfully waited.
I just didn't check the return value properly here, so it ended up just
setting the count of frames to 1 if overshot, ultimately causing sync
issues.
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.
This allows us to change the visible UI name of a property after it's
been created (particularly for a case where I want to change an
'Activate' button to 'Deactivate')
A slightly refactored version of R1CH's crash handler, allows crash
handling for windows which provides stack traces of all threads and a
list of all loaded modules. Also shows the processor, windows version,
and current libobs version.
When the encoder is set to scale to a different resolution than the obs
output resolution, make sure it uses the current video colorspace and
range by default.
I actually kind of hate how strstr returns a non-const even though it
takes a const parameter, but I can understand why they made it that way.
They really should have split it in to two functions though, one const
and one non-const or something. But alas, ultimately for a C programmer
who knows what they're doing it isn't a huge deal.
This adds support for the windows 8+ output duplicator feature which
allows the efficient capturing of a specific monitor connected to the
currently used device.
Previously, the design for the interaction between the encoder thread
and the graphics thread was that the encoder thread would signal to the
graphics thread when to start drawing each frame. The original idea
behind this was to prevent mutually cascading stalls of encoding or
graphics rendering (i.e., if rendering took too long, then encoding
would have to catch up, then rendering would have to catch up again, and
so on, cascading upon each other). The ultimate goal was to prevent
encoding from impacting graphics and vise versa.
However, eventually it was realized that there were some fundamental
flaws with this design.
1. Stray frame duplication. You could not guarantee that a frame would
render on time, so sometimes frames would unintentionally be lost if
there was any sort of minor hiccup or if the thread took too long to
be scheduled I'm guessing.
2. Frame timing in the rendering thread was less accurate. The only
place where frame timing was accurate was in the encoder thread, and
the graphics thread was at the whim of thread scheduling. On higher
end computers it was typically fine, but it was just generally not
guaranteed that a frame would be rendered when it was supposed to be
rendered.
So the solution (originally proposed by r1ch and paibox) is to instead
keep the encoding and graphics threads separate as usual, but instead of
the encoder thread controlling the graphics thread, the graphics thread
now controls the encoder thread. The encoder thread keeps a limited
cache of frames, then the graphics thread copies frames in to the cache
and increments a semaphore to schedule the encoder thread to encode that
data.
In the cache, each frame has an encode counter. If the frame cache is
full (e.g., the encoder taking too long to return frames), it will not
cache a new frame, but instead will just increment the counter on the
last frame in the cache to schedule that frame to encode again, ensuring
that frames are on time and reducing CPU usage by lowering video
complexity. If the graphics thread takes too long to render a frame,
then it will add that frame with the count value set to the total amount
of frames that were missed (actual legitimately duplicated frames).
Because the cache gives many frames of breathing room for the encoder to
encode frames, this design helps improve results especially when using
encoding presets that have higher complexity and CPU usage, minimizing
the risk of needlessly skipped or duplicated frames.
I also managed to sneak in what should be a bit of an optimization to
reduce copying of frame data, though how much of an optimization it
ultimately ends up being is debatable.
So to sum it up, this commit increases accuracy of frame timing,
completely removes stray frame duplication, gives better results for
higher complexity encoding presets, and potentially optimizes the frame
pipeline a tiny bit.
The boolean variables which stored whether frames have been
rendered/downloaded/converted/etc were not being reset when video
restarted, causing frames to not be sent in the correct order whenever
video was reset. This could lead to minor desync of video/audio.
In certain circumstances where the output was stopping, and where data
took a long enough time to send (such as when using an encoding preset
that causes high CPU usage), the output would sometimes still send data
even after it was stopped, typically causing the output to crash.
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.