Adds a source which tests whether audio buffering affects audio sync.
This sync test cycles through 7 audio/video frequencies at 250
millisecond intervals, and via a hotkey, will artificially move audio
time back by one second (and the audio cycle back by 4 frequencies).
This will artificially increase audio buffering by approximately 750
milliseconds.
Results from this test as of this writing: this test proves that dynamic
audio buffering does not affect sync.
This reverts commit 94b5982216.
Reverting this commit because it had some negative side effects, such as
adding 500 milliseconds to the startup time. NVENC detection should
really be done through its proper API, and not via creating an encoder
on startup.
Due to reports that the bandwidth test is randomly causing community
strikes on Youtube (likely due to bad automatic detection), the
bandwidth test will be disabled for Youtube until the Youtube API is
implemented.
Fixes an issue where text would not have language glyph fallback if
another language would used. This problem still needs a solution on
linux/mac (and preferably a fix for language glyph fallbacks when using
freetype 2 in general).
This commit fixes a bug that occurs on Windows 8+ when two or more
"Display Capture" sources are active that are configured to capture the
same monitor. Only one display capture would show, while all subsequent
display captures would display nothing.
Closesjp9000/obs-studio#1142
When a scene is added as a scene item with the same audio sources that
are already in the current scene, it would cause the current scene to no
longer output audio due to audio.
To replicate the issue, you would create two separate audio device
captures in scene 1, use add existing in scene 2 and add one of those
audio sources, then go back to scene 1, add scene 2 as a source, then
make scene 1 invisible.
There were cases where the channel format could be set to 7, which used
to be a valid format but now no longer is. If that format is set, just
use SPEAKERS_7POINT1 instead.
Makes it a bit more clear this option shouldn't be used unless you're on
SLI/crossfire.
In the future, something should be put in to the program that detects
laptops and warns on how to set up their adapter for efficient capture.
Closesjp9000/obs-studio#1138
This pull request changes the fallback sample format for pulse-audio
to from PA_SAMPLE_S16LE to PA_SAMPLE_FLOAT32LE.
The pulseaudio plugin can handle the following sample format:
* PA_SAMPlE_U8
* PA_SAMPLE_S16LE
* PA_SAMPLE_S32LE
* PA_SAMPLE_FLOAT32LE
When an audio device advertises itself as another format, the pulseaudio-plugin
will ask pulse audio to convert to the fallback sample format.
The fallback PA_SAMPLE_S16LE is not ideal when your audio interface advertises
as PA_SAMPLE_S24LE since the conversion will lose precision.
With PA_SAMPLE_FLOAT32LE there is no precision loss and it is also equals OBS's
internal format.
Some audio devices do not have a fixed number of channels. For example,
Soundflower. This was previously fixed by defaulting the speaker layout
to stereo. With surround sound support, the default has been changed to
the output speaker layout as set in Settings > Audio.
Closesjp9000/obs-studio#1110
The list of channel layouts available for decklink input is missing 2.1
& 4.1 layouts. The commit adds them. This aligns the decklink input
with the speaker layouts available at outputs. Having different layouts
as input and output invokes FFmpeg resampler, which remixes the channels
in non trivial way except when downmixing to stereo. This patch allows
to avoid such uncontrolled remix of channels with decklink input.
The core audio aac encoder has bitrates maps specific to speaker
layouts. Previously, the bitrate map maxed at 320 kbs and was the map
for stereo. The bitrate map is now tailored to the speaker layout. In
practice this unlocks higher bitrates. For instance up to 960 kbs for
7.1. Additionally the commit fixes a bug with 2.1 with channels not
ordered correctly.
