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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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6d27c0ac08
We were miscalculating the device period, i.e. the number of frames we'll get from WASAPI in each IAudioClient::GetBuffer call, due to a calculation mistake (truncate instead of round). For example, on my machine when the aux input is set to 44.1KHz, the reported device period is 101587, which comes out to 447.998 frames per ::GetBuffer call. In reality we will, of course, get 448 frames per call, but we were truncating, so we expected 447 and were discarding one frame every time. This led to glitching, and skew over time. Interestingly, I can only see this with 44.1Khz. 48Khz/96Khz are fine, because the device period is a more 'even' number. Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/806 |
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| androidmedia | ||
| applemedia | ||
| bluez | ||
| d3d11 | ||
| d3dvideosink | ||
| decklink | ||
| directsound | ||
| dshowdecwrapper | ||
| dshowsrcwrapper | ||
| dshowvideosink | ||
| dvb | ||
| fbdev | ||
| ipcpipeline | ||
| kms | ||
| msdk | ||
| nvcodec | ||
| opensles | ||
| shm | ||
| tinyalsa | ||
| uvch264 | ||
| wasapi | ||
| winks | ||
| winscreencap | ||
| meson.build | ||