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wasapisrc: Fix glitching and clock skew issues

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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
This commit is contained in:
Nirbheek Chauhan 2019-11-25 21:00:14 +05:30 committed by GStreamer Merge Bot
parent d8a51c6097
commit 6d27c0ac08
1 changed files with 3 additions and 1 deletions
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4
sys/wasapi/gstwasapiutil.c
View file

@ -919,7 +919,9 @@ gst_wasapi_util_initialize_audioclient (GstElement * self,
*ret_devicep_frames = n_frames;
} else {
*ret_devicep_frames = (rate * device_period * 100) / GST_SECOND;
/* device_period can be a non-power-of-10 value so round while converting */
*ret_devicep_frames =
gst_util_uint64_scale_round (device_period, rate * 100, GST_SECOND);
}
return TRUE;