From 96956cf4f2a761fd1de91a953c35147b9d13c62a Mon Sep 17 00:00:00 2001 From: Nirbheek Chauhan Date: Mon, 25 Nov 2019 21:00:14 +0530 Subject: [PATCH] wasapisrc: Fix glitching and clock skew issues 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 --- sys/wasapi/gstwasapiutil.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sys/wasapi/gstwasapiutil.c b/sys/wasapi/gstwasapiutil.c index 81df8d838..1651bdf80 100644 --- a/sys/wasapi/gstwasapiutil.c +++ b/sys/wasapi/gstwasapiutil.c @@ -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; -- 2.34.1