GNOME Bugzilla – Bug 781249
directsoundsrc: Correctly calculate segsize and segtotal
Last modified: 2017-05-12 09:21:57 UTC
See commit message. This was reverted again recently because it causes drop-outs and continuous underruns sometimes. Also it's not complete: the DirectSound buffer-size is always configured to 2s, regardless of the properties.
Created attachment 349766 [details] [review] directsoundsrc: Correctly calculate segsize and segtotal segsize should be based on latency-time, and must be a multiple of the frame size. segtotal should be based on buffer-time and segsize. This prevents errors caused by outputting buffers that are not a multiple of the frame size, and actually makes the buffer-time and latency-time properties do what they're supposed to do.
Created attachment 350998 [details] [review] Use latency-time and buffer-time settings I've squashed your commit into this one. It should fix this issue. Next on my list is https://bugzilla.gnome.org/show_bug.cgi?id=773681 which is due to similar problems in sleeping while polling.
Comment on attachment 350998 [details] [review] Use latency-time and buffer-time settings commit 1a8610f8f8305f20a235aa4fe78ab3fb0c785df4 Author: Nirbheek Chauhan <nirbheek@centricular.com> Date: Wed May 3 22:50:27 2017 +0530 directsoundsrc: Use latency-time and buffer-time settings Earlier, the plugin was ignoring those settings and blindly setting buffer-time to 2 seconds and latency-time to 200ms, which forced all pipelines to have a minimum latency of 200ms + sink latency. The values of segsize and segtotal were also not derived correctly. Now we obey these values, and you can get close to the previous behaviour by setting buffer-time and latency-time manually. Note that they are set in microseconds. As a consequence, when we haven't received enough data from the device, we now sleep for a time proportional to the data remaining. However, Directsound is a deprecated API so it maintains its own software ringbuffer which updates at arbitrary intervals. Hence we might have to wait a full segsize to get the last 10% of data. To avoid tight loops, we clamp our sleep floor at 10ms. In my testing, this keeps the wakeups not-too-high (proportional to the latency-time set on the source). Further improvements should be made by fixing the WASAPI audio source plugin instead of this. Directsound is deprecated and as the comments explain, it is impossible to get low latency, decent quality, or good performance from it. Based on a patch by Sebastian Dröge <sebastian@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=781249
Please also apply changes like in https://bugzilla.gnome.org/show_bug.cgi?id=773681#c31 (also see my review comments for some cleanup needed on top of that). After that I think we can close this bug.
Created attachment 351542 [details] [review] Use GstClockID to wait instead of Sleep() I've tested this on Windows 8.1 and it seems to work fine. Will push once I get a +1 from someone.
Comment on attachment 351542 [details] [review] Use GstClockID to wait instead of Sleep() Also push to 1.12 branch, and close this with target milestone 1.12.1 :)