gstreamer/sys/directsound
Nirbheek Chauhan 1a8610f8f8 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
2017-05-08 16:13:20 +02:00
..
gstdirectsoundplugin.c Fix FSF address 2012-11-04 00:09:59 +00:00
gstdirectsoundsrc.c directsoundsrc: Use latency-time and buffer-time settings 2017-05-08 16:13:20 +02:00
gstdirectsoundsrc.h directsoundsrc: Use latency-time and buffer-time settings 2017-05-08 16:13:20 +02:00
Makefile.am directsoundsrc: add missing -lole32 2016-04-03 13:37:16 +03:00
meson.build meson: directsound: Add ole32 library dependency 2016-10-21 06:05:27 -03:00