If we let the daemon decide freely by passing -1, we end up always getting 20ms.
We want to set this value because in some cases we want to select a higher
latency-time in order to save power.
Fixes#597601
In case that the pulse daemon runs the source device at a relatively low fixed
fragment size compared to the requested latency-time, configure the ring buffer
segsize to the largest integer multiple of the fragment size that is still
smaller than or equal to the requested latency-time.
Fixes bug #597463.
Remove the code to deal with a ringbuffer reset as this code is now in the base
class.
Bump the -base requirement as we need the new baseaudiosink code to function
properly.
Set the default slave method to the much better skew algorithm. This is the
default in the new base class but we override this here as well for the
upcomming release.
Otherwise that code will just be expanded to nothing when compiled
-DG_DISABLE_ASSERT (PS: why is mainloop_start() called in the init
function and not when changing state to READY?)
Keep track of the paused state of the source and leave the read function when
paused.
don't wait for a latency update when the delay is not yet known but simply
return 0 instead of blocking.
Keep track of the corked state of the stream.
Fix the state changes.
We can't wait for the ENTER/LEAVE messages to be be posted because the base
class sometimes calls the start method with the object lock, which would block
the message posting.
Instead, just assume that the message will be posted soon and continue. We'll
have to fix this in the base class.
Emit stream-status messages for the pulse thread.
Don't use our own GCond for signaling but simply use the pulse mainloop
mechanisms for synchronisation.
See #587695
Upper volume limmit was 1000. That appear unneceasrily high. It would also cause
sever distortion if accidentialy used. Now its 10 (~ +15db) which is also in
sync with volume and playbin2.
Since we map the ringbuffer to the pulseaudio internal ringbuffer, flush the
pulseaudio buffer when we are asked to clear the ringbuffer.
This avoids some leftover audio after a seek.
Hack around thread-safety issues in GObject and our racy _get_type()
functions (we could easily fix the _get_type() functions, but we still
need to hack around the GObject class races until we require a newer
GLib version, I think).
First we ignore request to fill the ringbuffer which are less then a segment.
The small request where causing stutter.
Then we disable flushing the stream when running against pa 0.9.12 as this
triggers an assertiong in the sound server and terminates it. It does not happen
with 0.9.10 and 0.9.14.
