Pulsesink was recently changed to defer uncorking until there is data
to write. This condition will however never occur when EOS in being
rendered (since that marks the end of data). Changing to PAUSED state
while EOS is being waited on results in a hang: pausing corks the
stream, which will never be undone since there is no more data when
going back to PLAYING. If pulsesink is the clock provider, deadlock
ensues since time doesn't continue in corked state and the clock id
for EOS wait never fires.
Fixes#645961.
Not doing so can result in a deadlock when two threads enter
gst_pulseringbuffer_open_device at the same time, as
pa_threaded_mainloop_wait releases the mainloop lock while waiting,
allowing another thread to take it, resulting in a deadlock as two
threads waits for the lock the other is holding.
https://bugzilla.gnome.org/show_bug.cgi?id=643087
By allowing larger chunks to be sent, PulseAudio will have a
lower CPU usage. This is especially important on low-end machines,
where PulseAudio can crash if packets are coming in at a higher
rate than PulseAudio can process them.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
After starting the ringbuffer, we wait for enough data to arrive before
uncorking the stream. This will cause the pipeline to stall if we get an
EOS (or otherwise need to flush the stream) before sufficient data
becomes available. This patch makes sure that the stream is uncorked
while flushing to avoid this problem.
Fixes issue with a webkit unit test testing reverse playback of
an MP4 H.264/AAC file.
https://bugzilla.gnome.org/show_bug.cgi?id=639740
This makes the call to pa_stream_cork() during ringbuffer pause()
synchronous, which makes sure that the clock does not advance after we
take a snapshot for start_time.
https://bugzilla.gnome.org/show_bug.cgi?id=639240
* ext/pulse/pulsesrc.c (gst_pulsesrc_class_init, gst_pulsesrc_init)
(gst_pulsesrc_set_property, gst_pulsesrc_get_property)
(gst_pulsesrc_open): Add a "client" property, as in pulsesink.
Fixes#634914
Don't uncork in the _start method just yet but wait until we have written some
samples to pulseaudio. This avoid underruns on pulseaudio and less crackling
noises when starting.
Make the is_dead check more clear and add an option to check for the status of
the stream in addition to the context.
We don't need a stream to get the device_description string.
Fixes#630317
We also need to share the main-loop threads as this owns the context. Thus have
a class wide main-loop thread. From this we create a context per client-name.
Instead of always looking up the context, we keep this with the instance. The
reverse mapping is only needed in pulse singal handlers. This saves a lot of
locking. Also one signal handler becomes simpler as ther eis only one mainloop
to notify.
Now valgind happy - no leaks, no bad reads/writes.
This reverts major parts of commit 69a397c32f.
Fixes#628996
Use g_slist_prepend as we don't care about the order. Check for list == NULL
instead of iterating the list to see if it is empty. Move ctx allocation down
to prevent leak in case of failure.
Don't leak the pulsesink element by having the clock keep a ref to the sink.
Create the clock only once in the constructor and use the baseaudiosink clock
cleanup code.
Allows the application to modify the client name used to connect when
connecting to the PulseAudio daemon. Note however that updating the
property after the element reached the READY state will have no
effect until the next NULL->READY transition.
Fixes bug #627174.
Avoid to create a new PA context for each new client by using a hash
table containing the list of ring-buffers and the shared PA context
for each client. Doing this will improve application memory usage in
the cases where multiple pipelines involving multiple pulsesink
elements are used.
Fixes bug #624338.
If the application requests a state-change and pulsesink fails to open
the ring_buffer device the mainloop attribute of the sink should be
cleaned up to avoid future state-change (NULL->READY) failures.
The existing get_type() implementation is racy, and the
g_type_class_ref() workaround didn't actually work because
it was in the wrong function. Since class creation in GObject
is thread-safe these days (since 2.16), the class_ref workaround
is no longer needed and it is sufficient to ensure the _get_type()
function is thread-safe, which G_TYPE_DEFINE does.
https://bugzilla.gnome.org/show_bug.cgi?id=624338
when we are shutting down, we might still receive state updates from pulseaudio
but since we are unparented we should not do anything with the NULL parent
anymore.
Use the acquired field of the ringbuffer in get_time to know when we are in an
invalid state. We don't clear the rate flag when releasing the ringbuffer so
this values is not usable.
Avoids some error messages being posted because the pulseaudio connection is
down.
Generally decisions on the volume of the stream should be done inside of
PA, not inside of Gst. Only PA knows how volumes translate between
devices and s on.
This patch makes sure that all volumes set via the volume property are
only applied *once* to the underlying stream. After applying them the
client side will not store them anymore. This should make sure that
really only user-triggered volume changes are forwarded to server, but
the client never tries to save/restore the volume internally.
Fixes bug #595231.
pthread does not guarantee that there are no spurious condition variable
wakeups, neither does pa_threaded_mainloop_xxx() which is a wrapper
around it. So we need to loop around the _wait() function to make sure
we get the right wakeup.
Also, unify the order of the wait loops across the file.
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.
