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.
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
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.
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?)
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.
We can use prebuf = 0 to instruct pulse to not pause the stream on underflows.
This way we can remove the underflow callback. We however have to manually
uncork the stream now when we have no available space in the buffer or when we
are writing too far away from the current read_index.
when we switch streams, the clock will reset to 0. Make sure that the provided
clock doesn't get stuck when this happens by keeping an initial offset. We also
need to make sure that we subtract this offset in samples when writing to the
ringbuffer.
If the caps changes, the sink is reset without transitioning through
a PAUSED->PLAYING state change, resulting in a corked stream. This avoids
the problem by checking that the stream is uncorked when writing samples
to it.
When trying to write out a segment, wait until there is enough free space
for the entire segment. This helps to reduce ripple in the clock reporting,
where the app might query the playback position while only half a segment
has been written (and is therefore reported by _delay(), even though
the ring buffer has not yet been advanced)
rather than PA thread.
pa_threaded_mainloop_lock() (a.o.) and by extension get_property should
not be done from a PA thread, but the latter may occur as a result of a
property change notification. Fixes#571204 (though current situation
not ideal, e.g. post message rather than signal).
newer pulseaudio.
Fixes: #567794
* Hook pulsesink's volume property up with the stream volume -- not the
sink volume in PA.
* Read the device description directly from the sink instead of going
via the mixer.
* Properly implement _reset() methods for both sink and source to avoid
deadlocks when shutting down a pipeline.
* Replace all simple pa_threaded_mainloop_wait() by proper loops to
guarantee that we wait for the right event in case multiple events are
fired. While this is not strictly necessary in many cases it
certainly is more correct and makes me sleep better at night.
* Replace CHECK_DEAD_GOTO macros with proper functions
* Extend the number of supported channels to 32 since that is the actual
limit in PA.
* Get rid of _dispose() methods since we don't need them.
* Increase the volume property upper limit of the sink to 1000.
* Reset function pointers after we disconnect a stream/context. Better
fix for bug 556986.
* Reset the state of the element properly if open/prepare fails
* Cork the PA stream when the pipeline is paused. This allows the PA
* daemon to
close audio device on pause and thus save a bit of power.
* Set PA stream properties based on GST tags such as GST_TAG_TITLE,
GST_TAG_ARTIST, and so on.
Signed-off-by: Lennart Poettering <lennart@poettering.net>
