Based on patch from Barnabás Pőcze <pobrn@protonmail.com>
Instead of trying to keep track of the missing bytes ourselves, use the
simple tlength - avail - requested formula to request more bytes from
the client.
Fixes#1981
Make the alignment parameter optional when negotiating buffers.
Default to a 16 bytes alignment and adjust for the max cpu
alignment.
Remove the useless align buffer parameter in plugins, we always
set it to 16 anyway.
For example, pulseaudio.js[1] immediately sends a
GET_SERVER_INFO request after AUTH, and only later
issues a SET_CLIENT_NAME.
See #1966.
[1]: https://github.com/janakj/pulseaudio.js
By default require that a client is authenticated and
has a manager to be allowed to run a command.
Specially:
* AUTH requires nothing
* SET_CLIENT_NAME and STAT only require authentication
Previously, when a sample is "committed" from an upload stream,
its reference count is set to 1. This is problematic if,
when the sample is committed for a second time, there are
streams playing the sample as the reference count will go out
of sync.
The problem can be easily triggered, especially with longer samples:
pactl upload-sample a-long-sample.ogg
pactl play-sample a-long-sample
pactl play-sample a-long-sample
pactl upload-sample a-long-sample.ogg # while playing
When the first stream finishes playing, it will free the sample,
which can cause problems e.g. for the second stream playing the
sample at that very moment.
Fix that by decoupling the buffer from the sample and making
the buffer reference counted. And remove the reference counting
from the samples as it is no longer needed.
Futhermore, previously, the reference counts were ignored
when the removal of a sample was requested. That is fixed
as well.
The previous issue can be triggered easily as well:
pactl upload-sample a-long-sample.ogg
pactl play-sample a-long-sample
pactl remove-sample a-long-sample # while playing
Fixes#1953
Create a new event for modules ('destroy') which is emitted from
`module_free()`. It is used by the module loading logic, to handle
when a module is destroyed without properly loading first.
Store the modules whose load has been initiated by a particular
client in the `pending_modules` list of the client. When the
client disconnects, "detach" the client from the pending module
objects. This way the reference count need not be increased
for asynchronous module loads.
Furthermore, if the module can load synchronously, do not create
the pending module object at all.
Only call `spa_list_remove()` in `stream_free()` if the
stream is pending. `spa_list_remove()` does not reinitialize
the list node, therefore calling `spa_list_remove()` again
after the stream has been removed from the pending list
will corrupt the pending list of the client.
When we can't fill a complete block, report the amount of data that we
used in missing/played instead of the complete missing part.
Fixes audio breaking up when looping in mpv.
Fixes#1132
Always reevaluate the tlength or total buffered samples when the
quantum changes, even for the first sample because it is possible that
we completely miscalculated the length when we started, like when the
quantum is force high and the requested latency is low.
Also only increase the calculated tlength, for smaller sizes we don't
need to do anything, we can keep the latency as is it.
See #1930
When we are draining or underrunning, read whatever we have in the
ringbuffer instead of silence. This places the last samples before
the drain into the sink, padded with 0.
Fixes#1549
When we have a fix_* flag set, make an extra format description with the
wildcards. This makes it possible for the session manager to fall back
to something when selecting a target and format.
Also only advertize the valid pulseaudio formats for the wildcards.
Fixes#1912
Keep track of the current quantum and recalculate the tlength in the
same way that pulseaudio does.
Send a bufferattr changed message to a client when we change the
parameters.
This fixes the case where the quantum is increased and there needs to be
more buffering to keep the stream going.
Because we keep everything in a ringbuffer and provide exactly the
required amount of data, we can use 1/4 buffers.
Also increase the buffer size. We don't want to limit the buffer size
to the negotiated tlength because it can be increased later. Instead
scale it to the max quantum size (8192) with a max resample rate of 32.
This reverts commit 1b94b66924.
It causes problems with qemu.
Without this patch, paplay --latency-msec=1 /some.wav hangs when
forcing the quantum to 8192. A different fix will be needed.
When we increase the quantum past the tlength, we need to be able to
ask for more bytes than the tlength or else we will never exit this
underrun state. Look at the last required bytes and use that if it's
larger then tlength.
If we are underrun, don't update the read pointer and let the write
pointer catch up.
If we don't have enough data to fill a buffer, skip all the available
data and wait for the new request to arrive.
Fixes#1857
Make the pulseaudio layer set the PW_KEY_STREAM_CAPTURE_SINK property
when a monitor device is selected as a source to make it easier for the
session manager to find the right source.
Drop client connections when pipewire goes away. pipewire-pulse daemon can remain running and pulseaudio clients will be able to connect again once pipewire is up and running.
Reorganize the latency setup in one place, return a desired device
latency for use as quantum.
PulseAudio assigns half of the (tlength - minreq) latency to the sink
but we can't do that because our sinks have a max-quantum of latency.
Fix this by clamping our calculated sink latency to the quantum
PulseAudio subtracts the sink latency from the tlength in adjust latency
mode, so we need to do the same.
This makes PULSE_LATENCY_MSEC values bahave more like pulseaudio.
See #1769
Make things work then there is no default input device and the default
source is actually the monitor of the default sink.
Also implement lookups of monitor sources with the monitor id as the
name.
Fixes#1691
Use the core.info clock rate as the default sample rate as soon as the
manager exposes the core object. Otherwise the default sample rate is 0
until someone calls GET_SERVER_INFO, which as a side effect sets the
default sample rate.
Fixes issues with sinks not appearing right away.
Fixes#1588
