Make easier to package A2DP codecs separately, by splitting each to a
separate SPA plugin. Adjust the code to not use a global variable for
the codec list.
The A2DP SPA interface API is in the bluez5 private headers, and not
exposed in installed SPA headers, as it's too close to the
implementation.
Instruct the policy to not configure audio adapter nodes in DSP mode. Instead,
Device nodes will always be configured in passthrough mode, and client nodes
will be configured in convert or passthrough mode depending on whether the
client format matches the device format or not.
The current _info_update() methods will always reset the change_mask in
the new info structure.
This causes problems if multiple updates are applied to the info before
the rescan in the session manager of pulse-server is excuted. The first
update is cleared and this causes the session manager to sometimes miss
the state changes of nodes and fail to suspend them.
Add a new method to merge with optional reset of the various
introspection info structures. We can use this instead and simply
accumulate all changes until the rescan code has processed all changes.
When we destroy a node that still has another driver, recalculate the
graph so that the driver has a chance to idle.
This can happen when we add an inactive node to the driver and then
destroy the node, like for jack clients.
Make sure we always suspend before reconfiguring a device.
Put the node and the device in passthrough mode when requested. Move
back to DSP mode after the node is unlinked.
Parse the exclusive flag of a stream once when the node info changes.
Use a new variable 'passthrough' to remember the current state of
a node and the peer.
Parse non-raw formats as well.
Check if two nodes can passthrough by intersecting the EnumFormat
params. If it is possible, configure the node for passthrough.
Don't try to reconnect nodes in passthrough.
Fail if we can't find a node compatible with passthrough.
See #629
Just like the latency, move the codecs to the device Route param.
This way, it is easier for the session manager to save and restore
the codecs as part of the Route settings.
If a program using pipewire-alsa calls snd_pcm_close() immediately after
snd_pcm_prepare() without reading or writing any data the client node
may be removed before the session manager can link it, which would
result in the following log warnings:
can't link 35:40 -> 43:48: link-factory: unknown input port 48
error id:25 seq:11467 res:-22 (Invalid argument): link-factory: unknown input port 48
can't link 35:41 -> 43:46: link-factory: unknown input port 46
error id:16 seq:11468 res:-22 (Invalid argument): link-factory: unknown input port 46
We start with setting the visitied flag on the driver. Still follow the
links to other visited nodes because it might make them active. for
example, adding a link between two midi ports should make the midi
driver running so that data flows between the ports.
See #1559
Actually set the right properties on the source and sink.
Not quite right because the pulseaudio ROC has a sink-input or a
playback stream in pipewire.
See #1538
Make the streams passive so that things can suspend.
Use a differend node.name for the input and output streams so that
autoconnect can actually remember the right target.
Make the media.name and description nicer.
See #1557
Audio with big frame sizes (especially audio with multiple channels) needs more
buffer size than the one calculated with the current formula. This patch uses
the frame size to calculate the buffer size, fixing playback issues for clients
configured in passthrough mode.
When the client adapter is configured in passthrough mode, the stream param
changed event in pipewire-pulse is emitted before the session manager creates
the link, and not after. Therfore, the peer can never be found when replying
create stream, and the pulseaudio application receives a stream error.
This patch delays the create stream reply until the link is added if the peer
cannot be found, fixing the above race conditon to allow passthrough mode to
work with pulseaudio applications.
When we inactivate the stream, clear the draining/drained state.
Otherwise, the stream stays in the drained state and won't call the
process function anymore when we activate it again.
This used to work before because we called the process function from the
Start command, which would queue a buffer and unset the drained flag.
Calling the process function from Start was however not right when the
process function needed to be called from the RT thread or when the
stream is a driver.
Fixes the issue where speaker-test would only play one channel.
Make the input buffer a little larger and leave the top blockSize
samples zero. That way we can fill up the lower part, leave the upper
part zero padded and feed this to the fft directly. Also only clear
the lower part when we can't fill it completely.
This removes some memcpy and memset operations.
Virtual devices tend to start with partial fields set in the EnumFormat
param (usually rate is missing). This causes virtual devices to be
invisible until they are used in some way.
Fix this by relaxing the parsing of EnumFormat and by falling back to
the server defaults for the unspecified fields.
Fixes#1413
When we send a Play request to the client but it deactivates before
sending the reply, make sure we send a Pause request as well so that
Play/Pause is always matched up.
Fixes#1548
When the allowed rates does not contain the default rate, also
fall back to only the default rate.
See the configured rates in the properties so it can be inspected.
Add option to set NULL data as the port data so that plugins can
skip processing.
Add 8 mixer ports and skip NULL data.
Move silence and discard samples to static area.
Improve the virtual sink examples. use the correct mixing for the HRIR
channels.
