Make the converter format a higher priority in all cases. The converter
has been negotiated first and is able to make a better suggestion for
the ideal format in all cases.
Make a function to create a filter. This is a pod that has all valid
defaults fixated and the invalid ones left unfixated.
Use this filter is a first attempt to negotiate a link format. The
effect is that a format will be chosen first that matches all the valid
defaults as much as possible instead of negotiating to the first thing
that matches.
Suppose we have a higher priority port with the format:
foo/bar
key: { default:1024, min:1, max:2048 }
And another port with two params:
foo/bar
key: 512
rate: 2/1
foo/bar
key: 1024
rate: 30/1
By first trying key: 1024 we negotiate to the more specific second property
with the higher rate.
Set different icons for A2DP & HFP output routes, so that they look
different (in Gnome).
Don't call the non-HFP output route as "headset" or "handsfree" in this
case, to be less ambiguous about microphone availability.
Also set device.icon-name for the device too.
While ASHA does not really use the D-Bus device set interface
but since ASHA has a device-wide HiSyncId and needs to handle
left and right side via a combine sink, use the device set
notion for ASHA as well.
Let's just directly use the next timer value from the other side
directly, and the reference time as well to calculate the expected next
sample position we want to send from on this side.
For ASHA stereo, the timer for flushing packets on both sides of a
pair should be as closed to each other as possible.
Also set seqnum before first flush so other side sends the correct
packets at the start.
In ASHA, we might need to sync sequence numbers between
left and right side media sink. If the sequence number
is added in start encode, it becomes difficult to set
the sequence number again when a call to reset_buffer
in media sink might have happened already.
This effectively reverts commit ab5f81b9a.
ASHA devices with the same HiSyncId are a pair and this
will be used later on to set them up together with a
combine sink like device set for BAP. Side information
is required for channel information when setting up the
combine sink.
The +BCS event may interrupt any of the initialization commands after
SLC is established and by changing the state here we may lose track
of the initialization sequence.
There is no reason to have the hfp_hf_bcs state anyway. If the
initialization sequence is over, we can remain in the hfp_hf_vgm state.
Some devices have a hardware volume control, but not a dedicated
hardware mute control. In some of these cases, the volume control is
described as having a hardware mute when volume is 0. This is described
in the TLV information of the volume control, when the
SNDRV_CTL_TLVD_DB_SCALE_MUTE flag is set in the TLV structure.
If set, alsa-lib will set the minimum dB value to -99999.99dB, which
can be detected inside PipeWire.
PipeWire can then use this hardware volume control to apply hardware
mute, when set.
In order to be able to set volumes and mutes separately, changing the
volume whilst muted will save the value, but not write it to the
hardware. When the device is unmuted, the saved value will be restored.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Print the error in td->err (the SO_ERROR) or else let the user know
this is a regular hangup. The previous printout was always reporting
EAGAIN (remainder in errno from the event loop code, I suppose)
as an error, without any real data.
Reset the ERR counters when pressing the c key. This only makes the
current pw-top start counting from 0 again, it does not change anything
globally on the server. It's still usefull because you can use it to
let it count the number of new errors since the keypress.
Do some of the counting of the nodes and controls when we add a node
to make things easier later.
Do the setup of the graph controls after loading the graph because we
know exactly how many controls we will have.
Fixes controls being unavailable until the filter-graph is activated.
Destroy the sources from the io handler immediately when there is an
error so that we don't end up in endless error wakeups.
Schedule the free from the main loop and make sure only one can ever
run.
The manager is actually not supposed to decide much about the number of
audio and midi ports. It should just suggest a default when connecting
driver doesn't know.
Add a audio.ports parameters to manager and driver to suggest/ask for
the amount of audio ports. Let the audio.position/audio.channels be a
specification of the channel mask in case it matches the requested
channels, otherwise use AUX channels for the ports.
This means that we must derive the mode (sink/source/audio/midi) from
the ports that are negotiated in the manager and the driver, so delay
this until after negotiation.
Make sure all the possible modes work. For midi only streams, we can't
wait for the session manager to perform a PortConfig so do that
ourselves. Make sure we only use a source trigger when we have a sink.
Fixes#4666
Keep a position info for the stream it was set and then use the position
info for the stream that is driving the graph. Otherwise we might use a
destroyed position info.
