Disabling dB volumes for max_dB < 0 was added in Pulseaudio in 2021,
based on a device which had -128..-127.07 range. However, negative
max_dB is valid value for USB devices, and there are devices that have
it.
Eg. Microsoft LifeChat LX-3000 has
numid=6,iface=MIXER,name='Speaker Playback Volume'
; type=INTEGER,access=rw---R--,values=2,min=0,max=151,step=0
: values=150,150
| dBminmax-min=-28.37dB,max=-0.06dB
and the dB range seems to be OK. Web search for "The decibel volume
range for element" also gives other hits with seemingly OK looking
ranges.
Don't disable dB volume unless both the max is negative and the range is
suspiciously small. This should still disable it for the device this
check was originally added for.
Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/merge_requests/447
Link: 10ac01a206
Instead of using timerfd, use the context timer-queue to schedule
timeouts. This saves fds and removes some redundant code.
Make the rtp-source timeout and standby code a bit better by using
atomic operations.
Move the code to recalculate the hash and version into make_sdp. Add a
boolean argument to the make_sdp function. Recalculate the hash and
version when we are making a new SDP and leave the old values if we
are making an SDP to compare against the current one.
Add support for rlimit.<resource> = <value> in the context.properties to
adjust the rlimits of the process. A value of -1 sets the max limit.
This can be used to increase the number of file descriptors in a
pipewire process when select() is not used, for example. Other resource
limits might be interesting as well maybe.
Fixes#4047
We need exclusive port use if we negotiated SyncTimeline because there
can only be one consumer of the syncobj.
We also need to enable reliable transport if synctimeline is supported
but the release flag isn't.
Add some more logging to the port when the exclusive and reliable states
changed.
Fixes#4885
Add a PW_KEY_NODE_RELIABLE and PW_KEY_PORT_RELIABLE property. the port
property value is inherited from the parent when not explicitly set.
Setting the property on a port will activate a more reliable tee, that
actually only recycles buffers that were consumed. It will also activate
a mode in stream that gives out new buffers only when the previous one
was recycled and nothing else is queued. This is necessary to avoid
queuing in the stream when the other side is not consuming.
When a link is async but the output node is a driver of the input, we
can avoid async io. This also removes a potential out-of-order buffer
recycling when the node resumes at a different cycle.
See #4885
Some ALSA devices have minimum HW volume value that is muted. ALSA
indicates it with SND_CTL_TLV_DB_GAIN_MUTE = -9999999 dB/100 volume dB.
When rounding down to HW volume, we may get this muted value.
When determining splitting of volumes to mixers and soft volume, we
don't want HW mixers to set volume to muted, unless the target volume is
actually muted.
Fix by adding element_ask_unmuted_dB_vol() that rounds up if the asked
rounding mode resulted to mute.
This fixes mic getting muted at low volume despite ALSA reporting the dB
values correctly.
Fixes#4890
Move accounting for pending ISO packet to the reference time. Make sure
rate matching is reset on start, and reset matching on resync properly.
Allow resync on first cycle, ok since iso_io->now is valid immediately.
Silence padding larger than ISO packet may be needed for resync when
quantum is large. We can't insert silence by adding data to encoding
buffer, as the encoding buffer may be then too small and it may also be
partially filled.
Fix by inserting silence from flush_data() just before buffers would be
consumed.
Fixes ISO stream alignment at playback start.
If BlueZ doesn't reply, it may consider the operation still active.
Try to Release the transport to get to a known state.
This can happen if device doesn't respond to operations in reasonable
time and BlueZ doesn't have its own timeout which is the case for BAP
currently (which is a bug there).
Some tests - for example test-fmt-ops - are compute heavy. Since tests
in non-x86 builds are run inside qemu, they can be significantly slower,
exceeding the default 30 second timeout.
So set the timeout multiplier to 2 to allow for slower execution.
The interface of string typed controls has recently been changed in
libcamera[0], which affects `properties::Model`, so adapt to that change
in such a way that is compatible with both the new and old versions.
[0]: f84522d7cd
Add an example producer and consumer using the SyncTimeline metadata.
The syncobj are just eventfd for the purpose of the example.
Also demonstrate the RELEASE feature when negotiated.
Add some switches to tweak the SyncTimeline and features support.
See #4885
Avoid shadowing some variables from the parent block.
The node of a target can be NULL when the target is running in another
instance. We already do some checks for this but make sure we never
deref the NULL pointer.
Fixes#4922
Add an example of a filter sink that requires MemFd memory on the
input port. Show that it gets automatically mapped and that it contains
MemFd memory.
Fixes#4918
Don't just blindly mmap the buffer but only when the data pointer
is NULL. If it was mapped already by the peer or the adapter or the
buffer allocation, we don't want to mmap it again and override the buffer
data pointer.
Also mmap with the permissions on the data. There is not much point in
limiting the permissions for an input port (to read only). We could do
this but then we would not be allowed to modify the existing data
pointer. The problem is that when the stream mmaps the data as READ only
and set the data pointer, if it is then handed to the mixer, it would
assume it is mapped with the permissions and then segfault when it
tries to write to the memory. It's just better to only mmap when the
data is NULL.
We don't need to do this ourselves, the MAP_BUFFERS port flag already
makes sure this is done for use.
We used to have to do this here to ensure the mixer could find the data
pointer and not error out. Now that the mixer can MMAP, this can go.
See #4918
There are really 2 options for the buffer allocation:
1. allocate the buffers skeleton and meta/chunk/data in malloc memory.
This is when the PW_BUFFERS_FLAG_SHARED is unset.
2. allocate buffers skeleton in alloc memory and the meta/chunk/data
in shared memory when the PW_BUFFERS_FLAG_SHARED is set.
Optionally the data can be left unallocated in both cases when the
PW_BUFFERS_FLAG_NO_MEM is set. In this case we also need to pass the
SPA_BUFFER_ALLOC_FLAG_NO_DATA flag to allocator or else it will set the
data pointers to 0 sized memory in the skeleton.
If we use SHARED and we allocated memory, we can also set the MemFd and
mapoffset into our shared mem. We can do this even if the data_type is
MemPtr.
We can decide on the datatype to use earlier, based on the negotiated
flags. In the MemFd case, make sure the buffer data is page aligned in
that case to make things easier. Also force everything in SHARED mem
when the data is in SHARED mem. We also don't need to
PW_BUFFERS_FLAG_SHARED_MEM because we work with the negotiated flags
now to decide if SHARED mem is needed or not.
With this change, a node port could provide a MemFd data_type mask in
the Buffers param and this would negotiate shared mem with the mixer.
Previously, it would only ever allocate malloc memory.
See #4918
When we have a mixer node and we need to negotiate buffers between the
mixer and the node, take the CAN_ALLOC flag into account.
This is for input ports, which can have a mixer. If you make a filter
with a CAN_ALLOC input port, it will now not already contain buffer
data.
See #4918
A fixed channel count makes no sense with an entity based 3D audio format
like MPEG-H, because MPEG-H decoders do not simply decode; they
"spatialize" the entities, meaning that said entities are decoded and
rendered accordingto the needs of the target playback system and its
channel count.
