The behavior before b8eeb2db45 was that spa_audio_info_raw_update()
always sets audio.channels when audio.position is updated. The new
behavior does not set audio.channels when parsing audio.position.
This breaks things e.g. when combine-sink and loopback nodes are created
with only audio.position specified.
Restore the previous behavior.
Remove the SPA_AUDIO_MAX_POSITION define and use the
SPA_AUDIO_MAX_CHANNELS again.
Make a compile time define to override the default max channels of 64.
Make sure we compile the SPA library with the default 64 channels. If
you use the SPA library on a spa_audio_info you will get 64 channel
support, like before. If you want more channels, you will need to make
a padded structure or redefine the MAX_CHANNELS before you use the
spa_audio_info structures. You can use the padded structure with the
new functions that take the structure size.
With the extra checks in the parsing code, we avoid making a
valid spa_audio_info with too many channels that don't fit in the
structure. This means that code that receives a spa_audio_info can
assume there is enough padding for all the channels.
Add functions that take the size of the spa_audio_info struct in various
functions. We can use this to determine how many channels and channel
positions we can store.
Error out if we try to use more channels than we can fit positions. This
is probably the safest thing to do because most code will blindly try to
get the positions without checking the channel count.
Make sure we also propagate errors to the callers.
Parse the audio.position spec completely so that we have the right
number of channels but only store the first max_position channels.
Also rename some field to make it clear that this is about the max
number of channel positions.
Add a new SPA_AUDIO_MAX_POSITION constant with the maximum number of
channel positions that can be kept in the various audio_info structures.
Repurpose the SPA_AUDIO_MAX_CHANNELS as a suggestion for applications
for the max allowed number of channels in the system. Make it possible
to make this a compile time constant.
Add a function that accepts the size of the position array when reading
the audio positions. This makes it possible to decouple the position
array size from SPA_AUDIO_MAX_CHANNELS.
Also use SPA_N_ELEMENTS to pass the number of array elements to
functions instead of a fixed constant. This makes it easier to change
the array size later to a different constant without having to patch up
all the places where the size is used.
Define some rules for how the position information works for channels >
SPA_AUDIO_MAX_CHANNELS. We basically wrap around and incrementing the
AUX channel counters. Make a function to implement this.
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.
