Unknown preprocessor symbols are replaced with 0 in preprocessor
conditionals, so if `__has_attribute()` is not supported, then
the condition will be the following:
if 0 && 0(__cleanup__)
which is syntactically incorrect.
So split the condition and only check `__has_attribute()` if
it exists.
Fixes: 65d949558b ("spa: utils: add scope based resource cleanup")
Fixes#4962
Check if the pod size is at least big enough to hold 1 item when getting
array or choice items.
Check that the number of choice values is at least enough to handle the
given choice type. Remove some redundant checks.
Instead of always renegotiating a new format and buffers when the
driver changes, only renegotiate when the converter notifies us
of an EnumFormat change. It's possible that changing the driver does
not actually change the result of EnumFormat and then we would simply
renegotiate to the same format and buffers for nothing.
In the audioconvert, check if the target_rate of the new driver has
changed. We keep the last value in our own clock so that we don't
renegotiate when we are removed from a driver (which sets our own
clock as the position). The target rate influences the result of the
EnumFormat params when we are resampling and so we need to emit a
EnumFormat change.
Makes it possible to define audio channels and position with a
predefined layout string.
It is easier and less error prone to say "5.1" than to spell out
[ FL FR FC LFE RL RR ].
AUX channels have a special syntax. AUX<N> will make <N> AUX
channels. Easier to say AUX128 than to write an array with the
128 AUX channels.
spa_json_init assumes that we start in an object and always requires a
key/value pair. If the last part is a key, it returns and error and does
not want to return the key value.
This causes problems when parsing AUX0,AUX1,AUX2 or any relaxed array
withand odd number of elements.
Make a new spa_json_init_relax that takes the type of the container
we're assuming we're in and set the state of the parser to array when we
are parsing a relaxed array.
Fixes#4944
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
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).
