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3230 commits

Author SHA1 Message Date
Frédéric Danis
e0c79959fc bluez5: Add stereo support to LE Audio LC3 codec 2022-09-15 11:17:20 +00:00
Frédéric Danis
fd1b331353 bluez5: Fix sink timeout for BAP 
Data should be written to the ISO Stream fd every 10ms or 7.5ms depending
on the configuration selected.
 2022-09-15 11:17:20 +00:00
Frédéric Danis
071730ecab bluez5: Use delay from QoS for BAP 2022-09-15 11:17:20 +00:00
Frédéric Danis
39ef812ed5 bluez5: Complete BAP audio location mapping 2022-09-15 11:17:20 +00:00
Frédéric Danis
f428b13d06 bluez5: Set BAP QoS depending on selected configuration 
This move the QoS setup to be codec specific and fills it depending on the
selected codec configuration.
 2022-09-15 11:17:20 +00:00
Frédéric Danis
284da66deb bluez5: Add BAP_SINK/SOURCE from PACS UUID to adapter profiles 
BAP sink/source are characteristics of the PACS profile, but we're more
interested by those characteristics then the more generic profile.
 2022-09-15 11:17:20 +00:00
Frédéric Danis
496dda1bcc bluez5: Check if BlueZ daemon support LE Audio 
Legacy BlueZ 5 API doesn't support to register BAP codecs, which prevents
to register A2DP enhanced codecs.
 2022-09-15 11:17:20 +00:00
Frédéric Danis
b7ad1d1869 bluez5: Add SelectProperties DBus method support 2022-09-15 11:17:20 +00:00
Frédéric Danis
96acc5a79a bluez5: Add LE Audio BAP support to bluez5-device 2022-09-15 11:17:20 +00:00
Frédéric Danis
201fd7755d bluez5: Add LC3 codec support to LE Audio BAP 
By default this codec is disabled as the BlueZ support for LE Audio is
still experimental.
 2022-09-15 11:17:20 +00:00
Frédéric Danis
fd0bcb1699 bluez5: Add LE Audio BAP support to media-codecs 2022-09-15 11:17:20 +00:00
Frédéric Danis
00d51c3d31 bluez5: Rename codec API from *a2dp* to *media* 
The BlueZ Media1 interface will not only be used for A2DP but also for
LE Audio and code related can be shared.
 2022-09-15 11:17:20 +00:00
Wim Taymans
565bb2c493 resample: fix peaks sse code 
Fix the shuffle arguments.
Use another way of making 0x80000000 and use andnot to remove the
sign bit.
 2022-09-14 16:58:41 +02:00
Dmitry Sharshakov
3b89e6f369 alsa-seq: add an option to disable longname in MIDI ports 
Untested yet

Signed-off-by: Dmitry Sharshakov <d3dx12.xx@gmail.com>
 2022-09-13 07:46:40 +00:00
Dmitry Sharshakov
ea646c2d98 alsa-seq: set card long name or number in port names 
Signed-off-by: Dmitry Sharshakov <d3dx12.xx@gmail.com>
 2022-09-13 07:46:40 +00:00
Wim Taymans
5e890925a0 module-echo-cancel: don't load newer modules 
Add some version comments for the new AEC methods
 2022-09-13 09:44:36 +02:00
Jonas Holmberg
70471989e5 module-echo-cancel: Add activate/deactivate methods 
Add methods activate() that is called before first call to run() when
stream starts and deactivate() that is called after last call to run()
when stream stops. This makes it possible for aec-plugins to reset their
state between streams.
 2022-09-12 18:24:07 +00:00
Wim Taymans
634d7107a4 v4l2: use the device name to open in BeginParams 2022-09-12 09:51:55 +02:00
Wim Taymans
a7c28d64ae resample-peaks: small improvement 
Makes it easier for the compiler to make a constant.
 2022-09-08 15:40:01 +02:00
Wim Taymans
46c8c9ec03 audioconvert: set monitor port buffer size correctly 
Also use quantum_limit samples on the monitor ports.

Fixes #2677
 2022-09-08 12:25:28 +02:00
Wim Taymans
903f831b2d alsa: scale min value back to nominal rate 2022-09-07 23:39:51 +02:00
Wim Taymans
7057cca05e alsa: adjust min and max rates 
Use the DSD scaler and interleave factors to calculate the min and
max rates.
 2022-09-07 21:58:32 +02:00
Wim Taymans
955815b468 alsa: only scale DSD samplerates 2022-09-07 21:42:45 +02:00
Wim Taymans
68581235ae alsa: fix min dsd rate 
We just need to allow from DSD64 onwards. Remove some unused fields.
 2022-09-07 20:23:53 +02:00
Wim Taymans
72b6788c68 alsa: don't set impossible rates 
When the max rate is smaller than the min rate, don't add the DSD
format at all because it is not possible to play it.

See #93
 2022-09-07 19:58:30 +02:00
Wim Taymans
23522651f9 alsa: dsd rates are expressed in byte rate 
DSD64 would be a rate of 44100 * 64 / 8. When packed in U32_BE, we would
negotiate 44100 * 8 / 4 (88200) with the device, this means all rates
from 88200 and up are allowed for DSD64 in U32_BE.
 2022-09-07 19:54:09 +02:00
Wim Taymans
c9ee142b78 resample-peaks: unroll loop a little 2022-09-07 16:18:58 +02:00
Wim Taymans
187006f30e audioconvert: remove some double defines 2022-09-07 16:08:21 +02:00
Wim Taymans
6e9e02b420 audioconvert: refactor peaks resampler 
Use common code in macro and generate arch specific version.
Compile with -Ofast to optimize some fmaxf calls.
 2022-09-07 16:00:31 +02:00
Wim Taymans
968bc382af log: initialize pos to help the compiler 
Although pos will be set to 0 in the first iteration, the compiler does
not seem to figure this out, so help it a little:

../spa/include/spa/support/log.h:306:21: warning: ‘pos’ may be used uninitialized [-Wmaybe-uninitialized]
  306 |                 pos += sprintf(str + pos, "%02x ", buf[i]);                             \
      |                     ^~
../spa/include/spa/support/log.h:301:13: note: ‘pos’ was declared here
  301 |         int pos;                                                                        \
      |             ^~~
 2022-09-07 09:17:19 +02:00
Frédéric Danis
2c72597271 bluez5: Use spa_log_hexdump() to dump media properties/configuration 2022-09-07 07:50:23 +02:00
Frédéric Danis
1d39ea004e log: Add spa_log_hexdump 2022-09-07 07:50:23 +02:00
Wim Taymans
5b2b93f915 alsa: add period-size, period-num and headroom in props 
Add the currently configured alsa hw_params in the properties for
easier access.
 2022-09-06 20:16:29 +02:00
Wim Taymans
ffc09ce55e videoconvert: refresh with latest audioadapter copy 2022-09-06 17:48:00 +02:00
Wim Taymans
201e6ae9fd audioconvert: use given channelmap for volume 
Use the given channelmap for the volume, like it used to be in old
audioconvert.

This makes new streams expose a volume even when not negotiated yet.
 2022-09-05 15:29:16 +02:00
Wim Taymans
71ec8650ba audioconvert: remove redundant set_volume calls 2022-09-05 13:23:18 +02:00
Wim Taymans
0c47ab76a7 channelmix: Only filter FC/LFE when present 2022-09-05 13:22:40 +02:00
Wim Taymans
7b432aec88 spa: make default samplerate 48000 
Remove some 44100 default rates and use 48000 instead.
 2022-09-05 12:21:20 +02:00
Wim Taymans
cad7db9f58 libcamera: close camera in error path 2022-09-05 09:25:42 +02:00
Pauli Virtanen
df858cae7c bluez5: clarify OPUS-A2DP spec 
Also remove DRAFT tag.
 2022-09-03 19:08:27 +00:00
Wim Taymans
9123710971 spa: clean up some port io checks 
We should not generate a warning.
 2022-09-01 15:39:34 +02:00
Wim Taymans
2fa1b4384b spa: don't warn for NULL io 
The io is set to NULL when the port becomes unnegotiated.
 2022-09-01 15:31:14 +02:00
Wim Taymans
2054dcf36f alsa: Improve format negotiation 
When the device is not running but has a format, close/open the
device to get all the available formats again. Do the same when
setting a format.

Otherwise, the configuration space of the device is restricted to the
current negotiated format and we can't query the other possibilities
or change it.

Fixes #2625
 2022-08-31 10:03:39 +02:00
Wim Taymans
b33ffcf683 test: print channel map as reported by alsa as well 
See #1707
 2022-08-30 16:45:35 +02:00
Wim Taymans
24f6225c5d audioconvert: don't emit changed events for rate changes 
Rate changes can happen very often when a stream is doing rate control,
so don't emit the changes every time.
 2022-08-30 16:00:00 +02:00
Thomas Weißschuh
2394413ec3 hook: don't remove from unitialized list 
A lot of code calls spa_hook_remove() from error paths where the hook
and therefore the list may not have been initialized.
This leads to null-derefences.
 2022-08-30 11:52:42 +00:00
Thomas Weißschuh
b23b0e6125 list: add spa_list_is_initialized() 
This function can be used to test if a list has been initialized.
 2022-08-30 11:52:42 +00:00
Wim Taymans
e04e3ef40e audioconvert: fix rate match for sources 
Only update the resampler rate when we ask for more data, when we have
more input data, use the previously configured rate to calculate how
many samples we will consume.

Fixes resync errors with multiple sources. One source would do rate
matching, audioconvert would ask it to produce X samples, the source is
scheduled to produce the samples, the rate match is updated with the new
rate correction, audioconvert is scheduled again. It should now use the
X samples it asked to produce and apply the new rate correction for the
next iteration.
 2022-08-30 12:43:14 +02:00
Pauli Virtanen
c0e8b397f6 bluez5: avoid (harmless) integer overflow 
If no packets have been received and spa_bt_decode_buffer_process is
called, this->packet_size.max == INT32_MIN, which can give overflows.

Guard against this condition, although it should be harmless.
 2022-08-24 16:59:21 +00:00
Thomas Weißschuh
6531bedcdb spa: add SPA_WARN_UNUSED_RESULT 2022-08-19 07:35:02 +00:00