snd_soc_dapm_new_dai() setups local variable "template" at (X) and (Y),
which is used at (Z). But these are difficult to read.
static struct snd_soc_dapm_widget * snd_soc_dapm_new_dai()
{
...
^ template.reg = ...
| template.id = ...
(X) template.name = ...
| template.event = ...
| template.event_flags = ...
v template.kcontrol_news = ...
if (rtd->dai_link->num_params > 1) {
...
^ template.num_kcontrols = ...
(Y) template.kcontrol_news = ...
v ...
}
...
(Z) w = snd_soc_dapm_new_control_unlocked(..., &template);
}
And this function has error message, but not for all cases.
This patch (1) setups "template" in one place, and indicate error message
for all cases. This patch cleanup the code, but nothing changed for
meaning.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/871qr4tzro.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
dapm_widget_set_power() (= X) is called only from
dapm_power_one_widget() (= Y), and total purpose of these functions are
calling dapm_seq_insert() (= a) accordingly for each widget.
(X) static void dapm_widget_set_power(...)
{
...
if (power)
(a) dapm_seq_insert(w, up_list, true);
else
(a) dapm_seq_insert(w, down_list, false);
}
(Y) static void dapm_power_one_widget(...)
{
..
switch (w->id) {
case snd_soc_dapm_pre:
(a) dapm_seq_insert(w, down_list, false);
break;
case snd_soc_dapm_post:
(a) dapm_seq_insert(w, up_list, true);
break;
default:
power = dapm_widget_power_check(w);
(X) dapm_widget_set_power(w, power, up_list, down_list);
break;
}
}
It should be more simple, but the code is unnecessarily complicated,
and difficult to read/understand. This patch merge these into one.
Link: https://lore.kernel.org/all/87tu42owdd.wl-kuninori.morimoto.gx@renesas.com/
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/8735bktzrx.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Current error handling on snd_soc_dapm_add_route() has some wastes.
It indicates *own* error message *only* for sink or source,
and return error directly at (A). OTOH, it has similar error message at
(B) which indicates *both* sink/source.
And more, (A) is using dev_err(), (B) is using dev_warn().
(B) is caring prefix, but (A) is not.
(X) int snd_soc_dapm_add_route(...)
{
...
if (wsource == NULL) {
(A) dev_err(...);
return -ENODEV;
}
if (wsink == NULL) {
(A) dev_err(...);
return -ENODEV;
}
...
ret = snd_soc_dapm_add_path(...);
if (ret)
(B) goto err;
return 0;
err:
(B) dev_warn(...);
return ret;
}
Above snd_soc_dapm_add_route() (= X) is called from
snd_soc_dapm_add_routes() (= Y).
(X) will indicate error message by itself, but (Y) will indicate
own error message at (C). (C) is duplicated.
(Y) int snd_soc_dapm_add_routes(...)
{
...
for (...) {
(X) int r = snd_soc_dapm_add_route(...);
if (r < 0) {
(C) dev_err(...);
ret = r;
}
...
}
...
}
This patch (1) merges these error message (= A,B) into one,
(2) use dev_err(), (3) remove duplicate error message (= C) from
snd_soc_dapm_add_routes().
By this patch, it will indicate error message like this.
- error message with prefix
- not found widget will have "(*)" mark
- it indicates [control] if exists.
ex)
[if no sink with control]
ASoC: Failed to add route SOURCE -> [CTRL] -> SINK(*)
[if no source without control]
ASoC: Failed to add route SOURCE(*) -> SINK
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/878rlctzt5.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Merge series from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
All Intel drivers for cAVS platforms contain a sequence for S3 resume
which doesn't seem justified nor necessary. Forensic Git investigation
in internal repositories did not provide any rationale for the
implementation, and tests show no impact when those sequences are
removed.
This sequence was identified as problematic during a large HDaudio
cleanup where all programming sequences were revisited before
extensions are added.
Merge series from Doug Brown <doug@schmorgal.com>:
This series adds devicetree support for the Wolfson WM8961 codec. The
first patch adds a schema, the second patch hooks it up in the code, and
the third patch allows it to be selected in Kconfig.
Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:
Two fixes are leading the way - one addresses the incorrect DMA mask
assignment (typo) at driver probe. The other, fixes a potential buffer
overflow when copying data received from firmware to kernel buffer.
However unlikely, the fix should still be there.
Then a range of patches providing the support for:
- AML with rt286 (machine board)
- KBL-R for rt298 (codec)
- KBL-R with rt298 (machine board)
- APL/KBL with da7219 (machine board)
- Addition of all the missing SKL-based PCI ids to core.c
Of the remaining changes, only one stands out - special case is provided
for "unsupported" IPCs. The driver supports a range of platforms,
however, on some generations given IPC may not be supported. Such call
shall not be treated as "invalid" - those are two different scenarios.
Everything else in the patchset is mostly a readability improvement:
spelling fixes and log messages issues, code simplification.
Merge series from Chancel Liu <chancel.liu@nxp.com>:
At a previous time, we have successfully created a virtual sound card
based on rpmsg. The sound card works under this mechanism Cortex-A core
tells the Cortex-M core the format, rate, channel, .etc configuration
of the PCM parameters and Cortex-M controls real hardware devices such
as SAI and DMA. From the view of Linux side, the sound card is bound to
a rpmsg channel through which it can access SAI.
Here these patches are introduced to create a new virtual sound card to
access MICFIL based on a new created rpmsg channel. It's easy to create
a new rpmsg channel for MICFIL through rpmsg name service announcment.
Also the other ASoC components bound to this rpmsg MICFIL sound card
will be registered with these patches.
If other sound cards using different hardware devices needs to be
created over rpmsg in the future, these patches can be referred.
Merge series from Siarhei Volkau <lis8215@gmail.com>:
The patchset fixes:
- Line In path stays powered off during capturing or
bypass to mixer.
- incorrectly represented dB values in alsamixer, et al.
- incorrect represented Capture input selector in alsamixer
in Playback tab.
- wrong control selected as Capture Master
The patchset improves:
- Exposes output stage (post mixer) gain control and makes it new
Master playback gain, DAC gain was the previous master.
However, no Master mute now.
- Exposes all mixer inputs (both Mics, LineIn and DAC) with their
gain controls.
- Exposes microphones widgets: single/differential input, boost.
Known issues:
- Bypass path enablement isn't applied immediately, for make
things going bit clock needs to be triggered for a bit,
e.g. by aplay dummy.wav
It might be a hardware bug, since the bit clock isn't
declared as required for codec operation.
Tested on:
- Ritmix RZX-27 (jz4725b).
- Ritmix RZX-50 (jz4755).
Merge series from Shang XiaoJing <shangxiaojing@huawei.com>:
Use DIV_ROUND_UP() instead of open-coding it, which intents and makes
it more clear what is going on for the casual reviewer.
Merge series from Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>:
This series adds machine board for max98927 codec present on some
Chromebook devices.
The Mixer couples analog input from 4 sources (DAC, Line In, Mic 1,
Mic 2) each input has its own gain & mute controls.
At the moment only DAC is implemented fully and Line In path can be
switched on/off. The patch implements Mic 1 and Mic 2 paths and fully
implements Line In path.
Manual states that these controls (16.6.3.3 Programmable attenuation:
GOi) gain varies from -22.5dB to +6.0dB with 1.5dB step. Also there's
extra values below the minimum, but they behave the same as the minimum
value.
Signed-off-by: Siarhei Volkau <lis8215@gmail.com>
Link: https://lore.kernel.org/r/20221016132648.3011729-7-lis8215@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
At the moment DAC control is used as Master Playback control, this is
incorrect as DAC's output goes to a Mixer.
The Mixer couples analog input from 4 sources (DAC, Line In, Mic 1,
Mic 2) each input has its own gain & mute controls.
Output of the Mixer goes to Output Stage control which is the best suite
for the new Playback Master. However, it doesn't have mute capability.
The patch implements Output Stage control as a new master and renames
old Master to DAC. Rest of mixer inputs will be implemented in the next
patch in the series.
Manual states that Output stage control (16.6.3.5 Programmable output
amplifier: PGAT) gain varies from -33.5dB to +4.5dB, gain step isn't
fixed and has 3 opts.
Signed-off-by: Siarhei Volkau <lis8215@gmail.com>
Link: https://lore.kernel.org/r/20221016132648.3011729-6-lis8215@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Line In Bypass control is used as Master Capture at the moment
this is completely incorrect.
Current control routed to Mixer instead of ADC, thus can't affect
Capture path. ADC control shall be used instead.
ADC volume control parameters are different, so the patch fixes that
as well. Manual says (16.6.3.2 Programmable input attenuation amplifier:
PGATM) that gain varies in range 0dB..22.5dB with 1.5dB step.
Signed-off-by: Siarhei Volkau <lis8215@gmail.com>
Link: https://lore.kernel.org/r/20221016132648.3011729-4-lis8215@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
DAC volume control is the Master Playback Volume at the moment
and it reports wrong levels in alsamixer and other alsa apps.
The patch fixes that, as stated in manual on the jz4725b SoC
(16.6.3.4 Programmable attenuation: GOD) the ctl range varies
from -22.5dB to 0dB with 1.5dB step.
Signed-off-by: Siarhei Volkau <lis8215@gmail.com>
Link: https://lore.kernel.org/r/20221016132648.3011729-3-lis8215@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The audio unit of Marvell Armada38x SoC is similar to the ones comprised by
other Marvell SoCs (Kirkwood, Dove and Armada 370). Therefore KW audio
driver can be used to support it and this commit adds new compatible string
to identify Armada 38x variant.
Two new memory regions are added: first one for PLL configuration and
the second one for choosing one of audio I/O modes (I2S or S/PDIF).
For the latter purpose a new optional DT property is added ('spdif-mode').
kirkwood-i2s driver is extended by adding a new init function for Armada
38x flavor and also a routine that enables PLL output (i.e. MCLK)
configuration.
Signed-off-by: Marcin Wojtas <mw@semihalf.com>
Tested-by: Star_Automation <star@marvell.com>
Reviewed-by: Nadav Haklai <nadavh@marvell.com>
Reviewed-by: Lior Amsalem <alior@marvell.com>
Tested-by: Lior Amsalem <alior@marvell.com>
Signed-off-by: Hezi Shahmoon <hezi@marvell.com>
Reviewed-by: Neta Zur Hershkovits <neta@marvell.com>
[pali: Fix support for pre-38x SoCs]
Signed-off-by: Pali Rohár <pali@kernel.org>
Reviewed-by: Andrew Lunn <andrew@lunn.ch>
Link: https://lore.kernel.org/r/20220926110533.13475-1-pali@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
The Hardware Voice Activity Detector (HWVAD) is a block
responsible for detect voice activity in a channel selected
by the user. It can be configured in Envelope-based or
Energy-based mode.
There are additional two interrupts for HWVAD, one is event
interrupt, another is error interrupt.
Enable hwvad in parallel with recording.
when voice activity detected, HWVAD will be disabled and
alsa control notification is triggerred.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1663925494-9941-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
