The negative error return from the call to to_sndif_format is being
assigned to an unsigned 8 bit integer and hence the check for a negative
value is always going to be false. Fix this by using ret as the error
return and hence the negative error can be detected and assign
the u8 sndif_format to ret if there is no error.
Detected by CoverityScan, CID#1469385 ("Unsigned compared against 0")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamoccchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error for a -ve value in ret is redundant as all previous
assignments to ret have an associated -ve check and hence it
is impossible for ret to be less that zero at the point of the
check. Remove this redundant error check.
Detected by CoveritScan, CID#1469407 ("Logically Dead code")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Convert the S_<FOO> symbolic permissions to their octal equivalents as
using octal and not symbolic permissions is preferred by many as more
readable.
see: https://lkml.org/lkml/2016/8/2/1945
Done with automated conversion via:
$ ./scripts/checkpatch.pl -f --types=SYMBOLIC_PERMS --fix-inplace <files...>
Miscellanea:
o Wrapped one multi-line call to a single line
Signed-off-by: Joe Perches <joe@perches.com>
Acked-by: Vinod Koul <vkoul@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At present, all of models produced by TC Electronic except for Konnekt Live
are supported with hard-coded their stream formats. Studio Konnekt 48 is
sore model to support dual streams for both directions. The second stream
has no MIDI conformant data channel in its data block. But current
implementation transfers the second stream with MIDI conformant data
channel.
This commit fixes this issue.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In MediaTek SoC chip we have multiple DAI,
such as I2S, ADDA, PCM, etc.
Organize each DAI in to one sub dai,
with its dai driver, controls, widgets, routes.
add mtk_afe_combine_sub_dai() to combine
dai driver from each DAI.
add mtk_afe_add_sub_dai_control() to register
the control, widget, routes to component.
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The newly introduced driver causes a harmless Kconfig warning when
compile-testing random configurations:
WARNING: unmet direct dependencies detected for SND_SDMA_SOC
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && DMA_OMAP [=n]
Selected by [y]:
- SND_OMAP_SOC [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && (ARCH_OMAP [=y] && DMA_OMAP [=n] || ARM [=y] && COMPILE_TEST [=y])
By simply allow build testing without DMA_OMAP, we can shut up that warning.
Fixes: dde637f2da ("ASoC: omap: Introduce the generic_dmaengine_pcm based sdma-pcm")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Before commit 3b5b899ca6 ("ALSA: hda: Make use of core codec functions
to sync power state"), hda_set_power_state() returned the response to
the Get Power State verb, a 32-bit unsigned integer whose expected value
is 0x233 after transitioning a codec to D3, and 0x0 after transitioning
it to D0.
The response value is significant because hda_codec_runtime_suspend()
does not clear the codec's bit in the codec_powered bitmask unless the
AC_PWRST_CLK_STOP_OK bit (0x200) is set in the response value. That in
turn prevents the HDA controller from runtime suspending because
azx_runtime_idle() checks that the codec_powered bitmask is zero.
Since commit 3b5b899ca6, hda_set_power_state() only returns 0x0 or
0x1, thereby breaking runtime PM for any HDA controller. That's because
an inline function introduced by the commit returns a bool instead of a
32-bit unsigned int. The change was likely erroneous and resulted from
copying and pasting snd_hda_check_power_state(), which is immediately
preceding the newly introduced inline function. Fix it.
Link: https://bugs.freedesktop.org/show_bug.cgi?id=106597
Fixes: 3b5b899ca6 ("ALSA: hda: Make use of core codec functions to sync power state")
Cc: Alex Deucher <alexander.deucher@amd.com>
Cc: Abhijeet Kumar <abhijeet.kumar@intel.com>
Reported-and-tested-by: Gunnar Krüger <taijian@posteo.de>
Signed-off-by: Lukas Wunner <lukas@wunner.de>
Acked-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop pci_device() macro that just leads to chip->pci->dev, and pass it
directly to request_firmware(). It was introduced for allowing the
external alsa-driver kernel module builds. Since it was discontinued
years ago, we should clean it up now.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Drop the superfluous #ifndef checks that had been put just for
allowing building the alsa-driver kernel modules externally.
Since the external build was discontinued years ago, let's clean up
the old kludges.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power-saving is causing plops on audio start/stop on ASRock H81M-HDS
machines, add these to the power_save blacklist.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power-saving is causing plops on audio start/stop on Gigabyte
P55A-UD3 and Gigabyte Z87-D3HP machines, add these to the power_save
blacklist.
Note these 2 boards both use 1458:a002 as subsystem ids, so they share
a single entry.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1525104
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power-saving is causing a plop and silences the first 2 seconds
(give or take) of audio, silencing notifications sounds on Medion /
Clevo W35xSS_370SS laptops.
Add the Clevo W35xSS_370SS to the power_save blacklist.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1581607
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Power-saving is causing a humming sound when active on the Intel
NUC7i3BNB, add it to the blacklist.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1520902
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The nau8824 codec can detect whether a headset or plain headphones is
inserted (as well as button presses on the headset) as such the jack_type
passed to snd_soc_card_jack_new() should include SND_JACK_MICROPHONE.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
some monitors care about the parity bit in the sub-frame of I2S,
but the cdn-dp always set this bit to "1", so these monitors
do not have sound output if use i2s, use spdif can fix this issue.
Signed-off-by: Chris Zhong <zyw@rock-chips.com>
Signed-off-by: Lin Huang <hl@rock-chips.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
HP Z2 G4 requires the same workaround as other HP machines that have
no mic-pin detection.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
FOr platforms that use the simple-card driver, the codec cannot be selected
through 'select' magic in Kconfig. So turn this into a real config option.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This is needed when the codec is instanciated from from a device tree.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
There's no need to read the register again prior to writing it, we did
that in the beginning of the function.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
The pxa-ssp driver currently assumes that .set_fmt() is called before
.set_clkdiv(), .set_pll() etc.
Commit a8bd0ee558 ("ASoC: raumfeld: Use static DAI format setup") broke
support for Raumfeld hardware (and possible other PXA based ones) because
it effectively changed the order of these calls. Also, as the call to
.set_fmt() is now done at probe time, the port clock is not yet enabled.
To fix this, strip all hardware register access code from the .set_fmt()
callback and memorize the desired value, so we can use it from the
.hw_params() callback. Also make the .set_fmt() callback less destructive
by reading all registers that it writes to in the beginning and only
masking out the bits that it possibly fiddles with.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
match_string() returns the index of an array for a matching string,
which can be used intead of open coded variant.
Signed-off-by: Yisheng Xie <xieyisheng1@huawei.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support to DB820c machine driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to q6asm dai driver which configures Q6ASM streams
to pass pcm data.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to q6afe backend dais driver.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch add support to MI2S mixers required to select path between
ASM stream and AFE ports.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to SLIMBus related mixers to control mux between
ASM stream and AFE port.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to q6 routing driver which configures route
between ASM and AFE module using ADM apis.
This driver uses dapm widgets to setup the matrix between AFE ports and
ASM streams.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to open, write and media format commands
in the q6asm module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to memory map and unmap regions commands in
q6asm module.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds basic support to Q6 ASM (Audio Stream Manager) module on
Q6DSP. ASM supports up to 8 concurrent streams. each stream can be setup
as playback/capture. ASM provides top control functions like
Pause/flush/resume for playback and record. ASM can Create/destroy encoder,
decoder and also provides POPP dynamic services.
This patch adds support to basic features to allow hdmi playback.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to Q6ADM (Audio Device Manager) module in
q6dsp. ADM performs routing between audio streams and AFE ports.
It does Rate matching for streams going to devices driven by
different clocks, it handles volume ramping, Mixing with channel
and bit-width. ADM creates and destroys dynamic COPP services
for device-related audio processing as needed.
This patch adds basic support to ADM.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
With in ACP, There are three I2S controllers can be
configured/enabled ( I2S SP, I2S MICSP, I2S BT).
Default enabled I2S controller instance is I2S SP.
This patch provides required changes to support I2S BT
controller Instance.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
System clock on the platform is 25Mhz and not 24Mhz.
PLL_OUT for da7219 codec to use DA7219_PLL_FREQ_OUT_98304
as it is for 48KHz SR.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
hw_param can be called multiple times and thus we can have
more clk enable. The clk may not get diabled due to refcounting.
startup/shutdown ensures single clk enable/disable call.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
rtd structure freed early may result in kernel panic in dma close
call back. moved releasing memory for rtd structure to the end of
dma close callback.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Added sram bank variable to audio_substream_data structure.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Added pte offset variable in audio_substream_data structure.
Added Stoney related PTE offset macros in acp header file.
Modified hw_params callback to assign the pte offset value
based on asic_type.
PTE Offset macros used to calculate no of PTE entries
need to be programmed when memory allocated for audio buffer.
Depending upon allocated audio buffer size, PTE offset values
will change.
Compared to CZ, Stoney has SRAM memory limitation i.e 48k
It is required to define separate PTE Offset macros for
Stoney.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Reviewed-by: Daniel Kurtz <djkurtz@chromium.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to LPASS Bit clock, LPASS Digital
core clock and OSR clock. These clocks are required for both
MI2S and PCM setup.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to 4 MI2S ports on LPASS.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to 6 SLIMBus AFE ports, which are used as
backend dais.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
This patch adds support to Q6AFE (Audio Front End) module on Q6DSP.
AFE module sits right at the other end of cpu where the codec/audio
devices are connected.
AFE provides abstraced interfaces to both hardware and virtual devices.
Each AFE tx/rx port can be configured to connect to one of the hardware
devices like codec, hdmi, slimbus, i2s and so on. AFE services include
starting, stopping, and if needed, any configurations of the ports.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Log the correct error code in case the .open() call to a component fails.
Signed-off-by: Daniel Mack <daniel@zonque.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Trivial fix to spelling mistake in snprintf literal string
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
HID is assumed to be made of TI PCI ID (0x104C) + part number, so all
four 104C5121, 104C5122, 104C5141 104C5142 are valid.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no platform code that uses i2c module table.
Remove it altogether and adjust ->probe() to be ->probe_new().
Signed-off-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The "entry" pointer is always non-NULL so this test for out of bounds
won't work.
Fixes: f1f0f330b1 ("ALSA: dice: add parameters of stream formats for models produced by TC Electronic")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make C-header and SPDX-License-Identifier header uniform.
Signed-off-by: Marco Felsch <m.felsch@pengutronix.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Trivial fix to spelling mistakes in SND_SOC_BYTES literal strings
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Trivial fix to spelling mistake in SOC_ENUM literal string
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
--nextPart3916812.EicPReet6m
Content-Transfer-Encoding: 7Bit
Content-Type: text/plain; charset="us-ascii"
Mytek manufactures some equipment with DICE-based firewire ports. These
devices contain old versions of DICE firmware which lacks detailed
stream format reporting for all sampling clock modes.
Building upon the recent work by Takashi Sakamoto, hard-coded parameters
are added for the Stereo 192 DSD-DAC. When the device vendor and model
match the coded parameters are copied into the stream format cache.
Signed-off-by: Melvin Vermeeren <mail@mel.vin>
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There are still many places calling the timer's hw.c_resolution
callback without lock, and this may lead to some races, as we faced in
the commit a820ccbe21 ("ALSA: pcm: Fix UAF at PCM release via PCM
timer access").
This patch changes snd_timer_resolution() to take the timer->lock for
avoiding the races. A place calling this function already inside the
lock (from the notifier) is replaced with the
snd_timer_hw_resolution() accordingly, as well as wrapping with the
lock around another place calling snd_timer_hw_resolution(), too.
Reported-by: Ben Hutchings <ben.hutchings@codethink.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of open-coding for getting the timer resolution, use the
standard snd_timer_resolution() helper.
The original code falls back to the callback function when the
resolution is zero, but it must be always so when the callback
function is defined. So this should be no functional change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There multiple open-codes to get the hardware timer resolution.
Make a local helper function snd_timer_hw_resolution() and call it
from all relevant places.
There is no functional change by this, just a preliminary work for the
following timer resolution hardening patch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Commit f65e0d2998 ("ALSA: timer: Call notifier in the same spinlock")
combined the start/continue and stop/pause functions, and in doing so
changed the event code for the pause case to SNDRV_TIMER_EVENT_CONTINUE.
Change it back to SNDRV_TIMER_EVENT_PAUSE.
Fixes: f65e0d2998 ("ALSA: timer: Call notifier in the same spinlock")
Signed-off-by: Ben Hutchings <ben.hutchings@codethink.co.uk>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
UAC3 channel map is created during interface parsing,
and in some cases was not freed in failure paths.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The error messages at sanity checks of memory pages tend to repeat too
many times once when it hits, and without the rate limit, it may flood
and become unreadable. Replace such messages with the *_ratelimited()
variant.
Bugzilla: http://bugzilla.opensuse.org/show_bug.cgi?id=1093027
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set PR-38 register to 0x1fe1 will make PLL function more stable.
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The ssm2305 is a simple Class-D audio amplifier. A application can
turn on/off the device by a gpio. It's also possible to hardwire the
shutdown pin.
Tested on a i.MX6 based custom board.
Signed-off-by: Marco Felsch <m.felsch@pengutronix.de>
Reviewed-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Trivial fix to spelling mistakes in audigy_outs arrays.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds support to core apr service, which is used to query
status of other static and dynamic services on the dsp.
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-and-tested-by: Rohit kumar <rohitkr@codeaurora.org>
Acked-by: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
These duplicate includes have been found with scripts/checkincludes.pl but
they have been removed manually to avoid removing false positives.
Signed-off-by: Pravin Shedge <pravin.shedge4linux@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As a side effect of the following commit, the active TX
serializer may get disabled which may result in distorted
audio output.
ASoC: davinci-mcasp: Add support for multichannel playback
(2952b27e2e)
For example, if a 4 channel I2S playback with two TX serializers
is activated. Later on, if a recording of 2 channels, with only 1 RX
serializer is started, which will also disable one of the TX
serializer because max_active_serializers is only calculated for
RX (recording) stream. This patch fixes this issue.
Signed-off-by: Vishal Thanki <vishalthanki@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Many X86 devices using a BYT SoC + RT5640 codec are cheap devices with
generic DMI strings, causing snd_soc_set_dmi_name() to fail to set a
long_name, making it impossible for userspace to have a correct UCM
profile which only uses inputs / outputs which are actually hooked up
on the device.
Our quirks already specify which input the internal mic is connected to
and if a single (mono) speaker is used or if the device has stereo
speakers.
This commit sets a long_name based on the quirks so that userspace can
have UCM profiles doing the right thing based on the long_name.
Note that if we ever encounter the need for a special UCM profile for
some device we can add a quirk to set a specific long_name for the
device,
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Even with our recently tweaked defaults, quite a few bytcr_rt5640 devices
still need quirks to be fully functional. This commits adds quirks where
necessary for the 16 bytcr_rt5640 devices I have access to.
The quirks are added for the following reasons:
1) Devices with only one speaker need the mono quirk to avoid driving an
unused and potentially short-circuited output. 8 of my sample of 16 devs
are mono, 4 of these would work with the defaults if it were not for their
mono speaker.
2) Devices using a different input for the internal mic then the default,
this is the case for 6 of my sample of 16 devices.
3) BYTCR devices without an ACPI channel map, which do not work with the
default of SSP0-AIF2, this is the case for 2 of my sample of 16 devices.
4) Devices which need non-default jack-detect settings, this is the case
for 6 of my sample of 16 devices.
This commit add quirks for the following devices:
Acer Iconia Tab 8 W1-810
Chuwi Vi8
HP Pavilion X2 10-n000nd
HP Stream 7
I.T. Works TW891
Lamina I8270
MSI S100
Pipo W4
PoV-mobii-800w (v2.0)
PoV-mobii-800w (v2.1)
Toshiba Click Mini L9W-B
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use dmi_first_match() instead of dmi_check_system() + callbacks, this
avoid the need to initialize dmi_system_id.callback for each
byt_rt5640_quirk_table entry.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
As we add more quirks it is useful to have some sort of order in the
quirk list, sort it alphabetically.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Out of the 11 BYTCR devices which I have access to for testing, 6 use
JD1IN4P for jack-detect, 2 use JD1IN4P non-inverted and the other 3 use
JD2IN4N, the ones not using JD1IN4P are all also special in other ways and
need a DMI quirk regardless.
All 5 BYT (non CR) devices which I have access to use JD2IN4N.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently we've 2 places with BYTCR defaults: 1. The generic catch-all
DMI_SYS_VENDOR=="Insyde" DMI quirk which selects SSP0-AIF1 for generic
Insyde BYTCR tablets without the ACPI channel package; and 2. the
defaults in the if (is_bytcr) {} code block.
Currently these are not identical, both select IN3 as the internal mic
output, but the "Insyde" DMI quirk leaves out the DIFF_MIC quirk. The
DIFF_MIC quirk should be enabled by default, because enabling diff. input
helps a lot for devices with a differential mic, where as it is a nop on
devices with a normal mic.
This commit adds the DIFF_MIC quirk to the "Insyde" DMI quirk path, by
adding a new BYTCR_INPUT_DEFAULTS define and using that in both code paths
which set BYTCR defaults.
Having a single place where the BYTCR input defaults are defined also
allows defining jack-detect defaults in a single place in a follow-up
commit.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Out of the 11 BYTCR devices which I have access to for testing,
7 use IN3 for the internal mic and only 1 uses IN1 for the internal mic,
the other 3 use DMIC1.
So IN3 clearly is a better default, using IN3 as default avoids the need
to add DMI quirks for some of these devices.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add code to support setting jack-detect parameters through quirks and
extend the existing DMI quirk table entries for the Asus T100TA and the
Dell Venue 8 Pro 5830 to enable jack detection.
Tested-by: Hans de Goede <hdegoede@redhat.com>
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
This fixes the following 3 issues:
1) The sys_vendor match should be for "Dell Inc." not "DellInc.",
without this fixed the quirk never gets applied
2) DMIC1 is used not DMIC2, this was not a problem sofar because for
regular BYT boards (rather then BYTCR) we default to DMIC1 and because
of 1. the quirk was not being applied
3) The Dell Venue 8 5830 Pro only has a single speaker
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Use device-properties for setting up the dmic, based on the
BYT_RT5640_MAP() value, instead of using the codec specific
rt5640_dmic_enable() function for this. This also removes the need
for the BYT_RT5640_DMIC_EN quirk, which was always set together with
a MAP() quirk of DMIC1_MAP or DMIC2_MAP.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The commit 289ca025ee ("ALSA: Use priority list for managing device
list") changed the way to register/disconnect/free devices via a
single priority list. This helped to make behavior consistent, but it
also changed a slight behavior change: namely, the control device is
registered earlier than others, while it was supposed to be the very
last one.
I've put SNDRV_DEV_CONTROL in the current position as the release of
ctl elements often conflict with the private ctl elements some PCM or
other components may create, which often leads to a double-free.
But, the order of register and disconnect should be indeed fixed as
expected in the early days: the control device gets registered at
last, and disconnected at first.
This patch changes the priority list order to move SNDRV_DEV_CONTROL
as the last guy to assure the register / disconnect order. Meanwhile,
for keeping the messy resource release order, manually treat the
control and lowlevel devices as last freed one.
Additional note:
The lowlevel device is the device where a card driver creates at
probe. And, we still keep the release order control -> lowlevel, as
there might be link from a control element back to a lowlevel object.
Fixes: 289ca025ee ("ALSA: Use priority list for managing device list")
Reported-by: Tzung-Bi Shih <tzungbi@google.com>
Tested-by: Tzung-Bi Shih <tzungbi@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes: 9958e8afbcad ("ASoC: rt5663: Use the set_jack() instead of the export function")
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The patch replaces the export function with the new API set_jack().
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
The sdma-pcm does not need any information from omap-dma.h, it only needs
to include the omap-dmaengine.h - for the omap_dma_filter_fn, but that
might not be needed at all as OMAP1 was converted to dma_slave_map, but
I can not test OMAP1.
Add the linux/device.h include as well for devm_kzalloc()
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
WARNING: modpost: missing MODULE_LICENSE() in sound/soc/omap/snd-soc-sdma.o
see include/linux/module.h for more information
WARNING: modpost: missing MODULE_LICENSE() in sound/soc/omap/snd-soc-sdma.o
see include/linux/module.h for more information
Add the missing MODULE_LICENSE.
This patch also going to solve:
snd_soc_sdma: Unknown symbol devm_kmalloc (err 0)
snd_soc_sdma: Unknown symbol omap_dma_filter_fn (err 0)
snd_soc_sdma: Unknown symbol snd_dmaengine_pcm_prepare_slave_config (err 0)
snd_soc_sdma: Unknown symbol devm_snd_dmaengine_pcm_register (err 0)
Reported-by: Tony Lindgren <tony@atomide.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
retire_capture_urb() may print warning messages when the given URB
doesn't align, and this may flood the system log easily.
Put the rate limit to the message for avoiding it.
Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1093485
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Implement essential initialization of the sound driver:
- introduce required data structures
- handle driver registration
- handle sound card registration
- register sound driver on backend connection
- remove sound driver on backend disconnect
Initialize virtual sound card with streams according to the
Xen store configuration.
Implement ALSA driver operations including:
- manage frontend/backend shared buffers
- manage Xen bus event channel states
Implement requests from front to back for ALSA
PCM operations.
- report ALSA period elapsed event: handle XENSND_EVT_CUR_POS
notifications from the backend when stream position advances
during playback/capture. The event carries a value of how
many octets were played/captured at the time of the event.
- implement explicit stream parameter negotiation between
backend and frontend: handle XENSND_OP_HW_PARAM_QUERY request
to read/update configuration space for the parameter given:
request passes desired parameter interval and the response to
this request returns min/max interval for the parameter to be used.
Signed-off-by: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Handle Xen event channels:
- create for all configured streams and publish
corresponding ring references and event channels in Xen store,
so backend can connect
- implement event channels interrupt handlers
- create and destroy event channels with respect to Xen bus state
Signed-off-by: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Read configuration values from Xen store according
to xen/interface/io/sndif.h protocol:
- introduce configuration structures for different
components, e.g. sound card, device, stream
- read PCM HW parameters, e.g rate, format etc.
- detect stream type (capture/playback)
- read device and card parameters
Signed-off-by: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Introduce skeleton of the para-virtualized Xen sound
frontend driver.
Initial handling for Xen bus states: implement
Xen bus state machine for the frontend driver according to
the state diagram and recovery flow from sound para-virtualized
protocol: xen/interface/io/sndif.h.
Signed-off-by: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com>
Reviewed-by: Juergen Gross <jgross@suse.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This module has a table for parameters of each effects. This table is
read-only and can have 'const' qualifier.
This commit adds this optimization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This module has some function-local strings just for printk therefore
it can be merged into format string.
This commit applies this optimization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This module has some strings just for printk therefore they can be
read-only.
This commit applies this optimization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An array of templates for control element set is passed as an
argument for snd_hda_add_new_ctls(). This argument has 'const'
qualifier therefore the passed array can have the qualifier.
This commit adds this optimization.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Obtain the number of channels for the Input Terminal from the
Logical Cluster Descriptor. This achieves a useful minimal parsing
of this unit so it can be used in other units in the topology.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Tested-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds support for the UAC3 insertion controls. The status
is reported as a boolean value in the same way it used to do
for UAC2. Hence, the presence of any connector in the response
will make the control saying the jack is connected.
The UAC2 support for this control has been moved to a dedicated
control for connectors as both UAC2 and UAC3 follow a specific
Control Request Parameter Block for this control. This parameter
block for UAC3 could not be read in the same simplistic
manner as in UAC2.
This implementation is not requesting additional information
from the HIGH CAPABILITY Connectors descriptor.
Tested with an UAC3 device with UAC2 as legacy configuration.
The connector status can be read with `amixer` and the interrupt
is also caught with `alsactl monitor`.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Tested-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This adds support for the MIXER UNIT in UAC3. All the information
is obtained from the (HIGH CAPABILITY) Cluster's header. We don't
read the rest of the logical cluster to obtain the channel config
as that wont make any difference in the current mixer behaviour.
The name of the mixer unit is not yet requested as there is not
support for the UAC3 Class Specific String requests.
Tested in an UAC3 device working as a HEADSET with a basic mixer
unit (same as the one in the BADD spec) with no controls.
Signed-off-by: Jorge Sanjuan <jorge.sanjuan@codethink.co.uk>
Reviewed-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Tested-by: Ruslan Bilovol <ruslan.bilovol@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai