IF YOU WOULD LIKE TO GET AN ACCOUNT, please write an
email to Administrator. User accounts are meant only to access repo
and report issues and/or generate pull requests.
This is a purpose-specific Git hosting for
BaseALT
projects. Thank you for your understanding!
Только зарегистрированные пользователи имеют доступ к сервису!
Для получения аккаунта, обратитесь к администратору.
The initial machine driver supports only j721e-cpb and the ivi addon, but
other EVMs for different K3 SoC can have similar audio setup which can
be supported by the driver with small or no modification.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20200908113204.12012-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
SND_SOC_J721E_EVM should not select SND_SOC_PCM3168A_I2C when I2C
is not enabled. That causes build errors, so make this driver's
symbol depend on I2C.
WARNING: unmet direct dependencies detected for SND_SOC_PCM3168A_I2C
Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && I2C [=n]
Selected by [m]:
- SND_SOC_J721E_EVM [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && (DMA_OMAP [=y] || TI_EDMA [=m] || TI_K3_UDMA [=n] || COMPILE_TEST [=y]) && (ARCH_K3_J721E_SOC [=n] || COMPILE_TEST [=y])
../sound/soc/codecs/pcm3168a-i2c.c:59:1: warning: data definition has no type or storage class
module_i2c_driver(pcm3168a_i2c_driver);
^~~~~~~~~~~~~~~~~
../sound/soc/codecs/pcm3168a-i2c.c:59:1: error: type defaults to ‘int’ in declaration of ‘module_i2c_driver’ [-Werror=implicit-int]
../sound/soc/codecs/pcm3168a-i2c.c:59:1: warning: parameter names (without types) in function declaration
../sound/soc/codecs/pcm3168a-i2c.c:49:26: warning: ‘pcm3168a_i2c_driver’ defined but not used [-Wunused-variable]
static struct i2c_driver pcm3168a_i2c_driver = {
^~~~~~~~~~~~~~~~~~~
cc1: some warnings being treated as errors
Fixes: 6748d05590 ("ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI)")
Signed-off-by: Randy Dunlap <rdunlap@infradead.org>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Mark Brown <broonie@kernel.org>
Link: https://lore.kernel.org/r/e74c690c-c7f8-fd42-e461-4f33571df4ef@infradead.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Link: https://lore.kernel.org/r/20200718112403.13709-1-grandmaster@al2klimov.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
Signed-off-by: Alexander A. Klimov <grandmaster@al2klimov.de>
Link: https://lore.kernel.org/r/20200718110857.11520-1-grandmaster@al2klimov.de
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_soc_dai_digital_mute() is internally using both
mute_stream() (1) or digital_mute() (2), but the difference between
these 2 are only handling direction.
We can merge digital_mute() into mute_stream
int snd_soc_dai_digital_mute(xxx, int direction)
{
...
else if (dai->driver->ops->mute_stream)
(1) return dai->driver->ops->mute_stream(xxx, direction);
else if (direction == SNDRV_PCM_STREAM_PLAYBACK &&
dai->driver->ops->digital_mute)
(2) return dai->driver->ops->digital_mute(xxx);
...
}
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87blkpxxip.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Looks like 'w' has remained unchecked since the driver's inception.
Fixes the following W=1 kernel build warning(s):
sound/soc/ti/omap-mcbsp-st.c: In function ‘omap_mcbsp_st_chgain’:
sound/soc/ti/omap-mcbsp-st.c:145:6: warning: variable ‘w’ set but not used [-Wunused-but-set-variable]
Peter suggested that the whole read can be removed, so that's
been done too.
Signed-off-by: Lee Jones <lee.jones@linaro.org>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Peter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Jarkko Nikula <jarkko.nikula@bitmer.com>
Cc: Samuel Ortiz <samuel.ortiz@nokia.com>
Cc: linux-omap@vger.kernel.org
Link: https://lore.kernel.org/r/20200707190612.97799-10-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Add the missing unlock before return from function j721e_audio_hw_params()
in the error handling case.
Fixes: 6748d05590 ("ASoC: ti: Add custom machine driver for j721e EVM (CPB and IVI)")
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20200703030910.75047-1-weiyongjun1@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The audio support on the board is using pcm3168a codec connected to McASP10
serializers in parallel setup.
The pcm3168a SCKI clock is coming via the j721e AUDIO_REFCLK2 pin.
In order to support 48KHz and 44.1KHz family of sampling rates the parent clock
for AUDIO_REFCLK2 needs to be changed between PLL4 (for 48KHz) and PLL15 (for
44.1KHz). The same PLLs are used for McASP10's AUXCLK clock via different
HSDIVIDER.
Generic card can not be used for the board as we need to switch between
clock paths for different sampling rate families and also need to change
the slot_width between 16 and 24 bit audio.
The audio support on the Infotainment Expansion Board consists of McASP0
connected to two pcm3168a codecs with dedicated set of serializers to each.
The SCKI for pcm3168a is sourced from j721e AUDIO_REFCLK0 pin.
It is extending the audio support on the CPB.
Due to the fact that the same PLL4/15 is used by both domains (CPB/IVI)
there are cross restriction on sampling rates.
The IVI side is represented as multicodec setup.
PCMs available on a plain CPB (no IVI addon):
hw:0,0 - cpb playback (8 channels)
hw:0,1 - cpb capture (6 channels)
When the IVI addon is present, additional two PCMs will be present:
hw:0,2 - ivi multicodec playback (16 channels)
hw:0,3 - ivi multicodec capture (12 channels)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20200630125843.11561-4-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When clk_set_parent() returns an error code, a pairing
runtime PM usage counter increment is needed to keep the
counter balanced.
Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20200525085848.4227-1-dinghao.liu@zju.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
We have snd_soc_dai/dai_stream/component_active() macro
This patch uses it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87mu6a58i3.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
If an error occurs after the call to 'omap_mcbsp_init()', the reference to
'mcbsp->fclk' must be decremented, as already done in the remove function.
This can be achieved easily by using the devm_ variant of 'clk_get()'
when the reference is taken in 'omap_mcbsp_init()'
This fixes the leak in the probe and has the side effect to simplify both
the error handling path of 'omap_mcbsp_init()' and the remove function.
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Acked-by: Peter Ujfalusi <peter.ujflausi@ti.com>
Link: https://lore.kernel.org/r/20200512134325.252073-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
davinci_mcasp_get_dma_type() invokes dma_request_chan(), which returns a
reference of the specified dma_chan object to "chan" with increased
refcnt.
When davinci_mcasp_get_dma_type() returns, local variable "chan" becomes
invalid, so the refcount should be decreased to keep refcount balanced.
The reference counting issue happens in one exception handling path of
davinci_mcasp_get_dma_type(). When chan device is NULL, the function
forgets to decrease the refcnt increased by dma_request_chan(), causing
a refcnt leak.
Fix this issue by calling dma_release_channel() when chan device is
NULL.
Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn>
Signed-off-by: Xin Tan <tanxin.ctf@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/1587818916-38730-1-git-send-email-xiyuyang19@fudan.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Fix the following coccicheck warning:
sound/soc/ti/omap-mcbsp.c:1188:5-11: WARNING: Comparison to bool
Signed-off-by: Jason Yan <yanaijie@huawei.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Link: https://lore.kernel.org/r/20200426094238.23914-1-yanaijie@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
This became again a busy development cycle. There are a few ALSA
core updates (merely API cleanups and sparse fixes), while majority
of other changes are found in ASoC scene.
Here are some highlights:
* ALSA core:
- More helper macros for sparse warning fixes (e.g. bitwise types)
- Slight optimization of PCM OSS locks
- Make common handling for PCM / compress buffers (for SOF)
* ASoC:
- Lots of code refactoring and modernization for (still ongoing)
componentization works
- Conversion of SND_SOC_ALL_CODECS to use imply
- Continued refactoring and fixing of the Intel SOF/SST support,
including the initial (but still incomplete) SoundWire support
- SoundWire and more advanced clocking support for Realtek RT5682
- Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom
DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563
and TLV320ADCX140
* HD-audio:
- Optimizations in HDMI jack handling
- A few new quirks and fixups for Realtek codecs
* USB-audio:
- Delayed registration support
- New quirks for Motu, Kingston, Presonus
-----BEGIN PGP SIGNATURE-----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=fAMQ
-----END PGP SIGNATURE-----
Merge tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This became again a busy development cycle. There are few ALSA core
updates (merely API cleanups and sparse fixes), with the majority of
other changes are found in ASoC scene.
Here are some highlights:
ALSA core:
- More helper macros for sparse warning fixes (e.g. bitwise types)
- Slight optimization of PCM OSS locks
- Make common handling for PCM / compress buffers (for SOF)
ASoC:
- Lots of code refactoring and modernization for (still ongoing)
componentization works
- Conversion of SND_SOC_ALL_CODECS to use imply
- Continued refactoring and fixing of the Intel SOF/SST support,
including the initial (but still incomplete) SoundWire support
- SoundWire and more advanced clocking support for Realtek RT5682
- Support for amlogic GX, Meson 8, Meson 8B and T9015 DAC, Broadcom
DSL/PON, Ingenic JZ4760 and JZ4770, Realtek RL6231, and TI TAS2563
and TLV320ADCX140
HD-audio:
- Optimizations in HDMI jack handling
- A few new quirks and fixups for Realtek codecs
USB-audio:
- Delayed registration support
- New quirks for Motu, Kingston, Presonus"
* tag 'sound-5.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (415 commits)
ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor
Revert "ALSA: uapi: Drop asound.h inclusion from asoc.h"
ALSA: hda/realtek - Remove now-unnecessary XPS 13 headphone noise fixups
ALSA: hda/realtek - Set principled PC Beep configuration for ALC256
ALSA: doc: Document PC Beep Hidden Register on Realtek ALC256
ALSA: hda/realtek - a fake key event is triggered by running shutup
ALSA: hda: default enable CA0132 DSP support
ASoC: amd: acp3x-pcm-dma: clean up two indentation issues
ASoC: tlv320adcx140: Remove undocumented property
ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function
ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver
ASoC: Intel: boards: add sof_sdw machine driver
ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms
ASoC: rt5682: move DAI clock registry to I2S mode
ASoC: pxa: magician: convert to use i2c_new_client_device()
ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities
Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread
ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire
ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks
ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers
...
Call cpu_latency_qos_add/update/remove_request() and
cpu_latency_qos_request_active() instead of
pm_qos_add/update/remove_request() and pm_qos_request_active(),
respectively, because the latter are going to be dropped.
No intentional functional impact.
Signed-off-by: Rafael J. Wysocki <rafael.j.wysocki@intel.com>
Reviewed-by: Ulf Hansson <ulf.hansson@linaro.org>
Acked-by: Mark Brown <broonie@kernel.org>
Acked-by: Takashi Iwai <tiwai@suse.de>
Reviewed-by: Amit Kucheria <amit.kucheria@linaro.org>
Tested-by: Amit Kucheria <amit.kucheria@linaro.org>
The assignment to ret is redundant as it is not used in the error
return path and hence can be removed.
Addresses-Coverity: ("Unused value")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20200210092423.327499-1-colin.king@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
There is no big difference at implementation for .suspend/.resume
between DAI driver and Component driver.
But because some driver is using DAI version, thus ALSA SoC needs
to keep supporting it, hence, framework becoming verbose.
If we can switch all DAI driver .suspend/.resume to Component driver,
we can remove verbose code from ALSA SoC.
Driver is getting its private data via dai->dev.
But dai->dev and component->dev are same dev, thus, we can convert
these. For same reason, we can convert dai->active to
component->active if necessary.
This patch moves DAI driver .suspend/.resume to Component driver
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/871rrvym3p.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We can use snd_soc_dai_link_component to specify codec_conf.
Let's use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/8736dp59ih.wl-kuninori.morimoto.gx@renesas.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
When McASP is master the bclk can be generated from two main source:
AUXCLK: functional clock for McASP or
AHCLK: from external source or internal mux in dra7x family
With this patch it is possible to select between the two source. The patch
is not breaking existing machine drivers since historically the clk_id was
ignored and left as 0 in all cases.
When output clock is configured - which can be only the AHCLK, we select
the AUXCLK as source for the internal HCLK. In this case the HCLK rate is
the same as the output clock.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20191204192005.31210-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
dma_request_slave_channel_reason() is:
#define dma_request_slave_channel_reason(dev, name) \
dma_request_chan(dev, name)
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20191113095445.3211-3-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When non standard names are used it is possible that one of the directions
are not provided, thus the flags needs to be present to tell the core that
we have half duplex setup.
Fixes: 642aafea88 ("ASoC: ti: remove compat dma probing")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20191028115207.5142-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A small smattering of ASoC fixes for v5.4 - nothing too exciting
here, all small standalone things.
-----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAl2JCfITHGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0GP1B/wImLx7u5jHiiAAzeW9kig7R+m91/wm
+QMwcguVLsrTiI8qz2YnH8bMEbLU66bHsy7faKl5hG1r6vnSEct6MTw236eM7l5/
XqLRmnn6W2GtxnpShL/eL3F73i7HYdSUBNsdeV5V9loTMJKehDJ/AqK7k+eMZt0y
5xxEpkiTZilu1ue+GwWN2bhKHu4HnBmESkmbO2Adsws1ZKUNfuhye+IkorllJFeG
v9FmP8j37Z5YKMAUaKmdy4uaecvwJZVpWNuaNNS7A02+6OpeIdrYtNUbj4dnPqGF
U+mOs+Xsnsv09ykBEBdFBk50sKjZnIMeO+cxz/tjT8nL9QI+LCFonf72
=Ijmx
-----END PGP SIGNATURE-----
Merge tag 'asoc-fix-v5.4-rc1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.4
A small smattering of ASoC fixes for v5.4 - nothing too exciting
here, all small standalone things.
SND_SOC_DM365_VOICE_CODEC is a 'bool' option in a choice statement,
meaning it cannot be set to =m, but it selects two other drivers
that we may want to be loadable modules after all:
WARNING: unmet direct dependencies detected for SND_SOC_CQ0093VC
Depends on [m]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m]
Selected by [y]:
- SND_SOC_DM365_VOICE_CODEC [=y] && <choice>
Selected by [m]:
- SND_SOC_ALL_CODECS [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && COMPILE_TEST [=y]
Add an intermediate symbol that sets SND_SOC_CQ0093VC and
MFD_DAVINCI_VOICECODEC to =m if SND_SOC=m.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Link: https://lore.kernel.org/r/20190920075046.3210393-1-arnd@arndb.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Since commit 1137ceee76 ("ARM: OMAP1: ams-delta: Don't request unused
GPIOs"), on-board audio has appeared muted. It has been discovered that
believed to be unused GPIO pins "hookflash1" and "hookflash2" need to be
set low for audible sound in handsfree and handset mode respectively.
According to Amstrad E3 wiki, the purpose of both pins hasn't been
clearly identified. Original Amstrad software used to produce a high
pulse on them when the phone was taken off hook or recall was pressed.
With the current findings, we can assume the pins provide a kind of
audio mute function, separately for handset and handsfree operation
modes.
Commit 2afdb4c41d ("ARM: OMAP1: ams-delta: Fix audio permanently
muted") attempted to fix the issue temporarily by hogging the GPIO pin
"hookflash1" renamed to "audio_mute", however the fix occurred
incomplete as it restored audible sound only for handsfree mode.
Stop hogging that pin, rename the pins to "handsfree_mute" and
"handset_mute" respectively and implement appropriate DAPM event
callbacks for "Speaker" and "Earpiece" DAPM widgets.
Fixes: 1137ceee76 ("ARM: OMAP1: ams-delta: Don't request unused GPIOs")
Signed-off-by: Janusz Krzysztofik <jmkrzyszt@gmail.com>
Reviewed-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190907111650.15440-1-jmkrzyszt@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The davinci McBSP (davinci-i2s) driver does not implement the set_sysclk
callback, which is fine and should not be treated as error.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190830103841.25128-5-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Most of the daVinci devices does not boot with DT. In this case the DMA
channel is looked up with dma_slave_map and for that the chan_names[]
must be configured.
Both McASP and ASP/McBSP uses "tx" and "rx" as channel names, so we can
just do this when the dev->of_node is not valid.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190830103841.25128-4-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
ASP/McBSP can support 8/16/20/24/32 bits word in theory. I have only tested
S16_LE and S32_LE, the other formats might not work so only extend the
supported formats with S32_LE for now.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190830103841.25128-2-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently the driver uses snd_soc_rtdcom_lookup() in it's mcbsp_start
function to try to stop/restart the DMA as the initial XSYNCERR workaround
need to be done before the DMA is armed.
There are couple of things wrong with this:
- the driver crashes with NULL pointer dereference as the
component->driver->ops is actually NULL
- the driver should not use snd_soc_rtdcom_lookup() in the first place
- Fiddling with DMA is never a good thing
Move the workaround handling to .prepare which is called before the DMA is
armed, so it complies with the requirements.
Reported-by (usage of snd_soc_rtdcom_lookup): Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190830103841.25128-3-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
We can use snd_soc_dai_link_component to specify aux_dev.
Let's use it.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/87imr86w96.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
A relatively large batch of mostly unremarkable fixes here, a couple of
small core fixes for fairly obscure issues, more comment/email updates
with no code impact than usual and a bunch of small driver fixes.
The support for new sample rates in the max98373 driver is a fix for the
fact that the driver declared support for those rates but would in fact
return an error if these rates were selected.
-----BEGIN PGP SIGNATURE-----
iQFHBAABCgAxFiEEreZoqmdXGLWf4p/qJNaLcl1Uh9AFAl1IKi0THGJyb29uaWVA
a2VybmVsLm9yZwAKCRAk1otyXVSH0NWqB/958QE+MLcaVNkjRAYSCGb56vzEUAlz
jXLuOvdlXWzEmuc6nV32a70KcFngXEvXHn9U+mB1pz45q4jacsghafFeWiL1Dq2E
UzLy2/FtZQtmAVMlxSwcdP1uJzAyfDchcyGJEO8QOIIXNDMSqEINdzGkfqCp8MDM
mHIz8pIW1FlAnZgXp+AWqgn0xrueUzjEiKr4IBp1BoaqOc4rXke823o/SekdNbyC
FkeMcb9LdTj22+DG8Z1DqOqjYHHx7/CwOcALjQlU2pK+3uGtOENXMYuwCcBKOIMc
DAbIlhXk+51fZ0r31vhjMrN7NkLqD/cNzn8QEzBJ7F3YckIDbbuu+uYH
=ENjL
-----END PGP SIGNATURE-----
Merge tag 'asoc-fix-v5.3-rc3' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.3
A relatively large batch of mostly unremarkable fixes here, a couple of
small core fixes for fairly obscure issues, more comment/email updates
with no code impact than usual and a bunch of small driver fixes.
The support for new sample rates in the max98373 driver is a fix for the
fact that the driver declared support for those rates but would in fact
return an error if these rates were selected.
Mark switch cases where we are expecting to fall through.
This patch fixes the following warning (Building: arm):
sound/soc/ti/n810.c: In function ‘n810_ext_control’:
sound/soc/ti/n810.c:48:10: warning: this statement may fall through [-Wimplicit-fallthrough=]
line1l = 1;
~~~~~~~^~~
sound/soc/ti/n810.c:49:2: note: here
case N810_JACK_HP:
^~~~
sound/soc/ti/rx51.c: In function ‘rx51_ext_control’:
sound/soc/ti/rx51.c:57:6: warning: this statement may fall through [-Wimplicit-fallthrough=]
hs = 1;
~~~^~~
sound/soc/ti/rx51.c:58:2: note: here
case RX51_JACK_HP:
^~~~
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Link: https://lore.kernel.org/r/20190729221534.GA18696@embeddedor
Signed-off-by: Mark Brown <broonie@kernel.org>
Implement custom snd_pcm_hw_rule to filter the available formats for the
second stream to make it symmetric and allow only formats which require
the same amount of bits on the bus as the running stream.
A simple constraint is not working correctly because for example:
the first stream is started with S24_LE
If we place 24 as constraint for the SAMPLE_BITS then the second stream
can not use S24_LE as it is physically 32bits.
If we would place 32 as constraint (physical width) then S32_LE would have
been allowed, but S24_3LE is not.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190726064244.3762-3-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The slot_width is a property for the bus while the constraint for
SNDRV_PCM_HW_PARAM_SAMPLE_BITS is for the in memory format.
Applying slot_width constraint to sample_bits works most of the time, but
it will blacklist valid formats in some cases.
With slot_width 24 we can support S24_3LE and S24_LE formats as they both
look the same on the bus, but a a 24 constraint on sample_bits would not
allow S24_LE as it is stored in 32bits in memory.
Implement a simple hw_rule function to allow all formats which require less
or equal number of bits on the bus as slot_width (if configured).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190726064244.3762-2-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When running McASP as master capture alone will not record any audio unless
a parallel playback stream is running. As soon as the playback stops the
captured data is going to be silent again.
In McASP master mode we need to set the PDIR for the clock pins and fix
the mcasp_set_axr_pdir() to skip the bits in the PDIR registers above
AMUTE.
This went unnoticed as most of the boards uses McASP as slave and neither
of these issues are visible (audible) in those setups.
Fixes: ca3d943334 ("ASoC: davinci-mcasp: Update PDIR (pin direction) register handling")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190725083423.7321-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
When multiple serializers are used we need to track the number of
serializers used by the other stream direction to avoid killing data lines
when the first stream used more serializers than the second would need.
We are still protected against the case when the second stream uses more
serializers which had affected the running stream as well.
To take advantage of the improved serializer logic we need to modify the
channel constraints rule as well to allow the use of multiple serializers
for the second stream as additional ones will not affect the FS/BCLK on
the bus.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190725083432.7419-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
The formater unit's rotation needs to be programmed differently for right
aligned bus format to have the data moved to the correct place.
Take the opportunity and simplify the formater unit setup code.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/20190725083411.7211-1-peter.ujfalusi@ti.com
Signed-off-by: Mark Brown <broonie@kernel.org>
