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Lenovo ThinkCentre M73 and M93 don't seem to have a proper beep
although the driver tries to probe and set up blindly.
Blacklist these machines for suppressing the beep creation.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204635
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some Realtek codec parsers didn't check the availability of PC beep.
Add has_cdefine_beep() check appropriately.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We have another Dell laptop which needs the DELL4_MIC_NO_PRESENCE,
and this laptop has different pincfg definitions from existing
ones in the pintbl, rather adding a new entry, let us define
a tbl in the fallback_pin_fixup_tbl and this tbl will match
all dell machines with alc289 codec and the pins of 0x19 and 0x1b
are undef by default.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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With the existing pintbl, we already have many entries in it. it is
better to figure out a new way to reduce the size of the pintbl.
We plan to define a new tbl which will match more machines with a
single tbl, To do that, this function doesn't need to match all valid
pins between machine and tbl, it just needs to match all pins defined
in the tbl with the machine.
And the plan is to move some tbls from pin_fixup_tbl to
fallback_pin_fixup_tbl gradually.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This reverts commit ee5f85d9290f ("ALSA: hda: Add codec on bus address
table lately"). The commit caused several regression since I've
overlooked that the function doesn't manage only the caddr_tbl but
also the codec linked list that is referred indirectly in the other
drivers.
Revert for now to make everything back to work.
Fixes: ee5f85d9290f ("ALSA: hda: Add codec on bus address table lately")
Reported-by: Chris Wilson <chris@chris-wilson.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Drop EXPORT_SYMBOL*() from a few more stuff in HD-audio core that
aren't used outside. Particular the unsol event handler can be
staticized now because the recent change removed all external
callers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_hdac_bus_add_device() and snd_hdac_remove_device() are called only
internally in hda-core. Let's drop the exports of them and move the
declarations into local.h.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The call of snd_hdac_bus_add_device() is needed only for registering
the codec onto the bus caddr_tbl[] that is referred essentially only
in the unsol event handler. That is, the reason of this call and the
release by the counter-part function snd_hdac_bus_remove_device() is
just to assure that the unsol event gets notified to the codec.
But the current implementation of the unsol notification wouldn't work
properly when the codec is still in a premature init state. So this
patch tries to work around it by delaying the caddr_tbl[] registration
at the point of snd_hdac_device_register().
Also, the order of snd_hdac_bus_remove_device() and device_del() calls
are shuffled to make sure that the unsol event is masked before
deleting the device.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204565
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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We don't need to deal with the unsol events for Intel chips that are
tied with the graphics via audio component notifier. Although the
presence of the audio component is checked at the beginning of
hdmi_unsol_event(), better to short cut by dropping unsol_event ops.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204565
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Variable result is initialized to a value that is never read and it is
re-assigned later. The initialization is redundant and can be removed.
Addresses-Coverity: ("Unused value")
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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sound/pci/hda/patch_sigmatel.c:981:30: warning:
stac9200_core_init defined but not used [-Wunused-const-variable=]
It is never used, so can be removed.
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Currently we set hdac_stream.fifo_size field only for the playback
stream by some odd reason I forgot, while this field isn't referred in
any places. Actually this fifo_size field would have been required in
the position report correction for VIA chipset, but due to the lack of
the fifo_size set for capture streams, snd-hda-intel driver fetches
the register by itself.
This patch straightens and simplifies the code by setting the
fifo_size field for both playback and capture streams, and use it in
the HD-audio controller driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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[ This is rather a revival of the patch Tomas sent in months ago, but
applying only with the quirk model option -- tiwai ]
Hard coded coefficients to make Huawuei Matebook X right speaker
work. The Matebook X has a ALC298, please refer to bug 197801 on
how these numbers were reverse engineered from the Windows driver
The reversed engineered sequence represents a repeating pattern
of verbs, and the only values that are changing periodically are
written on indexes 0x23 and 0x25:
0x500, 0x23
0x400, VALUE1
0x500, 0x25
0x400, VALUE2
* skipped reading sequences (0x500 - 0xc00 sequences are ignored)
* static values from reverse engineering are used
NOTE: since a significant risk is still considered, this is provided
as an experimental fix that isn't applied as default for now. For
enabling the fix, you'll have to choose huawei-mbx-stereo via model
option of snd-hda-intel module.
If we get feedback from users that this works stably, we may apply it
per default.
[ Some coding style fixes and replacement with AC_VERB_* by tiwai ]
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=197801
Signed-off-by: Tomas Espeleta <tomas.espeleta@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For the sake of debugging, show each response that took too long time
and its corresponding last verb. Since such an error happens in
series once when started, use the ratelimited variant for suppressing
the flood.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Pull yet another compile error fix in the previous HD-audio patchset.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Building with SND_SOC_SOF_HDA_AUDIO_CODEC fails:
sound/soc/sof/intel/hda-bus.c: In function sof_hda_bus_init:
sound/soc/sof/intel/hda-bus.c:16:25: error: implicit declaration of function
snd_soc_hdac_hda_get_ops; did you mean snd_soc_jack_add_gpiods? [-Werror=implicit-function-declaration]
#define sof_hda_ext_ops snd_soc_hdac_hda_get_ops()
Reported-by: Hulk Robot <hulkci@huawei.com>
Suggested-by: Takashi Iwai <tiwai@suse.de>
Fixes: d4ff1b3917a5 ("ASoC: SOF: Intel: Initialize hdaudio bus properly")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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sound/pci/ac97/ac97_codec.c:599:38: warning:
snd_ac97_controls_master_mono defined but not used [-Wunused-const-variable=]
It is never used, so can be removed.
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Pull the compile fix on arm due to missing include.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fixes: 19abfefd4c76 ("ALSA: hda: Direct MMIO accesses")
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Pull HD-audio bus ops cleanups.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The SOF HD-audio bus has its house-made initialization code. It's
supposedly for making the code independent from HD-audio bus drivers.
However, this is error-prone, and above all, the SOF driver has
already dependency on HD-audio bus driver when CONFIG_SND_SOF_HDA is
set. That is, if this Kconfig is set, there is no reason to avoid the
call to the proper bus init function.
Also, the ext_ops that is set at bus initialization can be better
handled inside sof_hda_bus_init(). We don't need to refer this
outside the bus initialization.
So this patch addresses these issues:
- sof_hda_bus_init() calls nothing but snd_hdac_ext_bus_init()
when CONFIG_SND_SOF_HDA is set. Otherwise some fields are
initialized locally like before for avoiding the dependency.
- ext_ops is referred inside sof_hda_bus_init(). The ext_ops argument
of snd_hda_bus_init() is dropped.
Acked-by: Mark Brown <broonie@kernel.org>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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HD-audio drivers access to the mmio registers indirectly via the
corresponding bus->io_ops callbacks. This is because some platform
(notably Tegra SoC) requires the word-aligned access. But it's rather
a rare case, and other platforms suffer from the penalties by indirect
calls unnecessarily.
This patch is an attempt to optimize and cleanup for this situation.
Now the special aligned access is used only when a new kconfig
CONFIG_SND_HDA_ALIGNED_MMIO is set. And the HD-audio core itself
provides the aligned MMIO access helpers instead of the driver side.
If Kconfig isn't set (as default), the standard helpers like readl()
or writel() are used directly.
A couple of places in ASoC Intel drivers have the access via io_ops
reg_writel(), and they are replaced with the direct writel() calls.
And now with this patch, the whole bus->io_ops becomes empty, so it's
dropped completely. The bus initialization functions are changed
accordingly as well to drop the whole bus->io_ops.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The HD-audio core allocates and releases pages via driver's specific
dma_alloc_pages and dma_free_pages ops defined in bus->io_ops. This
was because some platforms require the uncached pages and the handling
of page flags had to be done locally in the driver code.
Since the recent change in ALSA core memory allocator, we can simply
pass SNDRV_DMA_TYPE_DEV_UC for the uncached pages, and the only
difference became about this type to be passed to the core allocator.
That is, it's good time for cleaning up the mess.
This patch changes the allocation code in HD-audio core to call the
core allocator directly so that we get rid of dma_alloc_pages and
dma_free_pages io_ops. If a driver needs the uncached pages, it has
to set bus->dma_type right after the bus initialization.
This is merely a code refactoring and shouldn't bring any behavior
changes.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Pioneer DDJ-SX3 is a plain 12 32bit channel out and 10 channel in
PCM/midi controller. The PCM part is "vendor specific".
It needs the "ignore invalid bsynchaddress" patch as it uses 0 for that.
Signed-off-by: Ard van Breemen <ard@kwaak.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Linux kernel assumes that get_endpoint(alts,0) and
get_endpoint(alts,1) are eachothers feedback endpoints.
To reassure that validity it will test bsynchaddress to comply with that
assumption. But if the bsyncaddress is 0 (invalid), it will flag that as
a wrong assumption and return an error.
Fix: Skip the test if bSynchAddress is 0.
Note: those with a valid bSynchAddress should have a code quirck added.
Signed-off-by: Ard van Breemen <ard@kwaak.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some cards have alternate setting with non-PCM format as the first
altsetting in the interface descriptors. This confuses userspace, since
alsa-lib uses device 0 by default. So lets parse interfaces in two steps:
1. Parse altsettings with PCM formats.
2. Parse altsettings with non-PCM formats.
This fixes at least following cards:
- Audinst HUD-mx2
- Audinst HUD-mini
[ Adapted to 5.3 kernel by tiwai ]
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There are many open code for releasing audioformat object.
Provide a unified helper and call it from the all places.
Only a cleanup, no functional changes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Pull 5.3 development branch for further fixes of USB-audio stuff.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In snd_usb_get_audioformat_uac3(), a structure for channel maps 'chmap' is
allocated through kzalloc() before the execution goto 'found_clock'.
However, this structure is not deallocated if the memory allocation for
'pd' fails, leading to a memory leak bug.
To fix the above issue, free 'fp->chmap' before returning NULL.
Fixes: 7edf3b5e6a45 ("ALSA: usb-audio: AudioStreaming Power Domain parsing")
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.3
Incremental fix removing executable bits added in a prior patch
accidentally.
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Signed-off-by: Mark Brown <broonie@kernel.org>
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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.
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We shouldn't assume CPU physical address we get from page_to_phys()
is same as DMA address we get from dma_alloc_coherent(). On x86_64,
we won't run into any problem with the assumption when dma_ops is
nommu_dma_ops. However, DMA address is IOVA when IOMMU is enabled.
And it's most likely different from CPU physical address when AMD
IOMMU is not in passthrough mode.
This patch fixes page faults when IOMMU is enabled.
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Link: https://lore.kernel.org/r/1564753899-17124-2-git-send-email-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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AMD platform device acp3x_rv_i2s created by parent PCI device
driver. Pass struct device of the parent to
snd_pcm_lib_preallocate_pages() so dma_alloc_coherent() can use
correct dma_ops. Otherwise, it will use default dma_ops which
is nommu_dma_ops on x86_64 even when IOMMU is enabled and
set to non passthrough mode.
Signed-off-by: Vijendar Mukunda <vijendar.mukunda@amd.com>
Link: https://lore.kernel.org/r/1564753899-17124-1-git-send-email-Vijendar.Mukunda@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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88200 and 96000 sampling rate was not enabled on driver, so can't be played.
The error information:
max98373 3-0031：rate 96000 not supported
max98373 3-0031：ASoC: can't set max98373-aif1 hw params: -22
Signed-off-by: fengchunguo <chunguo.feng@amlogic.com>
Link: https://lore.kernel.org/r/20190731074156.5620-1-chunguo.feng@amlogic.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The A64 audio codec uses the original I2S block but the SR and
WSS computation currently assigned is for the newer block.
Fixes: 619c15f7fac9 (ASoC: sun4i-i2s: Change SR and WSS computation)
Signed-off-by: Marcus Cooper <codekipper@gmail.com>
Link: https://lore.kernel.org/r/20190729152130.27955-1-codekipper@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Adding myself to Intel ASoC drivers maintainers list.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190726181517.27655-1-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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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>
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This reverts commit db51707b9c9aeedd310ebce60f15d5bb006567e0.
Revert "ASoC: rockchip: i2s: Support mono capture"
Previous discussion in
https://patchwork.kernel.org/patch/10147153/
explains the issue of the patch.
While device is configured as 1-ch, hardware is still
generating a 2-ch stream.
When user space reads the data and assumes it is a 1-ch stream,
the rate will be slower by 2x.
Revert the change so 1-ch is not supported.
User space can selectively take one channel data out of two channel
if 1-ch is preferred.
Currently, both channels record identical data.
Signed-off-by: Cheng-Yi Chiang <cychiang@chromium.org>
Link: https://lore.kernel.org/r/20190726044202.26866-1-cychiang@chromium.org
Signed-off-by: Mark Brown <broonie@kernel.org>
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Fix some typo to have the filaname given in a comment match the real name
of the file.
Some 'acpi' have erroneously been written 'apci'
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Link: https://lore.kernel.org/r/20190725053523.16542-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
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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: ca3d9433349e ("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>
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If the DAI format setup fails, there is no valid communication format
between CPU and CODEC, so fail card instantiation, rather than continue
with a card that will most likely not function properly.
Signed-off-by: Ricard Wanderlof <ricardw@axis.com>
Link: https://lore.kernel.org/r/alpine.DEB.2.20.1907241132350.6338@lnxricardw1.se.axis.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Downgrade "nothing to do in IRQ thread" message from error to a debug
message in the IPC interrupt handler thread.
The spurious wake-up can happen if a HDA stream interrupt is
raised while the IPC interrupt thread is running. IPC functionality
is not impacted by this condition, so debug is a more appropriate
trace level.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20190722141402.7194-21-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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apq8016_sbc_parse_of() sets up multiple DAI links, depending on the
number of nodes in the device tree. However, at the moment
CPU and platform components are only allocated for the first link.
This causes an oops when more than one link is defined:
Internal error: Oops: 96000044 [#1] SMP
CPU: 0 PID: 1015 Comm: kworker/0:2 Not tainted 5.3.0-rc1 #4
Call trace:
apq8016_sbc_platform_probe+0x1a8/0x3f0
platform_drv_probe+0x50/0xa0
...
Move the allocation inside the loop to ensure that each link is
properly initialized.
Fixes: 98b232ca9e0e ("ASoC: qcom: apq8016_sbc: use modern dai_link style")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20190722130352.95874-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
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In snd_soc_dapm_new_control_unlocked(), a kernel buffer is allocated in
dapm_cnew_widget() to hold the new dapm widget. Then, different actions are
taken according to the id of the widget, i.e., 'w->id'. If any failure
occurs during this process, snd_soc_dapm_new_control_unlocked() should be
terminated by going to the 'request_failed' label. However, the allocated
kernel buffer is not freed on this code path, leading to a memory leak bug.
To fix the above issue, free the buffer before returning from
snd_soc_dapm_new_control_unlocked() through the 'request_failed' label.
Signed-off-by: Wenwen Wang <wenwen@cs.uga.edu>
Link: https://lore.kernel.org/r/1563803864-2809-1-git-send-email-wang6495@umn.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
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When CONFIG_UAPI_HEADER_TEST=y, exported headers are compile-tested to
make sure they can be included from user-space.
Currently, header.h and fw.h are excluded from the test coverage.
To make them join the compile-test, we need to fix the build errors
attached below.
For a case like this, we decided to use __u{8,16,32,64} variable types
in this discussion:
https://lkml.org/lkml/2019/6/5/18
Build log:
CC usr/include/sound/sof/header.h.s
CC usr/include/sound/sof/fw.h.s
In file included from <command-line>:32:0:
./usr/include/sound/sof/header.h:19:2: error: unknown type name ‘uint32_t’
uint32_t magic; /**< 'S', 'O', 'F', '\0' */
^~~~~~~~
./usr/include/sound/sof/header.h:20:2: error: unknown type name ‘uint32_t’
uint32_t type; /**< component specific type */
^~~~~~~~
./usr/include/sound/sof/header.h:21:2: error: unknown type name ‘uint32_t’
uint32_t size; /**< size in bytes of data excl. this struct */
^~~~~~~~
./usr/include/sound/sof/header.h:22:2: error: unknown type name ‘uint32_t’
uint32_t abi; /**< SOF ABI version */
^~~~~~~~
./usr/include/sound/sof/header.h:23:2: error: unknown type name ‘uint32_t’
uint32_t reserved[4]; /**< reserved for future use */
^~~~~~~~
./usr/include/sound/sof/header.h:24:2: error: unknown type name ‘uint32_t’
uint32_t data[0]; /**< Component data - opaque to core */
^~~~~~~~
In file included from <command-line>:32:0:
./usr/include/sound/sof/fw.h:49:2: error: unknown type name ‘uint32_t’
uint32_t size; /* bytes minus this header */
^~~~~~~~
./usr/include/sound/sof/fw.h:50:2: error: unknown type name ‘uint32_t’
uint32_t offset; /* offset from base */
^~~~~~~~
./usr/include/sound/sof/fw.h:64:2: error: unknown type name ‘uint32_t’
uint32_t size; /* bytes minus this header */
^~~~~~~~
./usr/include/sound/sof/fw.h:65:2: error: unknown type name ‘uint32_t’
uint32_t num_blocks; /* number of blocks */
^~~~~~~~
./usr/include/sound/sof/fw.h:73:2: error: unknown type name ‘uint32_t’
uint32_t file_size; /* size of file minus this header */
^~~~~~~~
./usr/include/sound/sof/fw.h:74:2: error: unknown type name ‘uint32_t’
uint32_t num_modules; /* number of modules */
^~~~~~~~
./usr/include/sound/sof/fw.h:75:2: error: unknown type name ‘uint32_t’
uint32_t abi; /* version of header format */
^~~~~~~~
Signed-off-by: Masahiro Yamada <yamada.masahiro@socionext.com>
Link: https://lore.kernel.org/r/20190721142308.30306-1-yamada.masahiro@socionext.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The TS3A227E says that the headset keypress detection needs the MICBIAS
power in order to report the key events to ensure proper operation
The headset keypress detection needs the MICBIAS power in order to report
the key events all the time as long as MIC is present. So MICBIAS pin
is forced on when a MICROPHONE is detected.
On Veyron Minnie I observed that if the MICBIAS power is not present and
the key press detection is activated (just because it is enabled when you
insert a headset), it randomly reports a keypress on insert.
E.g. (KEY_PLAYPAUSE)
Event: (SW_HEADPHONE_INSERT), value 1
Event: (SW_MICROPHONE_INSERT), value 1
Event: -------------- SYN_REPORT ------------
Event: (KEY_PLAYPAUSE), value 1
Userspace thinks that KEY_PLAYPAUSE is pressed and produces the annoying
effect that the media player starts a play/pause loop.
Note that, although most of the time the key reported is the one
associated with BTN_0, not always this is true. On my tests I also saw
different keys reported
Signed-off-by: Enric Balletbo i Serra <enric.balletbo@collabora.com>
Link: https://lore.kernel.org/r/20190719173929.24065-1-enric.balletbo@collabora.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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When sample rate of TX is different with sample rate of RX in
async mode, the MFreq selection will be wrong.
For example, sysclk = 24.576MHz, TX rate = 96000Hz, RX rate = 48000Hz.
Then ratio of TX = 256, ratio of RX = 512, For MFreq is shared by TX
and RX instance, the correct value of MFreq is 2 for both TX and RX.
But original method will cause MFreq = 0 for TX, MFreq = 2 for RX.
If TX is started after RX, RX will be impacted, RX work abnormal with
MFreq = 0.
This patch is to select proper MFreq value according to TX rate and
RX rate.
Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver support for CS42448/CS42888")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/20190716094547.46787-1-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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DPCM uses snd_soc_dapm_dai_get_connected_widgets to build a
list of the widgets connected to a specific front end DAI so it
can search through this list for available back end DAIs. The
custom_stop_condition was added to is_connected_ep to facilitate this
list not containing more widgets than is necessary. Doing so both
speeds up the DPCM handling as less widgets need to be searched and
avoids issues with CODEC to CODEC links as these would be confused
with back end DAIs if they appeared in the list of available widgets.
custom_stop_condition was implemented by aborting the graph walk
when the condition is triggered, however there is an issue with this
approach. Whilst walking the graph is_connected_ep should update the
endpoints cache on each widget, if the walk is aborted the number
of attached end points is unknown for that sub-graph. When the stop
condition triggered, the original patch ignored the triggering widget
and returned zero connected end points; a later patch updated this
to set the triggering widget's cache to 1 and return that. Both of
these approaches result in inaccurate values being stored in various
end point caches as the values propagate back through the graph,
which can result in later issues with widgets powering/not powering
unexpectedly.
As the original goal was to reduce the size of the widget list passed
to the DPCM code, the simplest solution is to limit the functionality
of the custom_stop_condition to the widget list. This means the rest
of the graph will still be processed resulting in correct end point
caches, but only widgets up to the stop condition will be added to the
returned widget list.
Fixes: 6742064aef7f ("ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets")
Fixes: 5fdd022c2026 ("ASoC: dpcm: play nice with CODEC<->CODEC links")
Fixes: 09464974eaa8 ("ASoC: dapm: Fix to return correct path list in is_connected_ep.")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20190718084333.15598-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The Irbis NB41 netbook has its internal mic on IN2, inverted jack-detect
and stereo speakers, add a quirk for this.
Cc: russianneuromancer@ya.ru
Reported-and-tested-by: russianneuromancer@ya.ru
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Link: https://lore.kernel.org/r/20190712112708.25327-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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