| Age | Commit message (Collapse) | Author |
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Building kernel with SND_SOC_IMX_AUDMUX=n leads to the following error:
sound/built-in.o: In function `fsl_asoc_card_probe':
>> fsl-asoc-card.c:(.text+0x1467b5): undefined reference to `imx_audmux_v2_configure_port'
>> fsl-asoc-card.c:(.text+0x1467d0): undefined reference to `imx_audmux_v2_configure_port'
>> fsl-asoc-card.c:(.text+0x1467ed): undefined reference to `imx_audmux_v2_configure_port'
>> fsl-asoc-card.c:(.text+0x146807): undefined reference to `imx_audmux_v2_configure_port'
Update Kconfig to select SND_SOC_IMX_AUDMUX when SND_SOC_FSL_ASOC_CARD=y.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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When build fsl-asoc-card as module, there is following error:
sound/soc/fsl/fsl-asoc-card.c: In function 'fsl_asoc_card_probe':
>> sound/soc/fsl/fsl-asoc-card.c:547:13: warning: 'asrc_np' may be used uninitialized in this function [-Wmaybe-uninitialized]
of_node_put(asrc_np);
^
vim +/asrc_np +547 sound/soc/fsl/fsl-asoc-card.c
531 if (width == 24)
532 priv->asrc_format = SNDRV_PCM_FORMAT_S24_LE;
533 else
534 priv->asrc_format = SNDRV_PCM_FORMAT_S16_LE;
535 }
536
537 /* Finish card registering */
538 platform_set_drvdata(pdev, priv);
539 snd_soc_card_set_drvdata(&priv->card, priv);
540
541 ret = devm_snd_soc_register_card(&pdev->dev, &priv->card);
542 if (ret)
543 dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
544
545 fail:
546 of_node_put(codec_np);
> 547 of_node_put(asrc_np);
548 of_node_put(cpu_np);
549
550 return ret;
551 }
552
553 static const struct of_device_id fsl_asoc_card_dt_ids[] = {
554 { .compatible = "fsl,imx-audio-cs42888", },
555 { .compatible = "fsl,imx-audio-sgtl5000", },
Add 'asrc_fail' branch for error jump after asrc_np initialized.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The AB8500_GPIO_DIR4_REG register define has the bank for the register in the
upper 8 bits and the register itself in the lower 8 bits. When passing it to
abx500_{set,get}_register_interruptible() the upper bits get truncated which
generates the following warning from sparse:
sound/soc/codecs/ab8500-codec.c:1972:53: warning: cast truncates bits
from constant value (1013 becomes 13)
sound/soc/codecs/ab8500-codec.c:1980:53: warning: cast truncates bits
from constant value (1013 becomes 13)
The bank is passed separately to abx500_{set,get}_register_interruptible() so
the code works fine as it is. Given that all users of AB8500_GPIO_DIR4_REG
always truncate the upper 8 bits just remove them from the define.
Also remove the unnecessary casts to u8.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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It is not always possible to interchange iomem pointers with normal pointers,
which why we have annotations for iomem pointers and warn when casting them to a
normal pointer or vice versa. In this case the casting is fine and unfortunately
necessary so add the proper annotations to tell code checkers that it is
intentional. This silences the following warnings from sparse:
sound/soc/samsung/idma.c:354:20: warning: incorrect type in argument 1
(different address spaces) expected void volatile [noderef]
<asn:2>*addr got unsigned char *area
sound/soc/samsung/idma.c:372:22: warning: cast removes address space of
expression
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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dmaengine_prep_slave_single() expects a enum dma_transfer_direction and not a
enum dma_data_direction. Since the integer representations of both DMA_TO_DEVICE
and DMA_MEM_TO_DEV aswell as DMA_FROM_DEVICE and DMA_DEV_TO_MEM have the same
value the code worked fine even though it was using the wrong type.
Fixes the following warnings from sparse:
sound/soc/sh/fsi.c:1307:42: warning: mixing different enum types
sound/soc/sh/fsi.c:1307:42: int enum dma_data_direction versus
sound/soc/sh/fsi.c:1307:42: int enum dma_transfer_direction
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Sparse spits out the following warning:
sound/soc/sh/rcar/gen.c:250:21: warning: dubious: x & !y
It does this because sometimes mixing boolean and bit-wise logic has not the
intended result. In this case we are fine, but replacing the bit-wise '&' with
the boolean '&&' silences the sparse warning. The generated code for both cases
is the same.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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odroidx2_drvdata and odroidu3_drvdata are not used outside this module so make
them static (and also const while we are at it).
Fixes the following warnings from sparse:
sound/soc/samsung/odroidx2_max98090.c:69:26: warning: symbol
'odroidx2_drvdata' was not declared. Should it be static?
sound/soc/samsung/odroidx2_max98090.c:74:26: warning: symbol
'odroidu3_drvdata' was not declared. Should it be static?
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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edma_pcm_platform_register() is declared in edma-pcm.h and defined in
edma-pcm.c. To make sure that the function signature matches for both
edma-pcm.c should include edma-pcm.h
Fixes the following sparse warning:
sound/soc/davinci/edma-pcm.c:48:5: warning: symbol
'edma_pcm_platform_register' was not declared. Should it be static?
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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init_name is basically a hack and should only be used for statically allocated
device structs. For dynamically allocated devices dev_set_name() should be used.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Improve build coverage of the dmic driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
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There is one design rule according to SAI's reference manual:
If the transmitter bit clock and frame sync are to be used by both transmitter
and receiver, the transmitter must be configured for asynchronous operation
and the receiver for synchronous operation.
And SYNC of TCR2 is a 2-width control bit:
00 Asynchronous mode.
01 Synchronous with receiver.
10 Synchronous with another SAI transmitter.
11 Synchronous with another SAI receiver.
So the driver should have set SYNC bit of TCR2 to 0x0, and meanwhile set SYNC
bit of RCR2 to 0x1 (Synchronous with transmitter).
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-fsl-esai
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This adds an initial machine driver for the ES8328 audio codec on Freescale
boards. The driver supports headphones and an audio regulator for an onboard
speaker amp.
Signed-off-by: Sean Cross <xobs@kosagi.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-fsl
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Add a codec driver for the Everest ES8328. It supports two separate audio
outputs and two separate audio inputs.
Signed-off-by: Sean Cross <xobs@kosagi.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Current code only allows direct routing of the WM8994 microphone
detection signal to a GPIO this change adds support to demux the
interrupt from the main interrupt line of the codec.
Signed-off-by: Nikesh Oswal <nikesh@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Add DAPM calls to enable/disable the Class D amp.
Also add a DAPM call to turn off the PLL upon
the stream completing.
Signed-off-by: Dan Murphy <dmurphy@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Signed-off-by: Rongjun Ying <rongjun.ying@csr.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Of_node_put supports NULL as its argument, so the initial test is not
necessary.
Suggested by Uwe Kleine-König.
The semantic patch that fixes this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression e;
@@
-if (e)
of_node_put(e);
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
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We define the DSP commands,structures here which will be used to send the IPCs
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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sst_ops need to use the sst driver context. So pass sst device as argument,
which can be used to retrieve sst context.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Using a byte control interface instead of generic_params ioctl.
Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The previous patch (ASoC: fsl_sai: Add asynchronous mode support) added
new Device Tree bindings for Asynchronous and Synchronous modes support.
However, these two shall not be present at the same time.
So this patch just simply makes them exclusive so as to avoid incorrect
Device Tree binding usage.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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SAI supports these operation modes:
1) asynchronous mode
Both Tx and Rx are set to be asynchronous.
2) synchronous mode (Rx sync with Tx)
Tx is set to be asynchronous, Rx is set to be synchronous.
3) synchronous mode (Tx sync with Rx)
Rx is set to be asynchronous, Tx is set to be synchronous.
4) synchronous mode (Tx/Rx sync with another SAI's Tx)
5) synchronous mode (Tx/Rx sync with another SAI's Rx)
* 4) and 5) are beyond this patch because they are related with another SAI.
As the initial version of this SAI driver, it supported 2) as default while
the others were totally missing.
So this patch just adds supports for 1) and 3).
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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There is one design rule according to SAI's reference manual:
If the transmitter bit clock and frame sync are to be used by both transmitter
and receiver, the transmitter must be configured for asynchronous operation
and the receiver for synchronous operation.
And SYNC of TCR2 is a 2-width control bit:
00 Asynchronous mode.
01 Synchronous with receiver.
10 Synchronous with another SAI transmitter.
11 Synchronous with another SAI receiver.
So the driver should have set SYNC bit of TCR2 to 0x0, and meanwhile set SYNC
bit of RCR2 to 0x1 (Synchronous with transmitter).
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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This patch adds software reset code in dai_probe() so as to make a true init
by clearing SAI's internal logic, including the bit clock generation, status
flags, and FIFO pointers.
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Original driver didn't store the number of slots, just fix the slot number
to 2, use this default number to calculate bclk and pins for TX/RX.
In this patch, add one parameter for slots, and update the calculation of
bclk and pins of TX/RX. Then driver will be compatible with slots > 2 in
TDM mode.
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The Freescale Generic ASoC Sound Card is a general ASoC DAI Link driver that
can be used, ideally, for all Freescale CPU DAI drivers and external CODECs.
The idea of this generic sound card is a bit like ASoC Simple Card. However,
for Freescale SoCs (especially those released in recent years), most of them
have ASRC (Documentation/devicetree/bindings/sound/fsl,asrc.txt) inside. And
this is a specific feature that might be painstakingly controlled and merged
into the Simple Card driver.
So having this driver will allow all Freescale SoC users to benefit from the
simplification to support a new card and the capability of wide sample rates
support through ASRC.
The driver is initially designed for sound card using I2S or PCM DAI formats.
However, it's also possible to merge those non-I2S/PCM type sound cards, such
as S/PDIF audio and HDMI audio, into this card as long as the merge will not
break the original function and as long as there is something redundant that
can be abstracted along with I2S type sound cards.
As an initial version, it only supports three cards that I can test:
imx-audio-cs42888, a new card that links ESAI with CS42888 CODEC
imx-audio-sgtl5000, just like the old imx-sgtl5000.c driver
imx-audio-wm8962, just like the old imx-wm8962.c driver
The driver is also compatible with the old Device Tree bindings of WM8962 and
SGTL5000. So we may consider to remove those two drivers after this driver is
totally enabled. (It needs to be added into defconfig)
Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Forgot to add a return for err_disable goto statement.
Causes compile warning of control reaching end of non-void
Signed-off-by: Brian Austin <briann.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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This patch adds support for the Cirrus Logic CS35L32 Boosted Amplifier
I2S output provides monitor data to the SOC/CODEC/DSP for speaker protection/enhancement algorithms
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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ALC269 & co have many vendor-specific setups with COEF verbs.
However, some verbs seem specific to some codec versions and they
result in the codec stalling. Typically, such a case can be avoided
by checking the return value from reading a COEF. If the return value
is -1, it implies that the COEF is invalid, thus it shouldn't be
written.
This patch adds the invalid COEF checks in appropriate places
accessing ALC269 and its variants. The patch actually fixes the
resume problem on Acer AO725 laptop.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=52181
Tested-by: Francesco Muzio <muziofg@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.17
Nothing too exciting here, a bunch of driver fixes that came along since
the initial pull request but none that really stand our and a warning
fix in the core.
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here is the additional fix patches that have been queued up since the
previous pull request. A few HD-audio fixes, a USB-audio quirk
addition, and a couple of trivial cleanup for the legacy OSS codes"
* tag 'sound-fix-3.17-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda - Set TLV_DB_SCALE_MUTE bit for cx5051 vmaster
ALSA: hda/ca0132 - Don't try loading firmware at resume when already failed
ALSA: hda - Fix pop noises on reboot for Dell XPS 13 9333
ALSA: hda - Set internal mic as default input source on Dell XPS 13 9333
ALSA: usb-audio: fix BOSS ME-25 MIDI regression
ALSA: hda - Fix parsing of CMI8888 codec
ALSA: hda - Fix probing and stuttering on CMI8888 HD-audio controller
ALSA: hda/realtek - Fixed ALC286/ALC288 recording delay for Headset Mic
sound: oss: Remove typedefs wanc_info and wavnc_port_info
sound: oss: uart401: Remove typedef uart401_devc
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'asoc/fix/fsl-esai', 'asoc/fix/intel', 'asoc/fix/mcasp' and 'asoc/fix/pxa' into asoc-linus
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git://git.kernel.org/pub/scm/linux/kernel/git/helgaas/pci
Pull DEFINE_PCI_DEVICE_TABLE removal from Bjorn Helgaas:
"Part two of the PCI changes for v3.17:
- Remove DEFINE_PCI_DEVICE_TABLE macro use (Benoit Taine)
It's a mechanical change that removes uses of the
DEFINE_PCI_DEVICE_TABLE macro. I waited until later in the merge
window to reduce conflicts, but it's possible you'll still see a few"
* tag 'pci-v3.17-changes-2' of git://git.kernel.org/pub/scm/linux/kernel/git/helgaas/pci:
PCI: Remove DEFINE_PCI_DEVICE_TABLE macro use
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Pull device tree updates from Grant Likely:
"The branch contains the following device tree changes the v3.17 merge
window:
Group changes to the device tree. In preparation for adding device
tree overlay support, OF_DYNAMIC is reworked so that a set of device
tree changes can be prepared and applied to the tree all at once.
OF_RECONFIG notifiers see the most significant change here so that
users always get a consistent view of the tree. Notifiers generation
is moved from before a change to after it, and notifiers for a group
of changes are emitted after the entire block of changes have been
applied
Automatic console selection from DT. Console drivers can now use
of_console_check() to see if the device node is specified as a console
device. If so then it gets added as a preferred console. UART
devices get this support automatically when uart_add_one_port() is
called.
DT unit tests no longer depend on pre-loaded data in the device tree.
Data is loaded dynamically at the start of unit tests, and then
unloaded again when the tests have completed.
Also contains a few bugfixes for reserved regions and early memory
setup"
* tag 'devicetree-for-linus' of git://git.secretlab.ca/git/linux: (21 commits)
of: Fixing OF Selftest build error
drivers: of: add automated assignment of reserved regions to client devices
of: Use proper types for checking memory overflow
of: typo fix in __of_prop_dup()
Adding selftest testdata dynamically into live tree
of: Add todo tasklist for Devicetree
of: Transactional DT support.
of: Reorder device tree changes and notifiers
of: Move dynamic node fixups out of powerpc and into common code
of: Make sure attached nodes don't carry along extra children
of: Make devicetree sysfs update functions consistent.
of: Create unlocked versions of node and property add/remove functions
OF: Utility helper functions for dynamic nodes
of: Move CONFIG_OF_DYNAMIC code into a separate file
of: rename of_aliases_mutex to just of_mutex
of/platform: Fix of_platform_device_destroy iteration of devices
of: Migrate of_find_node_by_name() users to for_each_node_by_name()
tty: Update hypervisor tty drivers to use core stdout parsing code.
arm/versatile: Add the uart as the stdout device.
of: Enable console on serial ports specified by /chosen/stdout-path
...
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Conexnat HD-audio driver has a workaround for cx5051 (aka CX20561)
chip to add fake mute controls to each amp (commit 3868137e). This
implies the minimum-as-mute TLV bit in TLV for each corresponding
control. Meanwhile we build the virtual master from these, but the
TLV bit is missing, even though the slaves have it.
This patch simply adds the missing TLV_DB_SCALE_MUTE bit for vmaster,
as already done in patch_sigmatel.c.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This mode is unsupported, as the DMA controller can't do zero-padding
of samples.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Johannes Stezenbach <js@sig21.net>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
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Original driver didn't store the number of slots, just fix the slot number
to 2, use this default number to calculate bclk and pins for TX/RX.
In this patch, add one parameter for slots, and update the calculation of
bclk and pins of TX/RX. Then driver will be compatible with slots > 2 in
TDM mode.
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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This reverts commit a603c8ee526f5ea9ad9b40710308766299ad8a69.
fsl_asoc_xlate_tdm_slot_mask() is different with snd_soc_xlate_tdm_slot_mask().
fsl_asoc_xlate_tdm_slot_mask() will set the enabled bit to 0, disabled bit
to 1. snd_soc_xlate_tdm_slot_mask() will set the enabled bit to 1, disabled
bit to 0.
For esai when the bit value is 1, the slot is enabled, when the bit value is 0,
the slot is disabled. If using fsl_asoc_xlate_tdm_slot_mask(), the esai will
work abnormally. So revert this patch, make the esai use default function.
Signed-off-by: Shengjiu Wang <shengjiu.wang@freescale.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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The implicit BLCK divider setting was broken by "ASoC: mcasp: don't
override bclk divider if it was provided by the machine"-patch. After
the BCLK divider is implicitly set for the first time the
mcasp->bclk_div gets a non zero value and the implicit setting is
"turned off".
Signed-off-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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TDM slot length was set same as word length, regardless of the value
received in set_tdm_slot. This patch sets the TDM slot length correctly
as received in set_tdm_slot DAI callback
Signed-off-by: Nikesh Oswal <Nikesh.Oswal@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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We should prefer `struct pci_device_id` over `DEFINE_PCI_DEVICE_TABLE` to
meet kernel coding style guidelines. This issue was reported by checkpatch.
A simplified version of the semantic patch that makes this change is as
follows (http://coccinelle.lip6.fr/):
// <smpl>
@@
identifier i;
declarer name DEFINE_PCI_DEVICE_TABLE;
initializer z;
@@
- DEFINE_PCI_DEVICE_TABLE(i)
+ const struct pci_device_id i[]
= z;
// </smpl>
[bhelgaas: add semantic patch]
Signed-off-by: Benoit Taine <benoit.taine@lip6.fr>
Signed-off-by: Bjorn Helgaas <bhelgaas@google.com>
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The source type should come before the direction specifier according to
ControlNames.txt.
Signed-off-by: Mark Brown <broonie@linaro.org>
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If soc_dapm_read() fails, reg_val will be uninitialized, and bogus
values will be written later:
sound/soc/soc-dapm.c: In function 'snd_soc_dapm_get_enum_double':
sound/soc/soc-dapm.c:2862:15: warning: 'reg_val' may be used uninitialized in this function [-Wmaybe-uninitialized]
unsigned int reg_val, val;
^
Return early on error to fix this.
Introduced by commit ce0fc93ae56e2ba50ff8c220d69e4e860e889320 ("ASoC:
Add DAPM support at the component level").
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Signed-off-by: Mark Brown <broonie@linaro.org>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
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There is no need to restore and restart PCM streams in case ADSP didn't
reach reset and power off state during system suspend/resume cycle. In that
case stream is still active but paused and firmware doesn't allow allocating
a new stream before paused stream is freed.
ADSP remains active in case suspend sequence didn't go to suspend_late
stage. This can happen when either suspend sequence is aborted by a wakeup
or by letting only devices suspend by "echo devices >/sys/power/pm_test".
Currently stream restoring fails in these suspend cases. Fix this by adding
a flag that indicates is complete stream reinitialization needed or is it
enough to resume paused stream. Flag is set when we know that ADSP reached
suspend_late.
Initial fix to this issue came from Fang Yang. I modified it a little and
forward ported it to top of two other suspend/resume patches from me.
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Tested-by: Borun Fu <borun.fu@intel.com>
Cc: yang fang <yang.a.fang@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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Remove sst_byt_pcm_dev_resume() and move waiting of firmware boot into
sst_byt_pcm_dev_resume_early(). Now suspend_late and resume_early phases are
in sync with each other so that we know that ADSP was put into reset and was
unpowered after suspend_late and is ready to resume IO after resume_early
during resume stage in sst_byt_pcm_trigger().
Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com>
Tested-by: Borun Fu <borun.fu@intel.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
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