use snd_dmaengine_pcm_prepare_slave_config to set slave config,
and remove the max_burst_size = 4 hard code.
select SND_SOC_GENERIC_DMAENGINE_PCM for mmp-pcm.
Signed-off-by: Qiao Zhou <zhouqiao@marvell.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When writing the patch write to the device asynchronously, allowing better
performance when used with a bus like SPI which supports this by
minimising the need to context switch back to the driver to get the
next bit of data.
Signed-off-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
When writing the patch write to the device asynchronously, allowing better
performance when used with a bus like SPI which supports this by
minimising the need to context switch back to the driver to get the
next bit of data.
Signed-off-by: Mark Brown <broonie@linaro.org>
Tested-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
When writing the patch write to the device asynchronously, allowing better
performance when used with a bus like SPI which supports this by
minimising the need to context switch back to the driver to get the
next bit of data.
Signed-off-by: Mark Brown <broonie@linaro.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Where possible write to the device asynchronously, allowing better
performance when used with a bus like SPI which supports this by
minimising the need to context switch back to the driver to get the
next bit of data.
Signed-off-by: Mark Brown <broonie@linaro.org>
Tested-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Because the "ASoC: dmaengine-pcm: Provide default config" has provided
us one defualt config of DMA. When using this, the config parameter of
devm_snd_dmaengine_pcm_register() will be NULL, so here we need to have
a check before using it.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
devm_request_and_ioremap() has been deprecated in favour of
devm_ioremap_resource(). Fixes the following coccinelle warning:
sound/soc/adi/axi-spdif.c:194:8-32: ERROR: deprecated devm_request_and_ioremap() API used on line 194
Generated by: coccinelle/api/devm_ioremap_resource.cocci
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
devm_request_and_ioremap() has been deprecated in favour of
devm_ioremap_resource(). Fixes the following coccinelle warning:
sound/soc/adi/axi-i2s.c:195:8-32: ERROR: deprecated devm_request_and_ioremap() API used on line 195
Generated by: coccinelle/api/devm_ioremap_resource.cocci
Signed-off-by: Fengguang Wu <fengguang.wu@intel.com>
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Store chip revision in struct sgtl5000_priv when sgtl5000_i2c_probe()
reads it out from register, so that we can use it in
sgtl5000_enable_regulators() with no need to read register again.
Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Signed-off-by: Mark Brown <broonie@linaro.org>
DPCM can dynamically alter the FE to BE PCM links at runtime based
on mixer/mux setting updates. Add soc_dpcm_runtime_update() calling in
get/put function for mixer/mux to support this feature.
Signed-off-by: Nenghua Cao <nhcao@marvell.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Add the missing clk_disable_unprepare() before return from
tegra20_ac97_platform_probe() in the error handling case.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Reviewed-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A few driver and error handling fixes plus a fix to ensure that we
mute streams when we should. The Atmel trigger addition is a fix to
ensure that we do the correct sequence of interactions with the
hardware.
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Merge tag 'asoc-v3.13-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.13
A few driver and error handling fixes plus a fix to ensure that we
mute streams when we should. The Atmel trigger addition is a fix to
ensure that we do the correct sequence of interactions with the
hardware.
On the Dell machines with codec whose Subsystem Id is 0x10280610,
0x10280629 or 0x1028063e, no external microphone can be detected when
plugging a 3-ring headset. If we add "model=dell-headset-multi" for
the snd-hda-intel.ko, the problem will disappear.
The codecs on these machines belong to alc_269 family.
BugLink: https://bugs.launchpad.net/bugs/1260303
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
While enabling these machines, we found we would sometimes lose an
interrupt if we change hardware volume during playback, and that
disabling msi fixed this issue. (Losing the interrupt caused underruns
and crackling audio, as the one second timeout is usually bigger than
the period size.)
The machines were all machines from HP, running AMD Hudson controller,
and Realtek ALC282 codec.
Cc: stable@vger.kernel.org
BugLink: https://bugs.launchpad.net/bugs/1260225
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Since all Exynos platforms have been converted to dmaengine and many of
the older platforms are in the process of conversion they do not need to
use the legacy s3c-dma APIs for DMA but can instead use the standard ASoC
dmaengine helpers. This both allows them to benefit from improvements
implemented in the generic code and supports multiplatform.
This patch includes some fixes from Padma for Exynos SoCs, her testing
was on a slightly earlier version of the patch due to unrelated breakage
preventing testing.
Signed-off-by: Mark Brown <broonie@linaro.org>
Tested By: Padmavathi Venna <padma.v@samsung.com>
In preparation for using the dmaengine helpers in ASoC rather than the
dmaengine wrappers for the Samsung API wrap the configuration of dma_data.
The dmaengine code expects different data to that used by the legacy API.
Signed-off-by: Mark Brown <broonie@linaro.org>
Check the return value of dma_request_slave_channel_reason() to see if
deferred probe happens, not the variable the return value will be
assigned to later.
Reported-by: kbuild test robot <fengguang.wu@intel.com>
Fixes: 5eda87b890 ("ASoC: dmaengine: support deferred probe for DMA channels")
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Enhance dmaengine_pcm_request_chan_of() to support deferred probe for
DMA channels, by using the new dma_request_slave_channel_or_err() API.
This prevents snd_dmaengine_pcm_register() from succeeding without
acquiring DMA channels due to the relevant DMA controller not yet being
registered.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Remove original filter from fsi_dma_probe(),
and use SH-DMA suitable filter.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
AD1986A codec is a pretty old codec and has really many hidden
restrictions. One of such is that each DAC is dedicated to certain
pin although there are possible connections. Currently, the generic
parser tries to assign individual DACs as much as possible, and this
lead to two bad situations: connections where the sound actually
doesn't work, and connections conflicting other channels.
We may fix this by trying to find the best connections more harder,
but as of now, it's easier to give some hints for paired DAC/pin
connections and honor them if available, since such a hint is needed
only for specific codecs (right now only AD1986A, and there will be
unlikely any others in future).
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66621
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Dell machines with codec whose Subsystem Id is 0x10280624,
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/bugs/1259790
Cc: David Henningsson <david.henningsson@canonical.com>
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In case a single HDA card has both HDMI and S/PDIF outputs, the S/PDIF
outputs will have their IEC958 controls created starting from index 16
and the HDMI controls will be created starting from index 0.
However, HDMI simple_playback_build_controls() as used by old VIA and
NVIDIA codecs incorrectly requests the IEC958 controls to be created
with an S/PDIF type instead of HDMI.
In case the card has other codecs that have HDMI outputs, the controls
will be created with wrong index=16, causing them to e.g. be unreachable
by the ALSA "hdmi" alias.
Fix that by making simple_playback_build_controls() request controls
with HDMI indexes.
Not many cards have an affected configuration, but e.g. ASUS M3N78-VM
contains an integrated NVIDIA HDA "card" with:
- a VIA codec that has, among others, an S/PDIF pin incorrectly
labelled as an HDMI pin, and
- an NVIDIA MCP7x HDMI codec.
Reported-by: MysterX on #openelec
Tested-by: MysterX on #openelec
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Cc: <stable@vger.kernel.org> # 3.8+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Treat both negative and zero return values from clk_round_rate()
as errors. This is needed since subsequent patches will convert
clk_round_rate()'s return value to be an unsigned type, rather
than a signed type, since some clock sources can generate rates higher
than (2^31)-1 Hz.
Eventually, when calling clk_round_rate(), only a return value of
zero will be considered a error; all other values will be
considered valid rates. The comparison against values less than
0 is kept to preserve the correct behavior in the meantime.
Signed-off-by: Paul Walmsley <pwalmsley@nvidia.com>
Acked-by: Hans-Christian Egtvedt <egtvedt@samfundet.no>
Cc: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Treat both negative and zero return values from clk_round_rate()
as errors. This is needed since subsequent patches will convert
clk_round_rate()'s return value to be an unsigned type, rather
than a signed type, since some clock sources can generate rates higher
than (2^31)-1 Hz.
Eventually, when calling clk_round_rate(), only a return value of
zero will be considered a error. All other values will be
considered valid rates. The comparison against values less than
0 is kept to preserve the correct behavior in the meantime.
Signed-off-by: Paul Walmsley <pwalmsley@nvidia.com>
Acked-by: Hans-Christian Egtvedt <egtvedt@samfundet.no>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Not all channels have been initialized, so far, especially when aamix
NID itself doesn't have amps but its leaves have. This patch fixes
these holes. Otherwise you might get unexpected loopback inputs,
e.g. from surround channels.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Optional DT property to specify the desired parent clock for the McASP fck
clock.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
An earlier patch overlooked this when the compatible has been changed from
omap2 -> am33x.
Rename omap2_mcasp_pdata to am33xx_mcasp_pdata.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Instead of passing __iomem address (mcasp->base + register_offset) pass
the main mcasp structure and only access the mcasp->base in the low level
IO functions.
In most cases this helps with code readability and it will make it easier
to switch over to regmap in the future.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The IP in DRA7xx is similar to the IP found in TI81xxAM3xxx/AM4xxx type of
SoCs but it is is integrated with sDMA instead of eDMA. The suitable pcm
driver for DRA7xx is the omap-pcm driver which is using dmaengine.
In the driver we can configure both dma related structures used for eDMA and
sDMA. The only thing we need to make sure that we set the correct dma_data
at startup with snd_soc_dai_set_dma_data()
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In synchronous mode both transmit and receive sections are using the TX
clocks. In setup like this the TX clocks need to be enabled when capture
is running.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The audio data to/from McASP can be sent/received via two method:
Via the data port (preferred) or via the configuration bus.
Currently the driver assumes that all data communication will be done via
the data port.
This patch adds support for selecting the configuration port as data
interface.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The FIFO registers base address is different in dm646x compared to newer
SoCs with McASP IP. Instead of using two paths (switch/case) to handle the
difference we can simply pick the correct base address beforehand and use
offsets to address the register we need to configure.
With this change the indentation depth can be reduced as well.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Replace mcasp->base use with plain base in the davinci_mcasp_set_dai_fmt()
function since it has been already used by the remaining part of the function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Rename the private struct from davinci_audio_dev to davinci_mcasp.
Change the local use of the pointer to this struct from *dev to *mcasp.
The aim is to have better readable code for the first look since having
dev->xxxx in the code when using the local private struct is a bit
surprising.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
It brings no benefit to inline this function due to it's size and function.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since it is a private struct strictly used by the davinci-mcasp driver it
can be moved from header file to the source file.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
It is better for readability to have the register definitions out from the
source file.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
These are not used, probably leftovers from the past.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Specify the dai formats to use within the snd_soc_dai_link structures. In
this way we can remove the code dealing with the dai format configuration
from the machin driver.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
AM43xx have the same McASP IP as AM33xx and both platform uses eDMA. Modify
the Kconfig so it will be possible to add audio support for AM43xx based
boards later.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
We have several boards using the same machine driver for audio support.
All of these machines can select a generic machine driver config option to
build the needed driver while keeping the config options used within the
driver for compile time code path selection.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The help text is misleading and the prompt itself explains the purpose of
this config section.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Gen2 has 0 - 9, total 10 channels, not 9 channels.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
On the Dell Inspiron 3045 machine (codec Subsystem Id: 0x10280628),
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/hwe-somerville/+bug/1259437
CC: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the Dell Optiplex 3030 machine (codec Subsystem Id: 0x10280623),
no external microphone can be detected when plugging a 3-ring
headset. If we add "model=dell-headset-multi" for the
snd-hda-intel.ko, the problem will disappear.
BugLink: https://bugs.launchpad.net/hwe-somerville/+bug/1259435
CC: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add fields to struct snd_dmaengine_pcm_config to allow custom:
- DMA channel names.
This is useful when the default "tx" and "rx" channel names don't
apply, for example if a HW module supports multiple channels, each
having different DMA channel names. This is the case with the FIFOs
in Tegra's AHUB. This new facility can replace
SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME.
- DMA device
This allows requesting DMA channels for a device other than the device
which is registering the "PCM" driver. This is quite unusual, but is
currently useful on Tegra. In much HW, and in Tegra20, each DAI HW
module contains its own FIFOs which DMA writes to. However, in Tegra30,
the DMA FIFOs were split out AHUB HW module, which then routes the data
through a cross-bar, and into the DAI HW modules. However, the current
ASoC driver structure does not expose this detail, and acts as if the
FIFOs are still part of the DAI HW modules. Consequently, the "PCM"
driver is registered with the DAI HW module, yet the DMA channels must
be looked up in the AHUB HW module's device tree node. This new config
field allows that to happen. Eventually, the Tegra drivers will be
reworked to fully expose the AHUB, and this config field can be
removed.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
If snd_dmaengine_pcm_register()'s call to snd_soc_add_platform() fails,
all objects allocated during registration are leaked. Fix this by adding
error-handling code.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Restructure the internals of dmaengine_pcm_request_chan_of() as a loop
over all channels to be allocated. This makes it easier to add logic
that applies to all allocated channels, without having to duplicate that
logic in each of the half-duplex/full-duplex paths.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
if codec driver is used for AIC3X_MODEL_3007 the mono iout controls overwrite
registers for class-d amplifier.
classd amplifier controls are only used for AIC3X_MODEL_3007.
Removing all mono snd_kcontrol_new, snd_soc_dapm_widget, snd_soc_dapm_route
and aic3x_init stuff from common code and call only for not AIC3X_MODEL_3007
codecs.
Testet only with AIC3X_MODEL_3007
Signed-off-by: Jan Weitzel <j.weitzel@phytec.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds a ASoC driver for the AXI-SPDIF softcore. The core implements a
simple SPDIF transmitter and is used on some Analog Devices' reference designs
for various FPGA platforms. For now the driver only support the PL330 as the the
DMA controller.
The driver uses the generic PCM dmaengine driver for its PCM. The only
restriction is that we need to set the SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag as
the dmaengine driver for the DMA core (PL330) that is used with this core has no
residue reporting capabilities yet. This will be fixed in the future though.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds support for the AXI-I2S softcore. The core implements a simple
bidirectional I2S transceiver and is used by Analog Devices in some of their
reference designs for various FPGA platforms.
The driver uses the generic PCM dmaengine driver for its PCM. The only
restriction is that we need to set the SND_DMAENGINE_PCM_FLAG_NO_RESIDUE flag as
the dmaengine driver for the DMA core (PL330) that is used with this core has no
residue reporting capabilities yet. This will be fixed in the future though.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
If we update it here, the set_bias_level() of Codec driver won't be normally
called and we will then miss some essential procedures in set_bias_level() of
the Codec driver. Thus drop it.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
In tegra*_i2s_set_fmt(), in the (fmt == SND_SOC_DAIFMT_CBM_CFM) case,
"val" is never assigned to, but left uninitialized. The other case does
initialized it. Fix this by initializing val at the start of the
function, and only ever ORing into it.
Update the handling of "mask" so it works the same way for consistency.
Update tegra20_spdif.c to use the same code-style for consistency, even
though it doesn't happen to suffer from the same problem at present.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Reviewed-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Fixes: 0f163546a7 ("ASoC: tegra: use regmap more directly")
Cc: <stable@vger.kernel.org>
Since there are more HD-audio compatible codecs, move the definitions
of HD-audio verbs into common header location, include/sound, so that
it can be included cleanly from other drivers than HD-audio driver.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
AD and VIA codecs had stereo mixer input enabled as default before
moving to the generic parser, and people think the lack of such a
regression. In this patch, the stereo mixer input is added back to
the input selection if no auto-mic is available, and if it's not
disabled explicitly via hint. This should satisfy most of demands,
i.e. stereo mix on desktop machines like what it worked before, and it
still keeps the new auto-mic feature on laptops.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sometimes the hardware reports LPIB being advanced than POSBUF.
When this happens, the driver adjusts to a positive value by adding
the buffer size. Then the driver detects it as an error (greater than
the period size), and stops the LPIB delay account from this point
on.
When I took a close look at these conditions, the values shown are all
very small numbers, and it'd be better to just ignore these values
instead of discontinuing the LPIB delay correction.
In this patch, the driver checks a negative delay value and ignores if
it's a significantly small error. Currently the threshold is set to
64 frames, but could be smaller.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The loopback mixing paths aren't initialized correctly at init
callback. Mostly this is harmless as codecs usually set the mute
state as default, but we still should make sure.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We have blindly assumed that all valid configurations should have
either analog or digital playback, but there can be capture-only
configurations. The parser shouldn't escape in such a case.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch skips the default depop delay before D3 for Haswell (10 ms) and
Valleyview2 (100 ms) display codec, to reduce codec suspend time.
The analog part of display audio is implemented in the external display. Some
displays have weak pop noise while others not when suspending, no matter there
is the default delay or not.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I've tested the old Dell Vostro 131 with the latest generic parser
and it works just fine, and as a bonus we get better jack detection
features in userspace. Therefore this quirk can be removed.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the following warning when optimizing for size with gcc-4.6.4:
sound/usb/mixer_quirks.c:1514:6: warning: 'err' may be used uninitialized in this function [-Wuninitialized]
Signed-off-by: Mikulas Patocka <mpatocka@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
DSPCLK_DIV can be only generated correctly after enabling SYSCLK. But if the
current bias_level hasn't reached SND_SOC_BIAS_ON, DAPM won't enable SYSCLK,
which would cause the calculation result from DSPCLK_DIV invalid since bit
DSPCLK_DIV will be finally turned to its true value after DAPM enables SYSCLK
while the driver won't calculate it again for the current instance. In this
circumstance, a playback which needs non-zero DSPCLK_DIV would be distorted
due to unexpected clock frequency resulted from an invalid DSPCLK_DIV value.
So this patch provisionally enables the SYSCLK to get a valid DSPCLK_DIV for
calculation and then disables it afterward.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Initially, this binding and driver only describe/support playback to
headphones and speakers, and capture from the external microphone, with
GPIO-based jack detection for the headphone jack only.
This driver is useful for the Venice2 board.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch add quirk for Acer Aspire E-572:
- fix external mic
- limit mic boost for internal mic with maximal noise level of -24dB
Signed-off-by: Oleksij Rempel <linux@rempel-privat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
clk_prepare_enable() may fail, so let's check its return value and propagate it
in the case of error.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This will allow a marginal speed improvement when used with a bus that
supports async I/O by reducing the amount of context thrashing between
writes, allowing the bus to be more fully utilised.
Signed-off-by: Mark Brown <broonie@linaro.org>
MacBook Air 2,1 has a fairly different pin assignment from its brother
MBA 1,1, and yet another quirks are needed for pin 0x18 and 0x19,
similarly like what iMac 9,1 requires, in order to make the sound
working on it.
Reported-and-tested-by: Bruno Prémont <bonbons@linux-vserver.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change sam9x5 with wm8731 work in DSP A mode, this will fix the
left/right channel swap issue.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Tested-by: Richard Genoud <richard.genoud@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
According to the SSC specifiation, it should be enabled after DMA is
enabled. So, add trigger operation to make sure the right sequence.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Tested-by: Richard Genoud <richard.genoud@gmail.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The snd_soc_dai_digital_mute() here will be never executed because we only
decrease codec->active in snd_soc_close(). Thus correct it.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch removed the redundant snd_soc_dai_digital_mute() in close() since
it's better to mute in hw_free() which's slightly earlier and symmetrical for
the case of reconfiguration: 'aplay 44k1.wav 48k.wav', for example, will be
open()->hw_params()->prepare(unmute)->playi1ng->hw_free(mute)->hw_params()->
parepare(unmute)->playing->hw_free(mute)->close()
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
A smattering of fixes here, some core ones for the rate combination
issues for things other than simple bitmasks, for readback of byte
controls and for updating the power of value muxes plus a bunch of
driver fixes of varying severity.
The warning fix in the i.MX FIQ driver is fixing a warning introduced
by a previous fix.
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Merge tag 'asoc-v3.13-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.13
A smattering of fixes here, some core ones for the rate combination
issues for things other than simple bitmasks, for readback of byte
controls and for updating the power of value muxes plus a bunch of
driver fixes of varying severity.
The warning fix in the i.MX FIQ driver is fixing a warning introduced
by a previous fix.
In the case of using jackpoll_ms instead of unsol events, the jack
was correctly detected, but ELD info was not refreshed on plug-in.
And without ELD info, no proper restriction of pcm, which can in turn
break sound output on some devices.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I forgot to remove the hp_automute_hook from alc283_fixup_chromebook.
It doesn't need this for other chrome os machine.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that there is a dmaengine driver for the jz4740 DMA core we can use the
generic dmaengine PCM driver. This allows us to remove the custom jz4740-pcm
code completely.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Even if the CONFIG_PM explicity is undefined, let's convert to the
modern PM ops.
Signed-off-by: Ulf Hansson <ulf.hansson@linaro.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
According to WM8731 "PD, Rev 4.9 October 2012" datasheet, when it
works in DSP mode A, LRP = 1, while works in DSP mode B, LRP = 0.
So, fix LRP for DSP mode as the datesheet specification.
Signed-off-by: Bo Shen <voice.shen@atmel.com>
Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
Let the core take care of applying sample rate and sample bits constraints
instead of open-coding this in the driver.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
Since we introduced symmetric_channels and symmetric_samplebits, we implement
these two features to fsl_ssi so as to drop some no-more-needed code and make
the driver neat and clean.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch sets a 0ms depop delay in fixup funtion 'alc_fixup_no_depop_delay'.
And Realteck ALC262 applies this on Intel Baytrail BayleyBay platform to reduce
codec suspend time.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Reviewed-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Create single model for HP.
The headset jack module was difference between other chrome book.
It need to manual control Mic jack detect.
Chrome OS loaded driver by models. Remove old assigned fixup table from
ALC269 fixup list entry.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
By trial and error, I found this patch could work around an issue
where the headset mic would stop working if you switch between the
internal mic and the headset mic, and the internal mic was muted.
It still takes a second or two before the headset mic actually starts
working, but still better than nothing.
Information update from Kailang:
The verb was ADC digital mute(bit 6 default 1).
Switch internal mic and headset mic will run alc_headset_mode_default.
The coef index 0x11 will set to 0x0041.
Because headset mode was fixed type. It doesn't need to run
alc_determine_headset_type.
So, the value still keep 0x0041. ADC was muted.
BugLink: https://bugs.launchpad.net/bugs/1256840
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The normal mode of SSI allows it to send/receive data to/from the first
slot of each period. So we can use this normal mode to trick I2S signal
by puting/getting data to/from the first slot only (the left channel)
so as to support monaural audio playback and recording.
Signed-off-by: Nicolin Chen <b42378@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
It seems that AD1986A cannot manage the dynamic pin on/off for
auto-muting, but rather gets confused. Since each output has own amp,
let's use it instead.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Cc: <stable@vger.kernel.org> [v3.11+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ad_vmaster_eapd_hook() needs to handle the inverted EAPD case
properly, too. Otherwise the output gets broken on Lenovo N100 with
AD1986A codec.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=64971
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ASUS Z35HL laptop also needs the very same fix as the previous one
that was applied to ASUS W7J.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66231
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HD-audio devices tend to take long time for finishing the whole
probing procedure. In this patch, the time-consuming part of the
probing procedure, the codec probe and the rest initializations, are
moved in the work, so that they can be done asynchronously in parallel
with probes of other devices.
Since we already have this mechanism in the driver code for the
firmware and i915 request_symbol() stuff, we just need to enable it
always; the resultant patch even reduces more lines, which is an
additional bonus.
Credit goes to David Henningsson, who suggested this workaround.
Reported-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The static checker found a possible array overflow in atmel/abdac.c:
static checker warning: "sound/atmel/abdac.c:373 set_sample_rates()
error: buffer overflow 'dac->rates' 6 <= 6"
This patch papers over the buggy point, by ensuring that dac->rates[]
update not overflowing the actual array size.
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Support for loading the simple-card module via DeviceTree.
It requests CPU/CODEC information.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
When the probe of snd-hda-intel driver is deferred due to f/w loading
or the nested module loading, complete_all() should be also delayed
until the initialization really finished. Otherwise, vga-switcheroo
client would start switching before the actual init is done.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It seems that EAPD on NID 0x16 is the only control over all outputs on
HP machines with AD1984A while turning EAPD on NID 0x12 breaks the
output. Thus we need to avoid fiddling EAPD on NID. As a quick
workaround, just set own_eapd_ctrl flag for the wrong EAPD, then
implement finer EAPD controls.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66321
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Unbalanced calls to imx_ssi_trigger() may result in endless
SSI activity and thus provoke eternal sound. While on the first glance,
the switch statement looks pretty symmetric, the SUSPEND/RESUME
pair is not: the suspend case comes along snd_pcm_suspend_all(),
which for fsl/imx-pcm-fiq is called only at snd_soc_suspend(),
but the resume case originates straight from the SNDRV_PCM_IOCTL_RESUME.
This way userland may provoke an unbalanced resume, which might cause
the ssi->enabled counter to increase and never return to zero again,
so eventually SSI_SCR_SSIEN is never disabled.
As the information on whether to enable the SSI or not is contained
in the two bits for TE/RE, we save all the software mirroring of
hardware state here and simply use the hardware register itself
to keep the state of whether someone is currently playing or capturing.
This is essentially the same stuff as in sound/soc/fsl/imx-pcm-fiq.c
which I send a patch for three days ago. Astonishing enough this
highly fragile scheme is used twice in parallel to serve the very
same control function, synchronously: Once out of sync you are lost
until reboot.
Note, that these fixes wont prevent state machine distortion on alsa
level to cut sound or the like. It just makes sure we have a chance
to synchronise again later on.
Signed-off-by: Oskar Schirmer <oskar@scara.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This moves us towards being able to remove the ASoC level I/O code which
duplicates regmap functionality. Currently the only difference between
the supported devices in the driver is the regmap so we can replace the
CODEC driver selections with regmap selection instead.
Signed-off-by: Mark Brown <broonie@linaro.org>
If there are symmetry constraints between the playback and the capture channel
set the SNDRV_PCM_INFO_JOINT_DUPLEX flag to let userspace know about this.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
For dmaengine drivers which do not support transfer residue reporting we update
the PCM pointer with period granularity. Set the SNDRV_PCM_INFO_BATCH flag in
this case to let userspace know about this.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
N810 audio driver has stopped working at some point. Probably when
OMAP2 was converted to common clock framework since now call to clk_enable
dumps the stack trace in drivers/clk/clk.c: __clk_enable() due
clk->prepare_count is zero.
Fix this by converting clk_enable/_disable calls to those that take care
of clock prepare/unprepare.
I'm not queueing this to linux-stable since OMAP2 common clock framework
conversion in commit ed1ebc4948 ("ARM: OMAP2: clock: Convert to common clk")
happened before N810 was really usable in mainline and user base for N810 is
anyway small. Potential linux-stable candidates are only those after
commit 3d3a6d18ab ("watchdog: introduce retu_wdt driver").
Signed-off-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
platform_set_drvdata(op, pdata) in pcm030_fabric_probe()
will be overwrited when calling snd_soc_register_card(card),
but cm030_fabric_remove() use drvdata as a type of struct
pcm030_audio_data, so we should move platform_set_drvdata()
below snd_soc_register_card() call.
Signed-off-by: Wei Yongjun <yongjun_wei@trendmicro.com.cn>
Signed-off-by: Mark Brown <broonie@linaro.org>
Originally snd_hrtimer_callback() used iprtd->period_time for
some jiffies based estimation to determine the right moment
to call snd_pcm_period_elapsed(). As timer drifts may well be a
problem, this was changed in commit b4e82b5b78 to be based
on buffer transmission progress, using iprtd->offset and
runtime->buffer_size to calculate the amount of data since last
period had elapsed.
Unfortunately, iprtd->offset counts in bytes, while
runtime->buffer_size counts frames, so adding these to find some
delta is like comparing apples and oranges, and eventually results
in negative delta values every now and then. This is no big harm,
because it simply causes snd_pcm_period_elapsed() being called
more often than necessary, as negative delta is taken for a
large unsigned value by implicit conversion rule.
Nonetheless, the calculation is broken, so one would replace
the runtime->buffer_size by its equivalent in bytes.
But then, there are chances snd_pcm_period_elapsed() is called
late, because calculating the moment for the elapsed period
into delta is based against the iprtd->last_offset, which is not
necessarily the first byte of the period in question, but some
random byte which the FIQ handler left us with in r8/r9 by
accident. Again, negative impact is low, as there are plenty of
periods already prefilled with data, and snd_pcm_period_elapsed()
will probably be called latest when the following period is
reached. However, the calculation is conceptually broken, and we
are best off removing the clever stuff altogether.
snd_pcm_period_elapsed() is now simply called once everytime
snd_hrtimer_callback() is run, which may not be most accurate,
but at least this way we are quite sure we dont miss an end of
period. There is not much extra effort wasted by superfluous
calls to snd_pcm_period_elapsed(), as the timer frequency
closely matches the period size anyway.
Signed-off-by: Oskar Schirmer <oskar@scara.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
No need to have a specific OOM message, since there is generic MM out of memory
message in place.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The recent kernels got regressions on ASUS W7J with ALC660 codec where
no sound comes out. After a long debugging session, we found out that
setting the pin control on the unused NID 0x10 is mandatory for the
outputs. And, it was found out that another magic of NID 0x0f that is
required for other ASUS laptops isn't needed on this machine.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=66081
Reported-and-tested-by: Andrey Lipaev <lipaev@mail.ru>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd_soc_bytes_put treats the data in the binary control as big endian
words, however snd_soc_bytes_get uses the endian of the host machine.
This causes the two functions to be inconsistant with how the mask is
applied on little endian machines.
This patch applies the big_endian format used in snd_soc_bytes_put to
snd_soc_bytes_get.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds 'depop_delay' to struct hda_codec, to indicate a depop delay
time in ms when power-down, in function set_power_state() to D3.
Default value is -1, for a default delay time.
Machine fixup can set a suitable value according to the codec chip and HW audio
design.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch defines a fixup 'alc_fixup_no_depop_delay', which sets alc_spec flag
'no_depop_delay' to indicate skipping delay in Realtek codec suspend/resume.
And Intel Baytrail BayleyBay board applies this fixup to reduce driver
suspend/resume time.
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch defines a flag "no_depop_delay" in alc_spec. If this flag is set,
delay in alc_eapd_shutup and alc_resume will be skipped.
Machine-specific fixup can set this flag to reduce suspend/resume time, if
the codec and hardware analog design can avoid pop noise without this delay.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Mengdong Lin <mengdong.lin@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell assigned alias name for more codecs.
ALC3220 ALC3221 ALC3223 ALC3226 ALC3234 ALC3661.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch adds Gen2 sound support for Renesas R-Car.
But, it is supporting PIO transfer only at this point
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current rcar driver gen.c is using rcar_gen_ops
which was made with the assumption that
Gen1 and Gen2 need different behavior.
but it was not needed.
This patch removes unnecessary complex method.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
rsnd_gen_ops has .path_init/exit callback function
which cares SRU/SSI (if Gen1) SCU/SSIU/SSI (if Gen2)
path settings.
But, the differences between Gen1/Gen2 are cared
in ssi.c/scu.c, and the path itself is same in Gen1/Gen2.
This patch removes .path_init/exit callback.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Current rcar driver is supporting Gen1,
and Gen2 will be supported soon.
Then, some registers are used from Gen1 only,
or from Gen2 only.
To avoid NULL pointer access, this patch adds
register accessible check function.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The repmap initialization difference between Gen1/Gen2 is
only register offset.
This patch separates rsnd_gen1_regmap_init()
into common part and Gen1 specific part.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The array limits are supposed to be in units of u32 instead of in bytes.
The current code has a potential array overflow.
Fixes: c614475b0e ('ALSA: dice: add a proc file to show device information')
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This machine also has mono output if run through DAC node 0x03.
Cc: stable@vger.kernel.org (v3.10+)
BugLink: https://bugs.launchpad.net/bugs/1256212
Tested-by: David Chen <david.chen@canonical.com>
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When locating the memory region relating to a coefficient block written
through a bin file we keep processing the list of regions even after we
have found the region we require. This patch adds a break, so we don't
process redundant list items.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
As the previous commit 1f0bbf03cb added the pin config for the bass
speaker, this patch adds the corresponding LFE-only channel map on
ASUS ET2700.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add a fixup entry for the missing bass speaker pin 0x16 on ASUS ET2700
AiO desktop. The channel map will be added in the next patch, so that
this can be backported easily to stable kernels.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=65961
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Makes the code slightly shorter.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Rajeev Kumar <rajeev-dlh.kumar@st.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
For many drivers using the generic dmaengine PCM driver one of the few (or the
only) things left to do in the drivers remove function is to unregister the PCM
device. This patch adds a resource managed version of snd_dmaengine_pcm_register()
which makes it possible to simplify the remove function as well as the error
path in the probe function for those drivers.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
This patch adds device tree support for the CS42L52 Codec
Signed-off-by: Brian Austin <brian.austin@cirrus.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
This both devices need limit for internal dmic.
[cosmetic change; renamed fixup name by tiwai]
Signed-off-by: Oleksij Rempel <linux@rempel-privat.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
