This is a quick fix for the below two issues when building spdif as modules.
1) If modprobing modules in order: (Step 1) snd-soc-fsl-spdif -> (Step 2)
snd-soc-imx-spdif -> (Step 3) snd-soc-spdif-tx/rx, we will fail to create
imx-spdif card and dai link unless we rmmod snd-soc-imx-spdif and modprobe
it again due to the execution platform_driver_unregister() in probe() when
meeting -EPROBE_DEFER at Step 2.
2) After "imx-spdif sound-spdif.17: dit-hifi <-> 2004000.spdif mapping ok",
'rmmod snd-soc-imx-spdif' would cause kernel dump with warning:
WARNING: CPU: 0 PID: 1301 at /home/rmk/git/linux-rmk/fs/sysfs/dir.c:915 sysfs_hash_and_remove+0x84/0x90()
sysfs: can not remove 'dapm_widget', no directory
This should be caused by disordered resourse releasing of the whole link.
And trying to unregister the card and then CODEC dev can't fix this issue.
Thus this patch just provides a simple fix to these two bugs by using the
snd-soc-dummy in the core instead of seperate snd-soc-spdif-tx/rx so that
there's no need to handle the registering and unregistering of CODEC or
CODEC dai any more.
Signed-off-by: Nicolin Chen <Guangyu.Chen@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
From "ASoC: make snd_soc_dai_link more symmetrical", can we see that
the name of CPU DAI maybe omitted. If the DAI name is omitted, try to
use the component name instead.
Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
The ASoC core assumes that the PCM component of the ASoC card transparently
moves data around and does not impose any restrictions on the memory layout or
the transfer speed. It ignores all fields from the snd_pcm_hardware struct for
the PCM driver that are related to this. Setting these fields in the PCM driver
might suggest otherwise though, so rather not set them.
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Signed-off-by: Mark Brown <broonie@linaro.org>
The fixes here are all driver specific ones, none of which particularly
stand out but all of which are useful to users of those drivers.
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Merge tag 'asoc-v3.13-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v3.13
The fixes here are all driver specific ones, none of which particularly
stand out but all of which are useful to users of those drivers.
On the Dell machines with codec whose Subsystem Id is 0x10280640,
no external microphone can be detected when plugging a 3-ring headset.
Using ALC255_FIXUP_DELL1_MIC_NO_PRESENCE can fix this problem.
The codec (Vendor ID: 0x10ec0255) on the machine belongs 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>
When wm8904 work in DSP mode B, we still need to configure it to
work in DSP mode. Or else, it will work in Right Justified mode.
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
Some devices are getting very close to the limit whilst polling the RAM
start, this patch adds a small delay to this loop to give a longer
startup timeout.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
When the process is sleeping at the SNDRV_PCM_STATE_PAUSED
state from the wait_for_avail function, the sleep process will be woken by
timeout(10 seconds). Even if the sleep process wake up by timeout, by this
patch, the process will continue with sleep and wait for the other state.
Signed-off-by: JongHo Kim <furmuwon@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the rates declared in the CPU DAI parameters:
- SNDRV_PCM_RATE_KNOT and the discrete rates SNDRV_PCM_RATE_xxx should
not be used with SNDRV_PCM_RATE_CONTINUOUS,
- SNDRV_PCM_RATE_CONTINUOUS asks for rate_min and rate_max,
- the device may do streaming down to 5512Hz.
Signed-off-by: Jean-Francois Moine <moinejf@free.fr>
Signed-off-by: Mark Brown <broonie@linaro.org>
Reported-by: Kyung-Kwee Ryu <kyung-kwee.ryu@wolfsonmicro.com>
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@linaro.org>
Cc: stable@vger.kernel.org
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>
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>
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>
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>
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>
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>
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 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>
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>
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>
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.
