Similar improvements for ALC262 codec like previous two commits:
assign a better name, either Master or Speaker, for the primary output
controls.
However, in the case of ALC262 codec, the necessary changes are larger
than others because we need to check the possibility of different mixer
amps depending on the pins. The pin 0x16 is mono, and bound with the
dedicated mixer 0x0e while other pins are bound with 0x0c. Thus, there
are two possible volumes.
When only one of them is used, we can name it as "Master". OTOH, when
both are used at the same time, they have to be named uniquely.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of fixed "Front" mixer name, try to assign a better name, e.g.
"Master" or "Speaker" fot the primary output volume controls of ALC260
codec.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When there is only one DAC is used for ALC880, try to assign a better
name, either Speaker or Front, depending on the output pin type.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
s3c24xx dma has the auto reload feature, when the the trnasfer is done,
CURR_TC(DSTAT[19:0], current value of transfer count) reaches 0, and DMA
ACK becomes 1, and then, TC(DCON[19:0]) will be loaded into CURR_TC. So
the transmission is repeated.
IRQ is issued while auto reload occurs. We change the DISRC and
DCON[19:0] in the ISR, but at this time, the auto reload has been
performed already. The first block is being re-transmitted by the DMA.
So we need rewrite the DISRC and DCON[19:0] for the next block
immediatly after the this block has been started to be transported.
The function s3c2410_dma_started() is for this perpose, which is called
in the form of "s3c2410_dma_ctrl(prtd->params->channel,
S3C2410_DMAOP_STARTED);" in s3c24xx_pcm_trigger().
But it is not correct. DMA transmission won't start until DMA REQ signal
arrived, it is the time s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1)
is called in s3c24xx_i2s_trigger().
In the current framework, s3c24xx_pcm_trigger() is always called before
s3c24xx_pcm_trigger(). So the s3c2410_dma_started() should be called in
s3c24xx_pcm_trigger() after s3c24xx_snd_txctrl(1) or
s3c24xx_snd_rxctrl(1) is called in this function.
However, s3c2410_dma_started() is dma related, to call this function we
should provide the channel number, which is given by
substream->runtime->private_data->params->channel. The private_data
points to a struct s3c24xx_runtime_data object, which is define in
s3c24xx_pcm.c, so s3c2410_dma_started() can't be called in s3c24xx_i2s.c
Fix this by moving the call to signal the DMA started to the DAI
drivers.
Signed-off-by: Shine Liu <liuxian@redflag-linux.com>
Signed-off-by: Shine Liu <shinel@foxmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Provide a standard parser for input pins to create the input mixer
and input source controls instead of having a difference one for each
Realtek codec. The new helper parses the codec connections dynamically
isntead of fixed indicies.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Functionality of functions omap_mcbsp_xmit_enable and omap_mcbsp_recv_enable
can be merged into omap_mcbsp_start and omap_mcbsp_stop since API of
those omap_mcbsp_start and omap_mcbsp_stop was changed recently allowing
to start and stop individually the transmitter and receiver.
This cleans up the code in arch/arm/plat-omap/mcbsp.c and in
sound/soc/omap/omap-mcbsp.c which was the only user for those removed
functions.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Commit ca6e2ce086 is setting up few XCCR and
RCCR bits for I2S and DPS_A formats. Part of the bits are already set
for all formats and I believe that XDISABLE and RDISABLE bits are
format independent.
As XCCR and RCCR are found only from OMAP2430 and OMAP34xx, I move setup
of XDISABLE and RDISABLE to where those cpu's are tested and remove format
dependent part for simplicity.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
snd_interval_list() expected a sorted list but did not document this, so
there are drivers that give it an unsorted list. To fix this, change
the algorithm to work with any list.
This fixes the "Slave PCM not usable" error with USB devices that have
multiple alternate settings with sample rates in decreasing order, such
as the Philips Askey VC010 WebCam.
http://bugzilla.kernel.org/show_bug.cgi?id=14028
Reported-and-tested-by: Andrzej <adkadk@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reuse a part of the code of ALC268 parser for ALC269.
This will change the default output volume either to Front or Speaker
depending on the pin configuration.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix a logic error in the range check of the input level control that
would prevent setting any volume less than the maximum.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Modify loops in such way that the register value is checked also after
the timeout condition, just in case the heavy interrupt load etc. caused
the thread to sleep for the time period exceeding the timeout value.
While at it remove an extra ALI_STIMER read from snd_ali_stimer_ready().
Reported-by: Jack Byer <ojbyer@usa.net>
Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Free socdev if snd_soc_init_card() fails.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently when built with DEBUG DAPM will dump information about
the power state decisions it is taking for each widget to dmesg.
This isn't an ideal way of getting the information - it requires
a kernel build to turn it on and off and for large hub CODECs the
volume of information is so large as to be illegible. When the
output goes to the console it can also cause a noticable impact
on performance simply to print it out.
Improve the situation by adding a dapm directory to our debugfs
tree containing a file per widget with the same information in
it. This still requires a decision to build with debugfs support
but is easier to navigate and much less intrusive.
In addition to the previously displayed information active streams
are also shown in these files.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is tested by ms7724se
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is very simple driver for ALSA
It supprt headphone output and stereo input only
This patch is tested by ms7724se
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add support for the tlv320aic3x CODEC on the Simtec Hermes board.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fixup the device changes by modifying the files that we just removed the
explicit device creation from with i2c_register_board_info() until this
can be moved into the relevant board files.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The tlv320aic3x driver managed its own i2c device, instead of an extant
one created by the board support code. Change the code to make it so that
the driver binds to an extant (in this case i2c) device.
Add explict tlv320aic33 as well as tlv320aic3x to the supported device
table and remove the old driver bindings from the users of this code.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add core support for the range of S3C24XX Simtec boards with TLV320AIC23
CODECs on them. Since there are also boards with similar IIS routing the
AMP and the configuration code is placed in a core file for re-use with
other CODEC bindings.
Signed-off-by: Ben Dooks <ben@simtec.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Configures DMA sync mode depending on McBSP operating mode value.
The value is configurable by McBSP instance. So, depending
on McBSP operating mode, the DMA sync mode is passed from
omap-mcbsp to omap-pcm. Besides that, it also configures
McBSP threshold value depending on which McBSP mode is activated.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch changes the way DMA is done in omap-pcm.c
in order to reduce power consumption. There is no need
to have so much SW control in order to have DMA in idle
state during audio streaming. Configuring McBSP threshold value
and DMA to FRAME_SYNC are sufficient.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
All these steps are required for ASoC to behave correctly.
rccr and xccr are format dependent, for example TDM audio
has different values than I2S or DSP_A. Also the
omap_mcbsp_xmit_enable and/or omap_mcbsp_recv_enable must
be called right after the DMA has started.
This provides no longer L and R channels switching at random.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This is, no RFIG or XFIG (Not defined in 34xx), correct
initiliazation of rccr and xccr.
Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Align DMA address to DMA burst transaction sizes.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Improve DMA transfers by enabling Burst transaction.
Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This driver is very simple.
It support playback only now.
This patch is tested by ms7724se board.
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
> Then it's a driver bug. If unaligned period size is allowed, it means
> that the irq is really generated in that period, not at the buffer
> boundary. Otherwise, it must have a proper hw-constraint to align the
> period size to the buffer size.
This patch will fix the bug metioned in the above mail. Force the peroid
size to be aligned with the buffer size.
Based and tested on linux-2.6.31-rc6.
Signed-off-by: Shine Liu <shinel@foxmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The current PCM core has the following problems regarding PCM draining
in non-blocking mode:
- the current f_flags isn't checked in snd_pcm_drain(), thus changing
the mode dynamically via snd_pcm_nonblock() after open doesn't work.
- calling drain in non-blocking mode just return -EAGAIN error, but
doesn't provide any way to sync with draining.
This patch fixes these issues.
- check file->f_flags in snd_pcm_drain() properly
- when O_NONBLOCK is set, PCM core sets the stream(s) to DRAIN state
but quits ioctl immediately without waiting the whole drain; the
caller can sync the drain manually via poll()
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Used for applications such as direct bluetooth connections on
smartphones which don't go via the CPU. This used to be supported
before the refactoring to share code but this check was removed
during that move.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There are many variants of Toshiba laptops with ALC268 codec, and
it seems that a few of them don't work with model=toshiba preset
since they have the secondary ALC268 codec just for HDMI output.
This is a regression due to the previous clean-up work to merge all
Toshiba quirk entries into a single check.
This patch adds the identification of such laptops to apply the
standard BIOS-probing method. Unfortunately, Toshiba laptops have
all the same PCI SSID, so we need to check the codec SSID to identify
each device.
Tested-by: Alexey Dobriyan <adobriyan@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the CODEC does not provide a set_bias_level() then update the
bias_level variable for it since other parts of the system expect
that to be maintained.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The BIOS pin configs are in fact correct and shall not be overwritten.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
HP Compaq 6530s and 6531s internal speaker is silence or becomes silence
within 1 minute after fresh boot. It is found that pin 0x1c must be set to
PIN_OUT mode to make the speaker work. This is weird - line-in pin 0x1c and
speaker pin 0x16 seem to be unrelated.
The codec differences before/after patch are:
@@ Node 0x17 [Pin Complex] wcaps 0x40020b:
Pin Default 0x41a6e130: [N/A] Mic at Ext Rear
Conn = Digital, Color = White
DefAssociation = 0x3, Sequence = 0x0
Misc = NO_PRESENCE
- Pin-ctls: 0x24: IN
+ Pin-ctls: 0x40: OUT
@@ Node 0x1c [Pin Complex] wcaps 0x40018d:
Pin Default 0x41813021: [N/A] Line In at Ext Rear
Conn = 1/8, Color = Blue
DefAssociation = 0x2, Sequence = 0x1
- Pin-ctls: 0x24: IN VREF_80
+ Pin-ctls: 0x40: OUT VREF_HIZ
Unsolicited: tag=00, enabled=0
Connection: 1
0x24
Tests show that it won't impact (external) Mic recording.
Reported-by: "Lin, Ming M" <ming.m.lin@intel.com>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The auto-mic clean-up patches caused regressions on some ALC268 models
that have no proper input_mux but with "Input Source" mixer elements.
Such a combination results in Oops when accessed.
[A reason why set_capture_mixer() isn't used in patch_alc268() is that
ALC268 codec have HDA_OUTPUT direction for capture volumes unlike other
codecs. Thus it needs own definitions of capture elements.]
This patch fixes the issues:
- Add a capture mixer definition without input-source
- Use the new capture mixer appropriately
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The WM8993 provides digital sidetone paths and also allows each
channel on the audio interface to be routed separtately to the
DACs and ADCs.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Currently DAPM interfaces with the audio streams to and from the
processor at the DAC and ADC widgets. As the digital capabilities
of parts increases this is becoming a less and less able to meet
the needs of parts.
To meet the needs of these devices create new widgets interfacing
with the TDM bus but not integrated into any other functionality.
Audio can then be routed to and from these widgets using existing
routing widgets.
A slot number is provided in the definition but this is currently
not used yet. This is intended to support devices which can use
more than one TDM slot on a single interface.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Joonyoung Shim reports that S3C64xx I2S is working on the NCP boards so
allow it to be selected in Kconfig.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmciro.com>
The data format configuration for S3C64xx IISv2 was hardcoded for IISMOD
register. This patch changes to the defined values it.
And instead of bits 9 and 10 of IISMOD we should clear bits 13 and 14.
Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Note that the number of slots used internally is specified in terms
of stereo slots while the external API works with mono slots.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
When used without the PLL we were accidentally clearing the MCLK/2
divider, resulting in a double rate SYSCLK when the divider should
have been used.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Speaker and headphone outputs do not need to be handled separately
since they can't be part of the same path.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
If the system doesn't have any DAPM widgets then we can't use their
state to check if the bias level for the codec should be up.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is a mistake in current uda134x_mute function: mute_reg has been
changed in line 162 or line 164, so uda134x_write should write
"mute_reg" but not "mute_reg & ~(1<<2)" to
UDA134X_DATA010.
Signed-off-by: Shine Liu <shinel@foxmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Ensure that userspace can remove only user controls. Controls created
by kernel drivers must not be removed because they might be referenced
in calls to snd_ctl_notify().
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move the decrementing of the user controls counter from
snd_ctl_elem_remove to snd_ctl_remove_unlocked_id; this saves the
separate locking of the controls semaphore, and therefore removes
a harmless race.
Since the purpose of the function is to operate on user controls (the
control being unlocked is just a prerequisite), rename it to
snd_ctl_remove_user_ctl.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use a common exit path to release the mutex and to return a possible
error.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make sure that no user element that has no values can be added.
The check for count>1024 is not needed because the count is checked
later for the individual control types.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Enhance period_index accuracy, particularly just before buffer rewind, by
making use of DMA interrupt status flags in addition to simply counting up
interrupts.
Created against linux-2.6.31-rc5.
Tested on Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Use newly implemented DMA channel self linking on OMAP1510 like on other OMAP
models. Remove unnecessary DMA transfer restart from interrupt handler
routine.
The interrupt routine used to maintain a period index, originally needed for
counting up periods up to a full buffer in order to restart the DMA transfer.
For some time, this counter is also used as a replacement for hardware DMA
progress counter that has been found unusable on OMAP1510 in case of playback.
Thus, the period index calculation cannot be omitted completely. However, the
accuracy of this counter can still suffer from missing DMA interrupts.
In order to work correctly, it requires patch 1 from this series also applied:
[RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510
Created against linux-2.6.31-rc5.
Tested on Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Acked-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The previous fix removed the definition of num_adc_nids wrongly, and
this resulted in the missing input-source control. Now readded again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the strings related to capture in order to be
interpreted correctly by alsamixer and possible other
UI based mixer applications.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Move the definition of the "generic" IRQ in the process.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Ben Dooks <ben-linux@fluff.org>
This patch adds support for passing platform data to ac97 bus devices
from PXA2xx-AC97 driver..
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
There is one instance of McASP on DA850/OMAP-L138 SoC. This is
connected to TLV320AIC3106 codec for audio playback and capture.
This patch adds audio support on this platform. Some of the
structure prefix names which are common for DA830/OMAP-L137 EVM and
DA850/OMAP-L138 EVM have been renamed to da8xx from da830.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch adds a DAI format: Codec bit clock master and frame sync slave,
to the driver.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral has FIFO
support. This FIFO provides additional data buffering. It also provides
tolerance to variation in host/DMA controller response times.
The read and write FIFO sizes are 256 bytes each. If FIFO is enabled,
the DMA events from McASP are sent to the FIFO which in turn sends DMA requests
to the host CPU according to the thresholds programmed.
More details of the FIFO operation can be found at
http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=
sprufm1&fileType=pdf
This patch adds support for FIFO configuration. The platform data has a
version field which differentiates the McASP on different SoCs.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8993 analogue control is shared with other devices in the same
product line. Since this is a very substantial proportion of the
driver move the definitions of these controls into a new wm_hubs module
which allows them to be shared between the two.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
A few improvements for IDT 92HD83xxx codec pareser:
- Remove unused / deprecated mixer-amp controls
- Handle d-mics as normal inputs since this codec has no separate
MUXes for analog and digital
- Don't create duplicated controls for capture volumes with Mux
capture volumes
Signed-off-by: Takashi Iwai <tiwai@suse.de>
There has been an ad1836 driver in sound/blackfin based on traditional alsa.
The new driver is based on asoc. The architecture of ad1836 codec driver is
very much like ad1938.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Dynamically control and control only the needed output amplifier
muting/un-muting.
The original code was muting and un-muting the following output
amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same time
regardless which pin is actually in use at the given moment.
Move these as separate PGA so only the needed amplifier will be touched.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
According to the function dapm_dac_check_power() in
sound/soc/soc-dapm.c, dac power can't be on/off stand-alone without any
output widget as sink. And according to dapm_adc_check_power(), adc
power can't be on/off stand-alone without any input widget as source. So
we can't only define some stand-alone SND_SOC_DAPM_DAC/SND_SOC_DAPM_ADC
to hope their power can be managed dynamically.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Enable line-out detection for IDT/STAC codecs only when speaker pins
exist. In some cases, the speaker itself is identified as line-out,
and this confuses the situation. Only the extra line-outs should do
auto-muting.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
the hdsp driver refuses to report any information via the proc
interface, if the io box is not connected. with this patch, the
content of the control and status registers is printed before the
iobox check.
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
With auto config model of alc268 realtek codec, it allows to select any
of possible available digital microphone inputs when only one is
available. For example, when only digital mic in nid 0x12 is available,
on second input source it will allow you to select unavailable digital
mic in nid 0x13. The problem is that selecting unavailable digital mic
creates a source of noise when recording (I'm not sure if this happens
on all machines with alc268 and only one digital mic input, but testing
on a quanta uw1 netbook a lot of noise is introduced in recording from
digital mic 0x12/first input source, when you select the unavailable
digital mic 0x13 for capture source 0x24 in the second input source in
mixer).
Then to avoid noise when recording from digital mic with auto model in
this case, prevent a digital mic input source to be selected if
microphone is not available.
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Move static codes to setup from init_hook for each model.
Also, use the common auto-mic selection helper for devices that support
auto-mic selection. They just need to set up ext_mic, int_mic and
auto_mic flag in the setup section.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Store the TDM slot width then if it's set use that rather than the
sample size to calculate BCLK. Leave imposing constraints to the
core (which should do this but doesn't yet) or machine driver.
Also allow 0 TDM slots to be configure (for use when disabling TDM).
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added setup hook to ALC preset struct to be called at in the parser
but not at each init callback.
This can be used for setting up the static pins, etc, while the
init hook should be used for updating the status again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Created a white-list to enable MSI since some devices require MSI
explicitly due to BIOS/ACPI problems. Simply using a quirk list.
As the first case, take HP Compaq CQ40.
Reference: Novell bnc#529971
https://bugzilla.novell.com/show_bug.cgi?id=529971
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix soc build errors when I2C is built as a loadable module:
(.text+0x5d26b): undefined reference to `i2c_master_send'
soc-cache.c:(.text+0x5d32d): undefined reference to `i2c_transfer'
Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Some Realtek codecs don't provide the full connections for certain pins
from each ADC; e.g. ACL662/ALC272 gives only one of two digital-mic pins
for each ADC. Thus, depending on the digital mic pin, the ADC/MUX to be
used has to be chosen properly.
This patch adds the check of the connectivity of pins at auto-mic mode.
If no proper connectivity is found, auto_mic flag is turned off to be
sure.
Also the mux_idx is determined during this check so it won't be checked
in the unsol event any more.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC269 and ALC861-VD parsers override the ADC definitions
unconditionally without checking the spec definition. This causes
the problem when any inconsistent ADC is set up in the device quirk
(like ALC272 with digital-mic).
This patch avoids the overriding by adding the proper checks.
Reference: Novell bnc#529467
https://bugzilla.novell.com/show_bug.cgi?id=529467
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the compile warning due to different integer types used in min():
sound/usb/usbaudio.c: In function 'init_substream_urbs':
sound/usb/usbaudio.c:1087: warning: comparison of distinct pointer types lacks a cast
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the support for automatic mic selection via plugging for
Realtek codecs (in auto-probing mode). The auto-mic mode is enabled
only when one internal mic and one external mic are present.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If any OSS support is enabled, regardless of built-in or module,
sound_core claims full OSS major number (that is, the old 0-255
region) to trap open attempts and request sound modules using custom
module aliases. This feature is redundant as chrdev already has such
mechanism. This preemptive claiming prevents alternative OSS
implementation.
The custom module aliases are scheduled to be removed and the previous
patch made soundcore emit the standard chrdev aliases too to help
transition.
This patch schedule the feature for removal in a year and makes it
optional so that developers and distros can try new things in the
meantime without rebuilding the kernel. The pre-claiming can be
turned off by using SOUND_OSS_CORE_PRECLAIM and/or kernel parameter
soundcore.preclaim_oss.
As this allows sound minors to be individually grabbed by other users,
this patch updates sound_insert_unit() such that if registering
individual device region fails, it tries the next available slot.
For details on removal plan, please read the entry added by this patch
in feature-removal-schedule.txt .
Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Till now missing OSS devices emitted sound-slot/service-* module
alises instead of the standard char-major-* if a missing device number
is opened if soundcore is loaded. The custom module aliases don't
have any inherent benefit than backward compatibility.
sound-slot/service-* module aliases is scheduled to be removed and to
help the transition this patch makes soundcore emit the standard
module alises along with the custom ones.
Signed-off-by: Tejun Heo <tj@kernel.org>
Cc: Alan Cox <alan@lxorguk.ukuu.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the 2.1.6 kernel, the output loop in midi_poll() was changed to
enable interrupts during the outputc() call. Unfortunately, the check
whether the device has accepted the current byte ("ok") was moved behind
the code that removes the byte from the output queue, so one byte would
be lost every time the hardware FIFO is full.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow the interval timer to be programmed with its full 96 kHz
precision.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Using more packets in one URB do avoid interrupts does not make sense
when we have a sync pipe whose packets generate interrupts more often.
Therefore, limit the URB size to the synchronization packet interval.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Without the initialization of vmaster NID, the dB information got
confused for ALC269 codec.
Reference: Novell bnc#527361
https://bugzilla.novell.com/show_bug.cgi?id=527361
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
The previous auto-mic patch for STAC/IDT codecs causes the Oops on
machines without digital mic pins. This patch fixes the problem.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for audio on DA830 EVM- here McASP1 is interfaced to
TLV320AIC3106 codec.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This fixes a build failure for 2.6.31-rc4-rt1 (ARCH=arm, s3c2410_defconfig):
CC [M] sound/soc/s3c24xx/s3c2443-ac97.o
sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: type defaults to 'int' in declaration of 'DECLARE_MUTEX'
sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: parameter names (without types) in function declaration
sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_read':
sound/soc/s3c24xx/s3c2443-ac97.c:59: error: 'ac97_mutex' undeclared (first use in this function)
sound/soc/s3c24xx/s3c2443-ac97.c:59: error: (Each undeclared identifier is reported only once
sound/soc/s3c24xx/s3c2443-ac97.c:59: error: for each function it appears in.)
sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_write':
sound/soc/s3c24xx/s3c2443-ac97.c:93: error: 'ac97_mutex' undeclared (first use in this function)
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The patch changes the line discipline name registered in include/linux/tty.h
and updates the ams-delta machine driver to use it.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The dma setup code assumes that the buffer size is a multiple
of the period size.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
dai is a parameter to the functions, so use it instead of
looking it up.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Simultaneous audio playback and capture on OMAP1510 can cause that second
stream is stalled if there is enough delay between startup of the audio
streams.
Current implementation of the omap_mcbsp_start is starting both transmitter
and receiver at the same time and it is called only for firstly started
audio stream from the OMAP McBSP based ASoC DAI driver.
Since DMA request lines on OMAP1510 are edge sensitive, the DMA request is
missed if there is no DMA transfer set up at that time when the first word
after McBSP startup is transmitted. The problem hasn't noted before since
later OMAPs are using level sensitive DMA request lines.
Fix the problem by changing API of omap_mcbsp_start and omap_mcbsp_stop by
allowing to start and stop individually McBSP transmitter and receiver
logics. Then call those functions individually for both audio playback
and capture streams. This ensures that DMA transfer is setup before
transmitter or receiver is started.
Thanks to Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> for detailed problem
analysis and Peter Ujfalusi <peter.ujfalusi@nokia.com> for info about DMA
request line behavior differences between the OMAP generations.
Reported-and-tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Tony Lindgren <tony@atomide.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Extend set_tdm_slot to allow the user to arbitrarily set the frame width
and active TX/RX slots.
Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c
still doesn't handle the slot_width override.
While being there, correct an incorrect use of SlotsPerFrm(7) use in
bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ).
(this series is meant for Mark's for-2.6.32 branch)
Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch is a workaround for the problem of several subsequent control
statements not being applied correctly to the codec controller (modem).
In order to follow the hook switch state change from handset to handsfree
while
in full duplex mode, two consecutive +VLS control commands were sent to the
modem. The first one was M1 (microphone only), the seconds one was M1S1 (both
microphone and speaker). As there was no real modem handshaking procedure
implemented, neither in the codec nor in the machine driver part of the line
discipline, the modem was having the second command missed.
Since a possibility to switch to microphone only mode (and speaker only mode
as well) seams of no value, I have modified the code to issue single M1S1
command only for any of those cases.
Tested on my Amstrad Delta.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This patch adds debugging statement that can help in tracing
how the driver is trying to control the codec device.
Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The WM8776 is a high performance, stereo audio CODEC with five channel
input selector. The WM8776 is ideal for surround sound processing
applications for home hi-fi, DVD-RW and other audio visual equipment.
This driver implements support for most WM8776 features - currently the
ADC automatic level control/limiter functionality is omitted.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This adds support for i.mx27_visstrim_sm10 board machine driver which
uses an i.mx27 processor plus a wm8974 codec.
It has been tested on a visstrim_sm10 board.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This adds support for DAI platform for the SSI present in MXC platforms.
It currently does not support i.MX3, the only thing necessary to do
this is to export DMA data for i.MX3 interface which I haven't done
because I don't have a i.MX3 based board available.
It has been tested on i.MX27 board.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
This adds support for DMA platform valid for i.MX1 and i.MX2 platforms.
This is not valid for i.MX3 since it doesn't share the same DMA
interface than i.MX1 and i.MX2.
It has been tested on i.MX27 board.
Signed-off-by: Javier Martin <javier.martin@vista-silicon.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Power management for the cs4270 codec is currently implemented as part
of the i2c_driver struct. The disadvantage of doing it this way is that
the callbacks registered in the snd_soc_card struct are called _before_
the codec's callbacks.
That doesn't work, because the snd_soc_card callbacks will most likely
switch down the codec's power domains or pull the reset GPIOs, and
hence make the i2c communication bail out.
Fix this by binding the suspend and resume code to the
snd_soc_codec_device driver model and let the I2C functions only call
the SoC core function for resume and suspend, which do nothing currently
but will do later.
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Cc: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that
s->runtime->control->appl_ptr can wrap around to the beginning of the
buffer. This change fixes this problem.
Signed-off-by: John Bonesio <bones@secretlab.ca>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: hda - Read buffer overflow
ALSA: hda: Correct EAPD for Dell Inspiron 1525
ALSA: hda: warn on spurious response
ALSA: hda: remember last command for each codec
ALSA: hda: read CORBWP inside reg_lock
ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_io
ALSA: hda: take cmd_mutex in probe_codec()
ALSA: hda: track CIRB/CORB command/response states for each codec
ALSA: hda - Fix quirk for Toshiba Satellite A135-S4527
When the line-out jack is plugged/unplugged, the driver needs to check
the headphone plug, not only the line-out jack itself. Otherwise the
headphone or the speaker may be wrongly muted/unmuted.
As a result, both STAC_HP_EVENT and STAC_LO_EVENT need to call the
same function, stac92xx_hp_detect().
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This converts all the Wolfson drivers using this format (the only devices
that do) except WM8753 to use it.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As part of this refactoring the type of the CODEC hw_read operation
is changed to match the regular read operation.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
While writes tend to be able to use a fairly bus independant format to
do the writes reads are all bus specific. To allow us to factor out
this code include the bus type as a parameter when setting up the
cache.
Initially just use this to factor out hw_write_t for I2C.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The commit fefd67f31e
ALSA: hda - Add line-out jack detection on IDT/STAC codecs
enabled wrong pins for jack detections. Fixed to the correct ones.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Return STRPIPE instead of EBADF when userspace attempts to rewind
of forward a stream that was suspended in meanwhile, so that it
can be recovered by snd_pcm_recover().
This was causing Pulseaudio to unload the ALSA sink module under a race
condition when it attempted to rewind the stream right after resume from
suspend, before writing to the stream which would cause it to revive the
stream otherwise. Tested to work with Pulseaudio patched to attempt to
snd_pcm_recover() upon receiving an error from snd_pcm_rewind().
Signed-off-by: Lubomir Rintel <lkundrak@v3.sk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The new IbexPeak HDMI codec has 3 pin nodes and 2 converter nodes.
Here we assume only the first ones will be used.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Check whether index is within bounds before testing the element.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The commit 24918b61b5 statically changes
the model from dell-bios to dell-3stack to solve the sound decreasing
regression (http://lkml.org/lkml/2008/9/12/203), however it leads to another
problem that the 2nd headphone jack doesn't work
(https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3987). So I think
the commit 249**2dc is just a workaround. I would like to give a true solution
here.
The datasheet for STAC9228 says, GPIO2 is the same pin as VOL DOWN, and
the EAPD pin is GPIO0. This is why the sound decreases if we set EAPD as
GPIO2. This patch changes EAPD to GPIO0 to solve the problem.
Signed-off-by: Chengu Wang <wangchengu@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This converts the last CORBWP access outside of reg_lock.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Just for safety. azx_init_cmd_io() and azx_free_cmd_io() may be
called when switching to single command mode.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now that each codec will have its own module, it is possible
for the user to load one codec while another one is running.
So cmd_mutex would be a safe addition to probe_codec().
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Recently we hit a bug in our dev board, whose HDMI codec#3 may emit
redundant/spurious responses, which were then taken as responses to
command for another onboard Realtek codec#2, and mess up both codecs.
Extend the azx_rb.cmds and azx_rb.res to array and track each codec's
commands/responses separately. This helps keep good codec safe from
broken ones.
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Use model=lenovo instead of model=dallas for Toshiba Satellite A135-S4527
with ALC861-VD codec.
Reference: Novell bnc#526325
https://bugzilla.novell.com/show_bug.cgi?id=526325
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Takashi Iwai