The pin setup for Dell S14 quirk is rather wrong for the latest driver.
Fixed pin 0x0a, 0x0b, 0x0d and 0x0f.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Remove unnecessary (and buggy) init sequences left for IDT92HD83*
codecs in the previous fixes. The DACs are now dynamically connected,
thus shouldn't be set statically in init verbs. Also, the mono_nid
is detected dynamically, thus shouldn't be set staticaly, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'devel' of master.kernel.org:/home/rmk/linux-2.6-arm: (257 commits)
[ARM] Update mach-types
ARM: 5636/1: Move vendor enum to AMBA include
ARM: Fix pfn_valid() for sparse memory
[ARM] orion5x: Add LaCie NAS 2Big Network support
[ARM] pxa/sharpsl_pm: zaurus c3000 aka spitz: fix resume
ARM: 5686/1: at91: Correct AC97 reset line in at91sam9263ek board
ARM: 5640/1: This patch modifies the support of AC97 on the at91sam9263 ek board
ARM: 5689/1: Update default config of HP Jornada 700-series machines
ARM: 5691/1: fix cache aliasing issues between kmap() and kmap_atomic() with highmem
ARM: 5688/1: ks8695_serial: disable_irq() lockup
ARM: 5687/1: fix an oops with highmem
ARM: 5684/1: Add nuc960 platform to w90x900
ARM: 5683/1: Add nuc950 platform to w90x900
ARM: 5682/1: Add cpu.c and dev.c and modify some files of w90p910 platform
ARM: 5626/1: add suspend/resume functions to amba-pl011 serial driver
ARM: 5625/1: fix hard coded 4K resource size in amba bus detection
MMC: MMCI: convert realview MMC to use gpiolib
ARM: 5685/1: Make MMCI driver compile without gpiolib
ARM: implement highpte
ARM: Show FIQ in /proc/interrupts on CONFIG_FIQ
...
Fix up trivial conflict in arch/arm/kernel/signal.c.
It was due to the TIF_NOTIFY_RESUME addition in commit d0420c83f ("KEYS:
Extend TIF_NOTIFY_RESUME to (almost) all architectures") and follow-ups.
Add the quirk entry for HP dv6. Also add a workaround for the headphone
detection by setting hp_detect=1 beforehand. Without this, the driver
won't do auto-muting because BIOS doesn't give any HP pin but only a
line-out pin.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's possible that hp_detect is set even though no headphone pin is
detected. The driver issues, however, an unsol event only to hp_pins[0],
which can be invalid.
This patch adds the check of the valid pin to send an unsol event
at initialization and resume callbacks.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
IDT92HD73xx and STAC927x codecs use GPIO0 bit as EAPD on many machines.
However, currently we don't set it unless the model is specified just
for safety reason. But, most machines do need this bit, so this safety
handling is rather annoying.
This patch enables GPIO0 setup as default for them. Many HP / Dell
laptops should work even without model override with this change.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Error handling code following a kzalloc should free the allocated data.
Error handling code following an ioremap should iounmap the allocated data.
The semantic match that finds the first problem is as follows:
(http://www.emn.fr/x-info/coccinelle/)
// <smpl>
@r exists@
local idexpression x;
statement S;
expression E;
identifier f,f1,l;
position p1,p2;
expression *ptr != NULL;
@@
x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...);
...
if (x == NULL) S
<... when != x
when != if (...) { <+...x...+> }
(
x->f1 = E
|
(x->f1 == NULL || ...)
|
f(...,x->f1,...)
)
...>
(
return \(0\|<+...x...+>\|ptr\);
|
return@p2 ...;
)
@script:python@
p1 << r.p1;
p2 << r.p2;
@@
print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line)
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* topic/usb-audio:
ALSA: usb-audio - Fix types taken in min()
sound: usb-audio: do not make URBs longer than sync packet interval
sound: usb-audio: add MIDI drain callback
sound: usb-audio: use multiple output URBs
sound: usb-audio: use multiple input URBs
sound: usb-audio: Xonar U1 digital output support
* topic/tlv-minmax:
ALSA: usb-audio - Correct bogus volume dB information
ALSA: usb-audio - Use the new TLV_DB_MINMAX type
ALSA: Add new TLV types for dBwith min/max
* topic/soundcore-preclaim:
sound: make OSS device number claiming optional and schedule its removal
sound: request char-major-* module aliases for missing OSS devices
chrdev: implement __[un]register_chrdev()
* topic/oss:
ALSA: allocation may fail in snd_pcm_oss_change_params()
sound: vwsnd: Fix setting of cfgval and ctlval in li_setup_dma()
sound: fix OSS MIDI output data loss
* topic/hda: (92 commits)
ALSA: hda - Use auto model for HP laptops with ALC268 codec
ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel + digital
ALSA: hda - Add support of Alienware M17x laptop
ALSA: hda - Remove dead codes from patch_sigmatel.c
ALSA: hda - Fix input source selection of IDT92HD73xx
ALSA: hda - Fix obsolete CONFIG_SND_DEBUG_DETECT
ALSA: hda - Unmute docking line-out as default with AD1984A codec
ALSA: hda - Add another entry for Nvidia HDMI device
ALSA: hda - Add missing GPIO initialization for AD1984A laptop model
ALSA: hda - Add support of docking auto-mute/mic for AD1984A laptop model
ALSA: hda - Fix ALC268/ALC269 headphone pin routing
ALSA: hda - Create "Digital Mic Capture Volume" correctly for IDT codecs
ALSA: hda - Add more quirk for HP laptops with AD1984A
ALSA: hda - Add / fix model entries for HD-audio driver
ALSA: hda - Add full audio support on Acer Aspire 7730G notebook
ALSA: hda - Improve auto-cfg mixer name for ALC662
ALSA: hda - Improve auto-cfg mixer name for ALC861-VD
ALSA: hda - Improve auto-cfg mixer name for ALC262
ALSA: hda - Improve auto-cfg mixer name for ALC260
ALSA: hda - Improve auto-cfg mixer name for ALC880
...
* topic/asoc: (226 commits)
ASoC: au1x: PSC-AC97 bugfixes
ASoC: Fix WM835x Out4 capture enumeration
ASoC: Remove unuused hw_read_t
ASoC: fix pxa2xx-ac97.c breakage
ASoC: Fully specify DC servo bits to update in wm_hubs
ASoC: Debugged improper setting of PLL fields in WM8580 driver
ASoC: new board driver to connect bfin-5xx with ad1836 codec
ASoC: OMAP: Add functionality to set CLKR and FSR sources in McBSP DAI
ASoC: davinci: i2c device creation moved into board files
ASoC: Don't reconfigure WM8350 FLL if not needed
ASoC: Fix s3c-i2s-v2 build
ASoC: Make platform data optional for TLV320AIC3x
ASoC: Add S3C24xx dependencies for Simtec machines
ASoC: SDP3430: Fix TWL GPIO6 pin mux request
ASoC: S3C platform: Fix s3c2410_dma_started() called at improper time
ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop
ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DAI
OMAP: McBSP: Use textual values in DMA operating mode sysfs files
ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510
ASoC: Select core DMA when building for S3C64xx
...
This patch fixes the following bugs:
- only reprogram bitdepth if it has changed since last call to hw_params.
- add locking inside ac97_read/write functions:
When reprogramming sample depth, the ac97 unit has to be disabled,
which should not be done in the middle of codec register accesses.
- retry timed-out codec register accesses.
- wait for status bits to set/clear when starting/stopping various
functional blocks; very important after reenabling AC97 unit else
sound may be distorted (e.g. high-pitch noise in 1kHz sine wave).
- clear fifos before/after starting/stopping RX/TX.
- longer timeouts waiting for PSC/AC97 ready after cold reset
with certain codecs this can take ridiculous amounts of time.
Run-tested on various Au1200 platforms with various codecs.
Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Increase the limit of PCM substreams to 128. The default value is
unchanged; only the max accept value is increased.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added the debug proc file to see or change the snd_pcm_hardware fields
to emulate. The parameters can be changed by writing to a proc file like:
# echo periods_min 4 > /proc/asound/card1/dummy_pcm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add appropriate const prefix to char * arguments in proc helper functions.
Also fixed the caller side to be proper const pointers.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Re-export snd_pcm_format_name() function to be used outside the PCM core.
As a first example, usbaudio is changed to use it now again.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The HP laptops with ALC268 codec seem working better with model=auto
than model=toshiba; e.g. the auto model fixes missing digital outputs.
Let's fix quirk entry to choose auto model explicitly.
Tested-by: Jens Jorgensen <jbj1@ultraemail.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix minimum period size for cs46xx cards. This fixes a problem in the
case where neither a period size nor a buffer size is passed to ALSA;
this is the case in Audacious, OpenAL, and others.
Signed-off-by: Sophie Hamilton <kernel@theblob.org>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
It's the 8th enum of a zero indexed array. This is why I don't let
new drivers use these arrays of enums...
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
The struct snd_monitor_file is used locally only in sound/core/init.c,
thus it should be moved there from the public sound/core.h.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the volume is changed continuously (e.g., when the user drags a
volume slider with the mouse), the driver does lots of I2C writes.
Apparently, the sound chip can get confused when we poll the I2C status
register too much, and fails to complete a read from it. On the PCI-E
models, the PCI-E/PCI bridge gets upset by this and generates a machine
check exception.
To avoid this, this patch replaces the polling with an unconditional
wait that is guaranteed to be long enough.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Tested-by: Johann Messner <johann.messner at jku.at>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of allocating the real buffers, use a fake buffer and ignore
read/write in the dummy driver so that we can save the resources.
For mmap, a single page (unique to the direction, though) is reused
to all buffers.
When the app requires to read/write the real buffers, pass fake_buffer=0
module option at loading time. This will get back to the old behavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The model clevo-m540r was created with 6-channel and digital support. All
functions verified except spdif. Tested with a VIT D2000 laptop which has:
[lspci extract]
Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio
Controller [8086:284b] (rev 03)
Subsystem: CLEVO/KAPOK Computer Device [1558:5409]
[/proc/asound/card0/codec\#0 header]
Codec: Realtek ALC883
Address: 0
Function Id: 0x1
Vendor Id: 0x10ec0883
Subsystem Id: 0x15585409
Revision Id: 0x100002
[Added a comment about HP mute and the model description by tiwai]
Signed-off-by: Dhionel Diaz <ddiaz@cenditel.gob.ve>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Today's linux-next fails to build with
sound/arm/pxa2xx-ac97.c: In function 'pxa2xx_ac97_probe':
sound/arm/pxa2xx-ac97.c:211: error: 'pxa2xx_audio_ops_t' has no member named 'codec_data'
make[2]: *** [sound/arm/pxa2xx-ac97.o] Error 1
It looks like commit e2365bf313 has
introduced this; patch below.
Signed-off-by: Robert Schwebel <r.schwebel@pengutronix.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Fix the expire-time calculation in the systimer mode when the buffer
size isn't aligned to the period size.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the system-timer mode, snd-dummy driver issues each tick to update
the position. This is highly inefficient and even inaccurate if the
timer can't be triggered at each tick.
Now rewritten to wake up only at the period boundary. The position
is calculated from the current jiffies.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow snd-dummy driver to use high-res timer as its timing source
instead of the system timer. The new module option "hrtimer" is added
to turn on/off the high-res timer support. It can be switched even
dynamically via sysfs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The card model detection code introduced in 2.6.30 that tries to work
around partially broken EEPROM contents by reading the EEPROM directly
does not handle cards where the EEPROM has been omitted. In this case,
we have to use the default ID to allow the driver to load.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-and-tested-by: Ozan Çağlayan <ozan@pardus.org.tr>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Avoids potential issues if we read back unexpected values during
a read/modify/write cycle.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Added the quirk for Alienware M17x with IDT 92HD73* codec chip.
It has two HP and one line-out jack, one mic jack, a built-in
speaker and a built-in mic.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the previous fix of input source for IDT92HD73xx, the amp mux
and amp vol stuff became unused. Let's rip off dead codes.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the mux_nids to select directly the input source instead of mux
mixers so that it works with the current mux enum handler for IDT
92HD73xx codecs.
Also, clean up useless / unnecessary mixer controls and init verbs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Bug was caught while trying to use WM8580 as I2S master on SMDK.
Symptoms were lesser LRCLK read by CRO(41.02 instead of 44.1 KHz) Solved
by referring to WM8580A manual and setting mask value correctly and
making the code to not touch 'reserved' bits of PLL4 register.
Signed-off-by: Jassi <jassi.brar@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
As discussed, the patch uses the original TDM order without rewriting.
For the match between TDM slot number and audio channel number, a new
API need be added.
Signed-off-by: Barry Song <21cnbao@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Unmute the docking-station line-out as default on machines with
AD1984A codec chip. It can be still muted via "Dock" mixer switch.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Do not forget to program the MCLK ratio for the I2S output.
Otherwise, the master clock frequency can be too high for
the DACs at sample frequencies above 96 kHz.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allocation may fail, show if it did.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
[Additional fix for invalid runtime->oss.prepare flag set by tiwai]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add the support of automatic mute and mic-switching of the docking
station HP and mic plugs for AD1984A laptop model for some HP machines.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fix the headphone pin routing of ALC268/ALC269 codecs. Using alc882
routine doesn't work because alc268/alc269 parser assumes the
independent DACs for both HP and speaker outputs. Need to assign the
DAC depending on the pin.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch fixes the wrong headphone output routing for MacBookPro 3,1/4,1
quirk with ALC889A codec, which caused the silent headphone output.
Also, this gives the individual Headphone and Speaker volume controls.
Reference: kernel bug#14078
http://bugzilla.kernel.org/show_bug.cgi?id=14078
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Cc: <stable@kernel.org>
In patch_vt1708(), the check of MUX nids is missing and this results in
the -EINVAL error in accessing Input Source mixer element. Simpliy
adding the call of get_mux_nids() fixes the problem.
Reference: Novell bnc#534904
https://bugzilla.novell.com/show_bug.cgi?id=534904
Signed-off-by: Takashi Iwai <tiwai@suse.de>
So far, the digital mic capture volume wasn't created. This is because
IDT codecs have output amps for digital mics, not input amps, while
input amps should be used for other analog pins. Thus the automatic
capture volume creation should check both directions for digital mics.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The McBSP1 port in OMAP3 processors (I believe OMAP2 too but I don't have
specifications to check it) have additional CLKR and FSR pins for McBSP1
receiver. Reset default is that receiver is using bit clock and frame
sync signal from those pins but it is possible to configure to use
also CLKX and FSX pins as well. In fact, other McBSP ports are doing that
internally that transmitter and receiver share the CLKX and FSX.
Add functionaly that machine drivers can set the CLKR and FSR sources by
using the snd_soc_dai_set_sysclk.
Thanks to "Aggarwal, Anuj" <anuj.aggarwal@ti.com> for reporting the issue.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Also, the codec setup data structure has to remain for successful
probe.
Signed-off-by: Chaithrika U S <chaithrika@ti.com>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Add debug module option to snd core.
This controls the debug print level. When CONFIG_SND_DEBUG_VERBOSE
is set, you can suppress the debug messages by giving or changing this
parameter to a lower value. debug=0 means no debug messsages.
As default, it's set to the verbose level 2.
Since this option can be changed dynamically via sysfs file, you can
suppress the verbose debug messages on the fly, which wasn't possible
before.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When modules are built with M= option, they pass long file paths to
__FILE__. This results in ugly outputs of snd_print*() when
CONFIG_SND_VERBOSE_PRINTK is set.
This patch adds a check of the path and strips the leading path dirs
if the file name is an absolute path to improve the readability of logs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the requested FLL configuration is the one we're currently running
in it's at best pointless to reconfigure the FLL.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Now that we don't need the I2C address for the device the platform data
is redundant so allow it to be omitted.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Tested-by: Chaithrika U S <chaithrika@ti.com>
No point in building them for S3C64xx, certainly no sense in running
into build issues there.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Since !LI_CCFG_* evaluates to 0, this did not change anything to
cfgval and ctlval.
Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- restructure to support multiple channel controllers by using
additional struct resources for each CC
- interface changes visible to EDMA clients
Introduce macros to build IDs from controller and channel number,
and to extract them. Modify the edma_alloc_slot function to take an
extra argument for the controller.
Also update ASoC drivers to use API. ASoC changes
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
- Move queue related mappings to dm<soc>.c
EDMA in DM355 and DM644x has two transfer controllers while DM646x
has four transfer controllers. Moving the queue to tc mapping and
queue priority mapping to dm<soc>.c will be helpful to probe these
mappings from platform device so that the machine_is_* testing will
be avoided.
- add channel mapping logic
Channel mapping logic is introduced in dm646x EDMA. This implies
that there is no fixed association for a channel number to a
parameter entry number. In other words, using the DMA channel
mapping registers (DCHMAPn), a PaRAM entry can be mapped to any
channel. While in the case of dm644x and dm355 there is a fixed
mapping between the EDMA channel and Param entry number.
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Sudhakar Rajashekhara <sudhakar.raj@ti.com>
Reviewed-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com>
Fix the write to PMBR1 register through I2C. Also, the constant which
holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This
name is based on TRM to avoid confusion.
Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
1) Added support of internal subwoofer (it sounds!!!)
2) Auto muting front speakers and internal subwoofer on headphones plug.
3) Internal mic works.
4) 3 channel mods (jack maps):
black pink blue
2ch: front ext mic line in
4ch: front ext mic surround
6ch: front CLFE surround
Can be changed in mixer.
5) Sound can be recorded from:
Internal mic
Ext mic
Cd
Line in
6) 2 separate capture channels.
Signed-off-by: Denis Kuplyakov <dener.kup@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
One more patch to give a better name for the primary output controls,
this time for ALC861-VD codec. The change is simple, just checking the
pin connection whether it's a speaker-out. When both speaker and HP
are assigned, we name the volume as "PCM" as this influences on both
outputs.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
