| Age | Commit message (Collapse) | Author |
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Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Smart 5.1 is for 3-jacks model, to reuse input pins as outputs.
While off, they act as "line out" / "line in" / "mic in".
While on, they acts as "line out" / "back left/right" / "center/lfe".
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Use hp_independent_mode_index to store hp index, and simplify function
via_independent_hp_put with it.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For VT1708S and VT1702, deactivate "Headphone Playback Volume" and
"Headphone Playback Mute" control if "Independent HP" mode is OFF.
and rename VT1702 "Independent HP" text.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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For VT1708B, VT1708S and VT1702, enter low current mode if no analog
stream is opened and all aa path mute.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Enter low power state if AA-Path volume is muted.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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according to customer request, VT1702 AA-Path max volume (12 dB) is too
high, so limit to 0 dB.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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IS_VT17*_VENDORID macros are used nowhere, so clean them up.
Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Logan Li <loganli@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Convert CS4231 mixer to dB scale after AD1848 mixer.
Also, add missing microphone boost control for the AD1848
and correct wrong bits for loopback volume on the AD1848.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix coding style errors in the driver.
Also, add missing argument for CMD_XXX_MIDI_VOL command.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix the num_total_dacs setting for Chaintech AV710. The existing comment
that only PSDOUT0 is connected is correct, but since the card is using
packed AC97 mode to send 6 channels to the codec, num_total_dacs should be
set to 6 and not 2. This allows 6-channel surround to work. Also clarify
a comment regarding the additional WM8728 codec on this card (it's connected
to the SPDIF output and always receives the same data).
Signed-off-by: Robert Hancock <hancockrwd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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- Staticise ttpa6130a2_client.
- Remove unneeded cast from void.
- Use explict NULL rather than 0.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Driver for Texas Instruments TPA6130A2 stereo headphone
amplifier.
The driver provides playback gain control and also pre-defined
DAPM_HP widgets and DAPM routings for power management.
The DAPM_HP widget names are:
"TPA6130A2 Headphone Left"
"TPA6130A2 Headphone Right"
From soc machine drivers to use with the tpa6130a2 amplifier,
the tpa6130a2_add_controls has to be called, which adds the alsa
controls and the DAPM routing needed for the tpa6130a2.
After that the machine driver can connect the codec's output
with 'TPA6130A2 Left' and 'TPA6130A2 Right':
{"TPA6130A2 Left", NULL, "CODEC LEFT OUT"},
{"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"},
Internally the left and right channels are powered separately.
When none of the channels are needed the amplifier is powered
down:
hard power: valid GPIO number is passed within platform data
soft power: Using the software shutdown of the amplifier
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Allow Nvidia HDMI to support more possible sample rates and formats.
At best, the really supported rates and formats should be determined
together with the negotiation with the HDMI receiver, but it's currently
not implemented yet (Nvidia stuff seems incompatible with HDMI 1.3
standard in this regard). As a compromise, we enable all bits, assuming
that all recent devices do support such rates/formats.
Tested-by: Alan Alan <alanwww1@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The modified revision of at91sam9g20 Evaluation Kit rev. C and onwards share
with previous ones its audio connexion to Wolfson wm8731. Modify the SoC file
to extend the machine ID checking.
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Increase the default and maximum PCM buffer prellocation size for ice1724's
SPDIF and independent stereo pair outputs to 256K, which is the hardware's
maximum supported size. This allows a reduction in interrupt rate and
potentially power usage when an application is not latency-critical.
Signed-off-by: Robert Hancock <hancockrwd@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Fix following circular locking in the opl3 driver.
=======================================================
[ INFO: possible circular locking dependency detected ]
2.6.32-rc3 #87
-------------------------------------------------------
swapper/0 is trying to acquire lock:
(&opl3->voice_lock){..-...}, at: [<cca748fe>] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth]
but task is already holding lock:
(&opl3->sys_timer_lock){..-...}, at: [<cca75169>] snd_opl3_timer_func+0x19/0xc0 [snd_opl3_synth]
which lock already depends on the new lock.
the existing dependency chain (in reverse order) is:
-> #1 (&opl3->sys_timer_lock){..-...}:
[<c02461d5>] validate_chain+0xa25/0x1040
[<c0246aca>] __lock_acquire+0x2da/0xab0
[<c024731a>] lock_acquire+0x7a/0xa0
[<c044c300>] _spin_lock_irqsave+0x40/0x60
[<cca75046>] snd_opl3_note_on+0x686/0x790 [snd_opl3_synth]
[<cca68912>] snd_midi_process_event+0x322/0x590 [snd_seq_midi_emul]
[<cca74245>] snd_opl3_synth_event_input+0x15/0x20 [snd_opl3_synth]
[<cca4dcc0>] snd_seq_deliver_single_event+0x100/0x200 [snd_seq]
[<cca4de07>] snd_seq_deliver_event+0x47/0x1f0 [snd_seq]
[<cca4e50b>] snd_seq_dispatch_event+0x3b/0x140 [snd_seq]
[<cca5008c>] snd_seq_check_queue+0x10c/0x120 [snd_seq]
[<cca5037b>] snd_seq_enqueue_event+0x6b/0xe0 [snd_seq]
[<cca4e0fd>] snd_seq_client_enqueue_event+0xdd/0x100 [snd_seq]
[<cca4eb7a>] snd_seq_write+0xea/0x190 [snd_seq]
[<c02827b6>] vfs_write+0x96/0x160
[<c0282c9d>] sys_write+0x3d/0x70
[<c0202c45>] syscall_call+0x7/0xb
-> #0 (&opl3->voice_lock){..-...}:
[<c02467e6>] validate_chain+0x1036/0x1040
[<c0246aca>] __lock_acquire+0x2da/0xab0
[<c024731a>] lock_acquire+0x7a/0xa0
[<c044c300>] _spin_lock_irqsave+0x40/0x60
[<cca748fe>] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth]
[<cca751f0>] snd_opl3_timer_func+0xa0/0xc0 [snd_opl3_synth]
[<c022ac46>] run_timer_softirq+0x166/0x1e0
[<c02269e8>] __do_softirq+0x78/0x110
[<c0226ac6>] do_softirq+0x46/0x50
[<c0226e26>] irq_exit+0x36/0x40
[<c0204bd2>] do_IRQ+0x42/0xb0
[<c020328e>] common_interrupt+0x2e/0x40
[<c021092f>] apm_cpu_idle+0x10f/0x290
[<c0201b11>] cpu_idle+0x21/0x40
[<c04443cd>] rest_init+0x4d/0x60
[<c055c835>] start_kernel+0x235/0x280
[<c055c066>] i386_start_kernel+0x66/0x70
other info that might help us debug this:
2 locks held by swapper/0:
#0: (&opl3->tlist){+.-...}, at: [<c022abd0>] run_timer_softirq+0xf0/0x1e0
#1: (&opl3->sys_timer_lock){..-...}, at: [<cca75169>] snd_opl3_timer_func+0x19/0xc0 [snd_opl3_synth]
stack backtrace:
Pid: 0, comm: swapper Not tainted 2.6.32-rc3 #87
Call Trace:
[<c0245188>] print_circular_bug+0xc8/0xd0
[<c02467e6>] validate_chain+0x1036/0x1040
[<c0247f14>] ? check_usage_forwards+0x54/0xd0
[<c0246aca>] __lock_acquire+0x2da/0xab0
[<c024731a>] lock_acquire+0x7a/0xa0
[<cca748fe>] ? snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth]
[<c044c300>] _spin_lock_irqsave+0x40/0x60
[<cca748fe>] ? snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth]
[<cca748fe>] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth]
[<c044c307>] ? _spin_lock_irqsave+0x47/0x60
[<cca751f0>] snd_opl3_timer_func+0xa0/0xc0 [snd_opl3_synth]
[<c022ac46>] run_timer_softirq+0x166/0x1e0
[<c022abd0>] ? run_timer_softirq+0xf0/0x1e0
[<cca75150>] ? snd_opl3_timer_func+0x0/0xc0 [snd_opl3_synth]
[<c02269e8>] __do_softirq+0x78/0x110
[<c044c0fd>] ? _spin_unlock+0x1d/0x20
[<c025915f>] ? handle_level_irq+0xaf/0xe0
[<c0226ac6>] do_softirq+0x46/0x50
[<c0226e26>] irq_exit+0x36/0x40
[<c0204bd2>] do_IRQ+0x42/0xb0
[<c024463c>] ? trace_hardirqs_on_caller+0x12c/0x180
[<c020328e>] common_interrupt+0x2e/0x40
[<c0208d88>] ? default_idle+0x38/0x50
[<c021092f>] apm_cpu_idle+0x10f/0x290
[<c0201b11>] cpu_idle+0x21/0x40
[<c04443cd>] rest_init+0x4d/0x60
[<c055c835>] start_kernel+0x235/0x280
[<c055c210>] ? unknown_bootoption+0x0/0x210
[<c055c066>] i386_start_kernel+0x66/0x70
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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MIXER to PCM type
* PLEASE NOTE - this change requires the corresponding update of
envy24control for ice1712 - kind of an ABI change.
* The "Multi Track Peak" control is read-only level meters indicator.
* The control is VERY confusing to most users since it is currently displayed
in regular mixers. E.g. alsamixer ignores its read-only status
and allows changing the levels with keys which makes no sense.
Signed-off-by: Pavel Hofman <pavel.hofman@ivitera.com>
Acked-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since patch_alc268() doesn't call set_capture_mixer() (due to its h/w
design different from other siblings), it needs to call fixup_automic_adc()
explicitly to set up the auto-mic routing. Otherwise the indices for
int/ext mics aren't set properly.
Reference: Novell bnc#544899
http://bugzilla.novell.com/show_bug.cgi?id=544899
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
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These should be handled via set_tdm_slot() now and cause build
failures as-is.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Sometimes it is desirable to have a mux which does not reflect any
direct register configuration but which will instead only have an
effect implicitly (for example, as a result of changing which parts
of the device are powered up). Provide a virtual mux for this purpose.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The "VIA DXS" controls are actually volume controls that apply to the
four PCM substreams, so we better indicate this connection by moving the
controls to the PCM interface.
Commit b452e08e73c0e3dbb0be82130217be4b7084299e in 2.6.30 broke the
restoring of these volumes by "alsactl restore" that most distributions
use; the renaming in this patch cures that regression by preventing
alsactl from applying the old, wrong volume levels to the new controls.
http://bugzilla.kernel.org/show_bug.cgi?id=14151
http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=532613
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Don't assume that enumerations are backed by registers when updating
mux power.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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We don't need to check for an event callback since we also check for
an appropriate event flag when applying mux status changes.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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alc_subsystem_id() tries to pick up a headphone pin if not configured,
but this caused side-effects as the problem in commit
15870f05e90a365f8022da416e713be0c5024e2f.
This patch fixes the driver behavior to pick up invalid HP pins; at least,
the pins that are listed as the primary outputs aren't taken any more.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into for-2.6.32
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ASUS A7K needs additional GPIO1 bit setup; it has to be cleared.
Added a new fixup hook for this laptop so that it works as is.
Refernece: Novell bnc#494309
http://bugzilla.novell.com/show_bug.cgi?id=494309
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Module parameters shouldn't be marked as __devinitdata since they can be
referred via sysfs even after probing.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Old Soundscape cards (pre PnP) work only with AD1848 codecs.
If the CS4231 codec is installed it must be used in AD1848
compatible mode.
Also, add gameport support and remove an unused mpu field.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The recent auto-parser doesn't work for machines with a single output
with ALC861, such as Toshiba laptops, because alc_subsystem_id() sets
the hp_pins[0] while it's listed in line_outs[0].
This ends up with the doubled initialization of the same mixer widget,
and it mutes the DAC route because hp_pins has no DAC assigned.
To fix this problem, just check spec->autocfg.hp_outs and speaker_outs
so that they are really detected pins.
Reference: Novell bnc#544161
http://bugzilla.novell.com/show_bug.cgi?id=544161
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There is no sense to limit open MIDI connections with limit
as high as ULONG_MAX.
Also, convert more messages to use the snd_printk.
Correct few old and misleading comments as well.
Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of ssh://master.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (21 commits)
ALSA: usb - Use strlcat() correctly
ALSA: Fix invalid __exit in sound/mips/*.c
ALSA: hda - Fix / improve ALC66x parser
ALSA: ctxfi: Swapped SURROUND-SIDE mute
sound: Make keywest_driver static
ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-B1VP
ALSA: hda - Fix digita/analog mic auto-switching with IDT codecs
ASoC: fix kconfig order of Blackfin drivers
ALSA: hda - Added quirk to enable sound on Toshiba NB200
ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2
ALSA: Don't assume i2c device probing always succeeds
ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-T350P
ALSA: echoaudio - Re-enable the line-out control for the Mia card
ALSA: hda - Resurrect input-source mixer of ALC268 model=acer
ALSA: hda - Analog Devices AD1984A add HP Touchsmart model
ALSA: hda - Add HP Pavilion dv4t-1300 to MSI whitelist
ALSA: hda - CD-audio sound for hda-intel conexant benq laptop
ASoC: DaVinci: Correct McASP FIFO initialization
ASoC: Davinci: Fix race with cpu_dai->dma_data
ASoC: DaVinci: Fix divide by zero error during 1st execution
...
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Fix for typo in commit 8d50e447d19fec64adebeef55f2b60d695435412
ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs
Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Don't pass the advanced position to strlcat() but just gives the buffer
head position so that the max size limit can be checked correctly.
Introduced a new helper function to standaralize strlcat() calls.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Change the way the debugfs entries are created:
If the codec->dev is valid, than use:
debugfs/asoc/{codec->name}.{dev_name(codec->dev)}/
if the codec->dev is NULL:
debugfs/asoc/{codec->name}/
as root for the debugfs entries.
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Since the SND_SOC_DAPM_LINE can be input or output, additional check is
needed in order to determine if the widget is connected as input or
output.
When checking for connected outputs, if the widget is line, than check
if the sources list is not empty (line is connected as output)
For input endpoint check, when the widget is line, also check if the
sinks list is not empty (line is connected as input).
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The remove callback has to be marked as __devexit, as the dynamic unbind
is possible.
Reported-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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