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* ALSA: pcm: fix delta calculation at boundary wraparoundClemens Ladisch2010-05-251-1/+3
| | | | | | | | | | | | | | | | | | | | | | | In the cleanup of the hw_ptr update functions in 2.6.33, the calculation of the delta value was changed to use the modulo operator to protect against a negative difference due to the pointer wrapping around at the boundary. However, the ptr variables are unsigned, so a negative difference would result in the two complement's value which has no relation to the actual difference relative to the boundary; the result is typically some value near LONG_MAX-boundary. Furthermore, even if the modulo operation would be done with signed types, the result of a negative dividend could be negative. The invalid delta value is then caught by the following checks, but this means that the pointer update is ignored. To fix this, use a range check as in the other pointer calculations. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb/caiaq: fix Traktor Kontrol X1 ABS_HAT2X axisDaniel Mack2010-05-251-1/+1
| | | | | Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: aw2-alsa.c: use pci_ids.h defines and fix checkpatch.pl noiseH Hartley Sweeten2010-05-251-7/+4
| | | | | | | | | | | | | Use the VENDOR/DEVICE ids provided in pci_ids.h instead of creating local ids of the same values. Also, fix the following checkpatch.pl warnings: WARNING: Use #include <linux/io.h> instead of <asm/io.h> WARNING: unnecessary whitespace before a quoted newline Signed-off-by: H Hartley Sweeten <hsweeten@visionengravers.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usb-audio: add support for Akai MPD16Krzysztof Foltman2010-05-215-0/+125
| | | | | | | | | The decoding/encoding is based on own reverse-engineering. Both control and data ports are handled. Writing to control port supports SysEx events only, as this is the only type of messages that MPD16 recognizes. Signed-off-by: Krzysztof Foltman <wdev@foltman.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: pcm: fix the fix of the runtime->boundary calculationClemens Ladisch2010-05-212-38/+10
| | | | | | | | | | | | | | | | | | | | | | | Commit 7910b4a1db63fefc3d291853d33c34c5b6352e8e in 2.6.34 changed the runtime->boundary calculation to make this value a multiple of both the buffer_size and the period_size, because the latter is assumed by the runtime->hw_ptr_interrupt calculation. However, due to the lack of a ioctl that could read the software parameters before they are set, the kernel requires that alsa-lib calculates the boundary value, too. The changed algorithm leads to a different boundary value used by alsa-lib, which makes, e.g., mplayer fail to play a 44.1 kHz file because the silence_size parameter is now invalid; bug report: <https://bugtrack.alsa-project.org/alsa-bug/view.php?id=5015>. This patch reverts the change to the boundary calculation, and instead fixes the hw_ptr_interrupt calculation to be period-aligned regardless of the boundary value. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'for-linus' of ↵Linus Torvalds2010-05-20248-8523/+36676
|\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (250 commits) ALSA: hda: Storage class should be before const qualifier ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT ASoC: sdp4430 - add sdp4430 pcm ops to DAI. ASoC: TWL6040: Enable earphone path in codec ASoC: SDP4430: Add support for Earphone speaker ASoC: SDP4430: Add sdp4430 machine driver ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function ALSA: sound/pci/asihpi: Use kzalloc ALSA: hdmi - dont fail on extra nodes ALSA: intelhdmi - add id for the CougarPoint chipset ALSA: intelhdmi - user friendly codec name ALSA: intelhdmi - add dependency on SND_DYNAMIC_MINORS ALSA: asihpi: incorrect range check ALSA: asihpi: testing the wrong variable ALSA: es1688: add pedantic range checks ARM: McBSP: Add support for omap4 in McBSP driver ARM: McBSP: Fix request for irq in OMAP4 OMAP: McBSP: Add 32-bit mode support ...
| * Merge branch 'topic/asoc' into for-linusTakashi Iwai2010-05-20137-2719/+10250
| |\ | | | | | | | | | | | | Conflicts: sound/soc/codecs/ad1938.c
| | * ASoC: tpa6130a2: Remove CPVSS and HPVdd suppliesJarkko Nikula2010-05-201-34/+18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | These pins are for decoupling capacitors for the internal charge pumps in TPA6130A2 and TPA6140A2 and not for connecting external supply. Thanks to Eduardo Valentin <eduardo.valentin@nokia.com> for pointing out the issue with TPA6130A2 and Ilkka Koskinen <ilkka.koskinen@nokia.com> with TPA6140A2. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Reviewed-by: Ilkka Koskinen <ilkka.koskinen@nokia.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUTJarkko Nikula2010-05-191-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | Codec output pin should be defined with SND_SOC_DAPM_OUTPUT as otherwise external widgets doesn't alter the output state. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: sdp4430 - add sdp4430 pcm ops to DAI.Liam Girdwood2010-05-191-0/+1
| | | | | | | | | | | | | | | | | | | | | Fix build warning about unused ops and add ops to the sdp4430 DAI link. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: TWL6040: Enable earphone path in codecJorge Eduardo Candelaria2010-05-191-1/+20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add control to enable earphone driver in TWL6040 codec. This driver is connected to HSDAC Left. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: SDP4430: Add support for Earphone speakerJorge Eduardo Candelaria2010-05-191-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | Enable earphone speaker in sdp4430 machine driver. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: SDP4430: Add sdp4430 machine driverMisael Lopez Cruz2010-05-193-0/+239
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add ASoC support for TI SDP4430. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com> Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: tlv320dac33: Avoid powering off while in BIAS_OFFPeter Ujfalusi2010-05-171-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Avoid calling the dac33_hard_power when the codec was already in BIAS_OFF state. This could happen in device suspend and module removal time. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power functionFelipe Balbi2010-05-171-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Since the cases when the same power state would be set again handled gracefully, we do not need to use dev_warn. Signed-off-by: Felipe Balbi <felipe.balbi@nokia.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ARM: McBSP: Add support for omap4 in McBSP driverJorge Eduardo Candelaria2010-05-141-7/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | McBSP module in OMAP4 needs to be able to set its tx/rx threshold and enable the transmitter/receiver when starting an audio stream. Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Signed-off-by: Margarita Olaya Cabrera <magi.olaya@ti.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ARM: McBSP: Fix request for irq in OMAP4Jorge Eduardo Candelaria2010-05-142-16/+16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In OMAP4, there is only one irq line for TX and RX paths. Use the correct irq line to avoid errors at runtime. Also, request irq line only once (instead of requesting for TX and RX). Signed-off-by: Jorge Eduardo Candelaria <jorge.candelaria@ti.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * OMAP: McBSP: Add 32-bit mode supportSergey Lapin2010-05-141-4/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patchs should allow to use 32-bit samples on e.g. TLV320AIC3x codec, or others. Signed-off-by: Sergey Lapin <slapin@ossfans.org> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: TWL4030: Add control for digimic Left Right swapPeter Ujfalusi2010-05-121-0/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The codec has support for swapping the left and right channels in the digimic interface. New kcontrol to handle this bit. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: Don't restart unconfigured WM8994 FLLsMark Brown2010-05-111-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | If the FLL is not configured attempting to resume it will produce a warning message so skip the resume. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: Reorder power down sequence for WM hubs devicesMark Brown2010-05-111-4/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | Disable the output stage prior to the delay stage rather than the other way around. Fixes merge issue with previous headphone output path corrections. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: Add additional WM hubs DC servo traceMark Brown2010-05-111-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | Log the values we're getting back from the DC servo and the values we write to it. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: Add register write logging for WM8994Mark Brown2010-05-111-0/+2
| | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: core: Fix for the volume limiting when invert is in usePeter Ujfalusi2010-05-112-20/+33
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If the register for the volume needs invert, than the inversion need to be done from the chip maximum, and not from the platform dependent limit. Introduce soc_mixer_control.platform_max value, which initially equals to chip maximum. The snd_soc_limit_volume function only modify the platform_max, all volsw_info call returns this as well. The .max value holds the chip default (maximum), and it is used for the inversion, if it is needed. Additional check in the volsw_info call has been added to check the validity of the platform_max in case, when custom macros used by codec drivers are not initializing it correctly. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Use more idiomatic driver name for WM8731Mark Brown2010-05-102-1/+31
| | | | | | | | | | | | | | | | | | | | | Make dev_() prints much prettier. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: Refactor WM8731 regulator management into bias managementMark Brown2010-05-101-23/+28
| | | | | | | | | | | | | | | | | | | | | This allows more flexible integration with subsystem features. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | * ASoC: Allow DAI links to be kept active over suspendMark Brown2010-05-102-1/+40
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As well as allowing DAPM pins to be marked as ignoring suspend allow DAI links to be similarly marked. This is primarily intended for digital links between CODECs and non-CPU devices such as basebands in mobile phones and will suppress all suspend calls for the DAI link. It is likely that this will need to be revisited if used with devices which are part of the SoC CPU. Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Allow active paths from the GSM modem while the GTA02 is suspendedMark Brown2010-05-101-0/+8
| | | | | | | | | | | | | | | Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Support leaving paths enabled over system suspendMark Brown2010-05-103-5/+60
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Some devices can usefully run audio while the Linux system is suspended. One of the most common examples is smartphone systems, which are normally designed to allow audio to be run between the baseband and the CODEC without passing through the CPU and so can suspend the CPU when on a voice call for additional power savings. Support such systems by providing an API snd_soc_dapm_ignore_suspend(). This can be used to mark DAPM endpoints as not being sensitive to system suspend. When the system is being suspended paths between endpoints which are marked as ignoring suspend will be kept active. Both source and sink must be marked, and there must already be an active path between the two endpoints prior to suspend. When paths are active over suspend the bias management will hold the device bias in the ON state. This is used to avoid suspending the CODEC while it is still in use. Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Refactor DAPM suspend handlingMark Brown2010-05-102-23/+33
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Instead of using stream events to handle power down during suspend integrate the handling with the normal widget path checking by replacing all cases where we report a connected endpoint in a path with a function snd_soc_dapm_suspend_check() which looks at the ALSA power state for the card and reports false if we are in a D3 state. Since the core moves us into D3 prior to initating the suspend all power checks during suspend will cause the widgets to be powered down. In order to ensure that widgets are powered up on resume set the card to D2 at the start of resume handling (ALSA API calls require D0 so we are still protected against userspace access). Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Remove unused DAPM suspend flagMark Brown2010-05-102-11/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | We now manage suspend within the main power analysis rather than by flipping the state of widgets. Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Remove unneeded suspend bias managment from CODEC driversMark Brown2010-05-1021-31/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The core will ensure that the device is in either STANDBY or OFF bias before suspending, restoring the bias in the driver is unneeded. Some drivers doing slightly more roundabout things have been left alone for now. Tested-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: SMDK64XX: Switch to IISv4 CPU driverJassi Brar2010-05-072-8/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Switch the MACHINE driver to use IISv4 CPU dai. Remove BROKEN dependency now that we have proper CPU driver available. Also, disable build for SMDK6400, since the S3C6400 doesn't have IISv4 controller. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: S3C64XX: IISv4: Add CPU driverJassi Brar2010-05-074-0/+217
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add the CPU driver for the IISv4 block found on S3C6410. For now, the driver is almost a copy of s3c64xx-i2s.c but it should diverge as more IISv4 specific stuff is added. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: tpa6130a2: Fix for the custom kcontrol functionsPeter Ujfalusi2010-05-071-8/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Since the functions arre only used for volume register, change their name, and also fix them to properly handle the cases, when via soc core the volume is limited. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * Revert "ASoC: tpa6130a2: Support for limiting gain"Peter Ujfalusi2010-05-072-69/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This reverts commit 6f3991152f20933b77eff30413e893bf1a15e578. Since core has now support for limiting the volume on controls this patch is not needed. Furthermore, this patch actually prevents the core to set new volume on the TPA. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: core: Support for limiting the volumePeter Ujfalusi2010-05-072-0/+41
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add support for the core to limit the maximum volume on an existing control. The function will modify the soc_mixer_control.max value of the given control. The new value must be lower than the original one (chip maximum) If there is a need for limiting a gain on a given control, than machine drivers can do the following in their snd_soc_dai_link.init function: snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21); This will modify the original 31 (chip maximum) to 21, so user space will not be able to set the gain higher than this. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * Merge branch 'topic/asoc' of ↵Mark Brown2010-05-0711-143/+619
| | |\ | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 into for-2.6.35
| | | * Merge branch 'for-2.6.35' of ↵Takashi Iwai2010-05-0611-143/+619
| | | |\ | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc
| | | | * ASoC: tlv320dac33: Use codec defaults for LOM/LOP and DAC powerPeter Ujfalusi2010-05-061-11/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Do not change the codec defaults for the following registers: 0x40, 0x41: Line output gains, do not use amplification 0x42: LOM/LOP Voltage hold, and selection 0x44: LOM inversion control It has been found, that the values configured to these registers can cause amplification, which can make the output of DAC33 distorted. The codec reset values are considered safe in all environmnts. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | | * ASoC: tpa6130a2: Support for limiting gainPeter Ujfalusi2010-05-062-8/+69
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add support for platform dependent gain limiting on the tpa6130a2 (and tpa6140a2) Headset amplifier. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | | * ASoC: tlv320aic3x: Add platform data and reset gpio handlingJarkko Nikula2010-05-062-0/+42
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Handle the reset GPIO within the codec driver in order to follow the startup protocol for the tlv320aic3x codecs. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | | * ASoC: omap: Add basic audio support for Nokia RX-51/N900Jarkko Nikula2010-05-063-0/+306
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds support for integrated stereo speakers and digital microphone found on Nokia RX-51 hardware. This is a cut down version based on Maemo kernel sources and earlier patchset by Eduardo Valentin et al. http://mailman.alsa-project.org/pipermail/alsa-devel/2009-October/022033.html Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Eduardo Valentin <eduardo.valentin@nokia.com> Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Eduardo Valentin <eduardo.valentin@nokia.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | | * ASoC: tpa6130a2: TLV mapping for tpa6140a2Peter Ujfalusi2010-05-041-3/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Both tpa6130a2, and tpa6140a2 is supported by the same driver, but the gain dB scaling is different on the amplifiers. Provide different mixer control for the chips with correct TLV mapping. User space will see: "TPA6130A2 Headphone Playback Volume" in case of 6130 "TPA6140A2 Headphone Playback Volume" in case of 6140 The way machine drivers are using this amplifier remained the same. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | | * ASoC: tlv320dac33: Support for turning off the codecPeter Ujfalusi2010-05-031-21/+45
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Let the codec to hit OFF instead of STANDBY, when there is no activity. When the codec is off, than the associated regulator can be also turned off (if the number of users on the regulator is 0). After initialization, the codec remains in power off, it is only turned on for reading the ID registers (also testing the regulators). The codec power is enabled, when the codec is moving from BIAS_OFF to BIAS_STANDBY. The codec is turned off, when it hits BIAS_OFF. There are few scenarios, which has to be taken care:: 1. Analog bypass caused BIAS_OFF -> BIAS_ON We need to power on the codec, and do the chip init, but we does not need to execute the playback related configuration 2. Playback caused BIAS_OFF -> BIAS_ON We need to power on the codec, and do the chip init, and also we need to execute the playback related configuration. 3. Playback start, while Analog bypass is on (BIAS_ON -> BIAS_ON) We need to execute the playback related configuration. The codec is already on. 4. Analog bypass enable, while playback (BIAS_ON -> BIAS_ON) Nothing need to be done. 5. Playback start withing soc power down timeout (BIAS_ON -> BIAS_ON) We need to execute the playback related configuration. The codec is still on. Since the power up, and the codec init is optimized, the added overhead in stream start is minimal. Withing this patch, the hard_power function is now only doing what it supposed to: only handle the powers, and GPIO reset line. The codec initialization and state restore has been moved out. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | | * ASoC: tlv320dac33: Manage a pointer for snd_pcm_substream in private structurePeter Ujfalusi2010-05-031-0/+28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As a preparation for supporting codec to be turned off, when we are in BIAS_STANDBY. The substream must be easily available in other places than pcm_* callbacks. Manage a pointer in _startup, and _shutdown for this. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | | * ASoC: tlv320dac33: Revised module loading, and DAC33 ID readPeter Ujfalusi2010-05-031-19/+18
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Optimize the way how tlv320dac33 is powered uppon module and soc initialization. Also read the DAC33 ID registers, and update the reg_cache to reflect it. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | | * ASoC: tlv320dac33: Optimize power up, and restorePeter Ujfalusi2010-05-031-67/+39
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | On power up we only need to initialize the codec, and restore only registers, which are not in either in DAPM nor in the playback start sequence. These are mostly gain related registers. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | | * ASoC: TWL4030: Remove OUTL/R outputsPeter Ujfalusi2010-05-031-4/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | OUTL/R are leftovers from the original driver, and they are no longer needed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| | | | * ASoC: TWL4030: AIF/APLL fix in DAPM domainPeter Ujfalusi2010-05-031-22/+60
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch orders the APLL and AIF power sequence in case of HiFi (audio in TWL4030 terms) playback/capture. We also need to make sure that the AIF is running during playback/capture, when there is no valid DAPM route available. For this purpose I introduce these virtual widgets: /* To have complete playback route all the time */ DAPM_OUTPUT("Virtual HiFi OUT") /* Will keep AIF/APLL enabled */ /* To have complete capture route all the time */ DAPM_INPUT("Virtual HiFi IN") /* Will keep AIF/APLL enabled */ /* To have complete playback route for the voice module */ DAPM_OUTPUT("Virtual Voice OUT") /* Will keep APLL enabled */ The DAPM_SUPPLY widgets for APLL and AIF are placed in a way, that during any audio activity the needed configuration of AIF and APLL will be enabled (playback, capture, analog loopback, digital loopback, and voice activity). The apll reference counting code has been lifted, and modified from Liam Girdwood's earlier patch. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-24 09:40:45 +0100