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* ASoC: twl6040: Return -ENOMEM if create_singlethread_workqueue failsAxel Lin2011-03-261-1/+3
| | | | | | Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
* ASoC: tlv320dac33: Restore L/R DAC power control registerPeter Ujfalusi2011-03-261-0/+4
| | | | | | | | | Register 0x40, 0x41 need to be restored after power up, since it contains gain related fields, which affects playback volume. 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: Fix inconsistent spinlock usagePeter Ujfalusi2011-03-221-12/+16
| | | | | | | | | | The lock is used within the interrupt handler. Correct the spinlock usage, and use irqsave/irqrestore flavour of spin_lock/unlock. 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: omap: rx51: Enable McBSP2 sidetoneJarkko Nikula2011-03-091-0/+4
| | | | | | | | | | | | | | | | | | | | | McBSP sidetone is needed in telephony applications. McBSP sidetone is a configurable FIR filter that forms a loopback from McBSP input to output. This patch enables the McBSP2 sidetone ALSA controls so that it can be used on Nokia RX-51/N900. Sidetone feature can be tested with following commands: (set up codec input and output paths) # Enable and configure sidetone amixer -D hw:0 set 'McBSP2 Sidetone' on amixer set -D hw:0 'McBSP2 Sidetone Channel 0' 32767 echo 32767 >/sys/devices/platform/omap-mcbsp.2/st_taps # Do not loop audio via CPU arecord -f dat >/dev/null |aplay /dev/zero Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
* ASoC: omap: rx51: Add FM transmitter supportJarkko Nikula2011-02-221-0/+4
| | | | | | | | | | | | | Si4713 FM transmitter on Nokia RX-51/N900 is connected to same Line out signals of TLV320AIC34 than TPA6130 headphone amplifier. This patch adds route to transmitter and "FM Transmitter" control to keep route active when needed. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> 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: omap: rx51: Report headset insertion instead of video out cableJarkko Nikula2011-02-151-2/+3
| | | | | | | | | | | It is more usefull to report headset instead of video out cable in response to jack insertion as this is more usual use-case and because now the headset feature is supported. Automatic accessory detection is not possible at the moment so most sensible static accessory type have to be used. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
* ASoC: omap: rx51: Add headset supportJarkko Nikula2011-02-151-2/+23
| | | | | | | | | | | | | | | | | This patch adds support for headset microphone in Nokia RX-51/N900. The mic signal from audio jack is routed to codec A LINE1L via two switches and the mic bias is coming from codec B part. First switch is the tv-out switch that is already supported and the second switch selects between voltage detection circuit and codecs. As there is no use for voltage detection at the moment the second switch is connected statically to codecs in rx51_soc_init. Headset can be active when control "Jack Function" is set to "Headset". Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
* ASoC: omap: rx51: Use gpio_request_one to configure tvout_sel gpioJarkko Nikula2011-02-151-2/+2
| | | | | | | | Just slight cleanup to be sync with upcoming change. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
* Merge branch 'for-2.6.39' of ↵Mark Brown2011-02-013-6/+92
|\ | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into for-2.6.39
| * ASoC: omap: rx51: Add earphone supportJarkko Nikula2011-02-011-0/+50
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Earphone in Nokia RX-51/N900 is connected to left HP output of B part of the TLV320AIC34 dual codec. In RX-51 the codec A is used as a traditional codec and the codec B as an auxiliary device. Audio from codec A goes via the codec B to earphone: MONO_LOUT of A -> LINE2R of B (B interconnects) -> HPLOUT of B -> Earphone. Take earphone into use by utilizing the recent ASoC auxiliary and cross-device support. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * ASoC: omap: rx51: Add stereo output support to audio jackJarkko Nikula2011-01-272-4/+40
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Audio jack in Nokia RX-51/N900 is driven by TPA6130 headphone amplifier. This patch adds support for it and stereo output can be active when "Jack Function" == "TV-OUT" || "Headphone". As the TPA6130 can output very high volume levels the output is limited with snd_soc_limit_volume. Limiting value is found from Maemo kernel sources. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
| * ASoC: Staticise twl6040_hs_jack_report()Mark Brown2011-01-211-2/+2
| | | | | | | | | | | | | | | | It's an internal function so shouldn't be exported (as sparse points out). Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | ASoC: Use snd_pcm_format_width() in snd_soc_params_to_frame_size()Mark Brown2011-02-011-20/+3
| | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | ASoC: soc-core: Ensure codec_reg has fixed length fieldsDimitris Papastamos2011-02-011-3/+24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Make the format of the codec_reg file more easily parsable. Remove the header field which gives the codec name. These changes are important when it comes to extend the debugfs codec_reg file to dump more than PAGE_SIZE bytes to make it easier to calculate offsets within the file. We still need to handle the case when the snd_soc_read() call fails and <no data: %d> is outputted. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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*-. \ Merge branches 'for-2.6.38' and 'tegra-arch' into for-2.6.39Mark Brown2011-02-012-2/+3
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| * | | ASoC: CX20442: fix NULL pointer dereferenceJanusz Krzysztofik2011-02-011-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The CX20442 codec driver never provided the snd_soc_codec_driver's .reg_cache_default member. With the latest ASoC framework changes, it seems to be referred unconditionally, resulting in a NULL pointer dereference if missing. Provide it. Created and tested on Amstrad Delta against linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: Amstrad Delta: fix const related build errorJanusz Krzysztofik2011-02-011-2/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Amstrad Delta ASoC driver used to override the digital_mute() callback, expected to be not provided by the on-board CX20442 CODEC driver, with its own implementation. While this is still posssible when substituting the whole empty snd_soc_dai_driver.ops member (the CX20442 case), replacing snd_soc_dai_ops.digital_mute only is no longer correct after the snd_soc_dai_driver.ops member has been constified, and results in build error. Drop this actually not used code path in hope the CX20442 driver never provides its own snd_soc_dai_ops structure. Created and tested against linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Tegra: Harmony: Remove redundant !!Stephen Warren2011-02-011-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | gpio_set_value* should accept logic values not just 0 or 1. The WM8903 GPIO driver has been fixed to work this way, so remove the redundant !! previously required when it didn't accept values >1. Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Accept any logical value WM8903 GPIO set()Mark Brown2011-01-311-1/+2
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | | ASoC: Accept any logical value for WM8962 GPIO set()Mark Brown2011-01-311-1/+1
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | | ASoC: neo1973_wm8753 audio support does not require scoopH Hartley Sweeten2011-01-311-1/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This driver does not use any of the functionality provided by the scoop hardware. Remove the unneeded header. Signed-off-by: H Hartley Sweeten <hsweeten@visionengravers.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Tegra: I2S: Use dev_err not pr_errStephen Warren2011-01-311-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Tegra: utils: Don't use global variablesStephen Warren2011-01-313-54/+68
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Instead, have the machine driver provide storage for the utility data somehow. For Harmony in particular, store this within struct tegra_harmony, itself referenced by snd_soc_card's drvdata. Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Tegra: Harmony: Use dev_err not pr_errStephen Warren2011-01-311-5/+6
| | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Tegra: Harmony: Fix indentation issue.Stephen Warren2011-01-311-5/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Indent with TABs not spaces. Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Tegra: Harmony: Support the internal speakerStephen Warren2011-01-311-0/+51
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add DAPM widget definitions for the internal speaker paths. Currently, this path is always enabled while playback is active. Add code to control the speaker amplifier GPIO. The GPIO is requested during _init, since that's the first time it is guaranteed that the WM8903 module is loaded, probed, and hence has exported its GPIO chip. Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Tegra: Harmony: Don't use soc-audio platform deviceStephen Warren2011-01-311-28/+66
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Previously, snd-soc-tegra-harmony internally instantiated a platform device object whenever the module was loaded. Instead, switch to a more typical model where arch/arm/mach-tegra defines a platform device, and snd-soc-tegra-harmony acts as a driver for such a platform device. Define a new struct tegra_harmony to store driver data in the future. Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Move card list initialization to snd_soc_register_cardStephen Warren2011-01-311-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | All ASoC cards need snd_soc_initialize_card_lists called. Previously, it was only called for cards backed by a "soc-audio" platform device, via soc_probe(). However, it's also needed for cards backed by other platform devices, and registered directly via snd_soc_register_card(). Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: mid-x86: Add support for capture in machine driverHarsha Priya2011-01-311-0/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This configures the capture unused pins Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Harsha Priya <priya.harsha@intel.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: sst_platform: add support for capture stream on headset daiHarsha Priya2011-01-311-0/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Harsha Priya <priya.harsha@intel.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: sn95031: add capture supportHarsha Priya2011-01-311-0/+281
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch adds the support for capture path in sn95031 codec. This codec supports upto 6DMICs, 2 AMICs and Linein. The linein and AMICs are connected through a MUX to ADC. The TX paths can be assigned to any of the ADCs or DMICs. Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Harsha Priya <priya.harsha@intel.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | Merge branch 'for-2.6.38' into for-2.6.39Mark Brown2011-01-282-4/+4
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| * | | ASoC: Fix mask/val_mask confusion snd_soc_dapm_put_volsw()Stephen Warren2011-01-281-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_soc_dapm_put_volsw() has variables for both the unshifted and shifted mask for updates commit 97404f (ASoC: Do DAPM control updates in the middle of DAPM sequences) got confused between the two of these. Since there's no need to keep a copy of the unshifted mask fix this and simplify the code by using only one mask variable. [Completely rewrote the changelog to describe the issue -- broonie.] Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | | ASoC: DaVinci: fix kernel panic due to uninitialized platform_dataManjunathappa, Prakash2011-01-281-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch fixes the Kernel panic issue on accessing davinci_vc in cq93vc_probe function. struct davinci_vc is part of platform device's private driver data(codec->dev->p->driver_data) and this is populated by DaVinci Voice Codec MFD driver. Signed-off-by: Manjunathappa, Prakash <prakash.pm@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: soc-core: Increment codec and platform driver refcounts before probingJarkko Nikula2011-01-281-8/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers" moved codec and platform driver refcount increments from soc_bind_dai_link to more appropriate places. Adjust a little them so that refcounts are incremented before executing the driver probe functions. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Vinod Koul <vinod.koul@intel.com> Cc: Harsha Priya <priya.harsha@intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | Merge branch 'for-2.6.38' into for-2.6.39Mark Brown2011-01-271-3/+0
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| * | | ASoC: Fix module refcount for auxiliary devicesJarkko Nikula2011-01-261-3/+0
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers" moved codec driver refcount increments from soc_bind_dai_link into soc_probe_codec. However, the commit didn't remove try_module_get from soc_probe_aux_dev so the auxiliary device reference counts are incremented twice as the soc_probe_codec is called from soc_probe_aux_dev too. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: Add subsequence information to seq_notify callbacksMark Brown2011-01-271-2/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Allows drivers to distinguish which subsequence is being notified when they get called back. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | | ASoC: Make cache status available via debugfsMark Brown2011-01-271-0/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Could just as well live in sysfs but sysfs doesn't have the simple value export helpers debugfs does. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | | ASoC: Export card PM callbacks for use in direct registered cardsMark Brown2011-01-271-17/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Allow hookup of cards registered directly with the core to the PM operations by exporting the device power management operations to modules, also exporting the default PM operations since it is expected that most cards will end up using exactly the same setup. Note that the callbacks require that the driver data for the card be the snd_soc_card. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | | ASoC: Replace pdev with card in machine driver probe and removeMark Brown2011-01-275-17/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In order to support cards instantiated without using soc-audio remove the use of the platform device in the card probe() and remove() ops. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | | ASoC: Use card rather than soc-audio device to card PM functionsMark Brown2011-01-273-10/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The platform device for the card is tied closely to the soc-audio implementation which we're currently trying to remove in favour of allowing cards to have their own devices. Begin removing it by replacing it with the card in the suspend and resume callbacks we give to cards, also taking the opportunity to remove the legacy suspend types which are currently hard coded anyway. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | | ASoC: Staticise non-exported symbols in cs4271Mark Brown2011-01-261-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
* | | | ASoC: sh: fsi: modify selection method of I2S/PCM/SPDIF formatKuninori Morimoto2011-01-263-53/+59
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Current format selection of FSI-codecs depended on platform information for FSI, and chip default settings for codecs. It is not understandable/formal method. This patch modify FSI and FSI-codecs to use snd_soc_dai_set_fmt. But FSI can use I2S/PCM and SPDIF format today. It can be selected to I2S/PCM by snd_soc_dai_set_fmt, but can not select SPDIF. So, this patch change FSI platform information to have DAI/SPDIF mode. If platform selects DAI mode (default), FSI-codecs can select I2S/PCM by snd_soc_dai_set_fmt, and if it is SPDIF mode, FSI become SPDIF format. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Paul Mundt <lethal@linux-sh.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: sh: fsi: free from NULL pointer of struct sh_fsi_platform_infoKuninori Morimoto2011-01-261-4/+17
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Current FSI driver assumed master->info is not NULL. This patch allow NULL in master->info Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Paul Mundt <lethal@linux-sh.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | ASoC: sh: fsi: move chan_num from fsi_stream to fsi_privKuninori Morimoto2011-01-261-19/+15
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Paul Mundt <lethal@linux-sh.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | Merge branch 'for-2.6.38' into for-2.6.39Mark Brown2011-01-2510-18/+18
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| * | | ASoC: correct link specifications for corgi, poodle and spitzDmitry Eremin-Solenikov2011-01-253-5/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ASoC DAI link descriptions for Corgi, Poodle and Spitz platforms contained incorrect names for cpu_dai and codec, which effectievly disabled sound on theese platforms. Fix that errors. Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| * | | ASoC: Samsung: Fix outdated cpu_dai_name for s3c24xx i2sLars-Peter Clausen2011-01-255-5/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | During the multi-component patch the s3c24xx i2s driver was renamed from "s3c24xx-i2s" to "s3c24xx-iis", while the sound board drivers were not updated to reflect this change as well. As a result there is no match between the dai_link and the i2s driver and no sound card is instantiated. This patch fixes the problem by updating the sound board drivers to use "s3c24xx-iis" for the cpu_dai_name. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| * | | ASoC: Fix codec device id format used by some dai_linksLars-Peter Clausen2011-01-256-8/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The id part of an I2C device name is created with the "%d-%04x" format string. So for example for an I2C device which is connected to the adapter with the id 0 and has its address set to 0x1a the id part of the devices name would be "0-001a". Currently some sound board drivers have the id part the codec_name field of their dai_link structures set as if it had been created by a "%d-0x%x" format string. For example "0-0x1a" instead of "0-001a". As a result there is no match between the codec device and the dai_link and no sound card is instantiated. This patch fixes it. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
generated by cgit v1.1 at 2024-12-25 03:03:36 +0000