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* ASoC: Add support for CS42L52 CodecBrian Austin2012-05-014-0/+1582
| | | | | | | | This patch adds support for Cirrus Logic CS42L52 Low Power Stereo Codec Signed-off-by: Brian Austin <brian.austin@cirrus.com> Signed-off-by: Georgi Vlaev <joe@nucleusys.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: core: Fix dai_link dereference.Liam Girdwood2012-04-301-1/+1
| | | | | | | | We should check dailess before dereferencing. Reported-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: fsl: add sgtl5000 clock support for imx-sgtl5000Richard Zhao2012-04-271-8/+32
| | | | | | | | It tries to clk_get the clock. And if it failed, it assumes the clock by default enabled. Signed-off-by: Richard Zhao <richard.zhao@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: imx-sgtl5000: add of_node_put when probe fail.Richard Zhao2012-04-271-11/+18
| | | | | Signed-off-by: Richard Zhao <richard.zhao@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm_hubs: Enable class W for output mixer pathsMark Brown2012-04-271-21/+40
| | | | | | | | | | | | | | | | | Class W can be used for any path where only data from the DAC is routed to the headphones. Currently we only enable it when the direct DAC to headphone path is used but it can also be enabled for paths that go via the output mixer providing the DAC is the only input to the output mixer. Implement support for this, including updates to the class W status when the output mixer configuration is changed. This also allows us to enable the DC servo optimisations for DAC to headphone paths where the output mixer is used. In general the direct DAC path is still preferred as this will offer better performance on most wm_hubs devices but these additional paths can simplify use case management. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm_hubs: Factor out class W managementMark Brown2012-04-274-158/+82
| | | | | | | | | Since the analogue portions of the checks for class W are the same over all the devices factor out these checks into wm_hubs and while we're at it also use wm_hubs_dac_hp_direct() to enable class W optimisations on more paths. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm_hubs: Special case headphones for digital paths in more use casesMark Brown2012-04-274-14/+45
| | | | | | | | | | | | | The optimisations which we can do with caching the headphone DCS result in wm_hubs have only been enabled in cases where class W is enabled. However, there are more use cases which can benefit from the cache, especially with WM8994 series devices with their more advanced digital routing. Rather than keying off the class W information from the CODECs have a check in wm_hubs for a suitable path and use that to determine if we can deploy our headphone optimisations. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: dpcm: Fixup debugFS for DPCM state.Liam Girdwood2012-04-271-12/+5
| | | | | | | | Remove writable debugFS permission, use simple_open() and fix indentation. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: da7210: Minor bugfix for non pll slave modeAshish Chavan2012-04-271-6/+7
| | | | | | | | | | | This patch fixes a bug discovered during testing of non pll slave mode. Due to the bug chip was not getting correctly configured and as a result there was no sound output while playback. After applying this patch, both pll and non pll modes work fine. Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com> Signed-off-by: David Dajun Chen <dchen@diasemi.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: dapm: Move CODEC<->CODEC params off stackMark Brown2012-04-271-12/+20
| | | | | | | | | Reduce our stack consumption by moving the params off the stack, they are reasonably large and might be an issue on platforms with small stacks. Reported-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Ackeded-by: Liam Girdwood <lrg@ti.com>
* ASoC: wm8994: Add trace showing wm8958_micd_set_rate()Mark Brown2012-04-261-0/+4
| | | | | | This can be helpful to users when tuning their systems. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm8994: Allow rate configuration with custom mic callbackMark Brown2012-04-261-1/+2
| | | | | | | If a driver using a custom mic detection callback has provided a table of mic detection rates via platform data then use it. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm8994: Tune debounce rates for jack detect modeMark Brown2012-04-261-2/+4
| | | | | | | | Use a slightly larger debounce when identifying accessory type and a slightly smaller one when detecting buttons in response to user feedback from large scale testing. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm8996: Put the microphone biases into bypass mode when idleMark Brown2012-04-261-0/+12
| | | | | | | | | When we're not actively doing audio we don't need the microphone biases to be regulated, noise is not important when we are not looking at the audio signal. Save some power by putting the MICBIAS regulators into bypass mode when not doing audio. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: pcm: Add pcm operation for pcm ioctl.Liam Girdwood2012-04-261-0/+13
| | | | | | | | | | | | Provide an ioctl marshaller for ASoC platform drivers. This will use the default ALSA handler if no platform handler exists. This is also required for DPCM BE PCMs as snd_pcm_info() will call the ioctl as part of stream startup. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: dpcm: Add bespoke trigger()Liam Girdwood2012-04-261-10/+89
| | | | | | | | | | | | | | | | Some on SoC DSP HW is very tightly coupled with DMA and DAI drivers. It's necessary to allow some flexability wrt to PCM operations here so that we can define a bespoke DPCM trigger() PCM operation for such HW. A bespoke DPCM trigger() allows exact ordering and timing of component triggering by allowing a component driver to manage the final enable and disable configurations without adding extra complexity to other component drivers. e.g. The McPDM DAI and ABE are tightly coupled on OMAP4 so we have a bespoke trigger to manage the trigger to improve performance and reduce complexity when triggering new McPDM BEs. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: dpcm: Add API for DAI link substream and runtime lookupLiam Girdwood2012-04-261-0/+29
| | | | | | | | Some component drivers will need to be able to look up their DAI link substream and RTD data. Provide a mechanism for this. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: dpcm: Add runtime dynamic route updateLiam Girdwood2012-04-262-2/+227
| | | | | | | | | | | | | | | | | This patch allows DPCM to dynamically alter the FE to BE PCM links at runtime based on mixer setting updates. DAPM is looked up after every mixer update and we perform a DPCM runtime update if the mixer has a change of value. This patchs adds/changes the following :- o Adds DPCM runtime update core. o Changes soc_dapm_mixer_update_power() and soc_dapm_mux_update_power() to return if a change has occured rather than 0. No other users check atm. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: dpcm: Add debugFS support for DPCMLiam Girdwood2012-04-262-0/+163
| | | | | | | Add debugFS files for DPCM link management information. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: dpcm: Add Dynamic PCM core operations.Liam Girdwood2012-04-262-30/+1221
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Dynamic PCM core allows digital audio data to be dynamically routed between different ALSA PCMs and DAI links on SoC CPUs with on chip DSP devices. e.g. audio data could be played on pcm:0,0 and routed to any (or all) SoC DAI links. Dynamic PCM introduces the concept of Front End (FE) PCMs and Back End (BE) PCMs. The FE PCMs are normal ALSA PCM devices except that they can dynamically route digital audio data to any supported BE PCM. A BE PCM has no ALSA device, but represents a DAI link and it's substream and audio HW parameters. e.g. pcm:0,0 routing digital data to 2 external codecs. FE pcm:0,0 ----> BE (McBSP.0) ----> CODEC 0 +--> BE (McPDM.0) ----> CODEC 1 e.g. pcm:0,0 and pcm:0,1 routing digital data to 1 external codec. FE pcm:0,0 --- +--> BE (McBSP.0) ----> CODEC FE pcm:0,1 --- The digital audio routing is controlled by the usual ALSA method of mixer kcontrols. Dynamic PCM uses a DAPM graph to work out the routing based upon the mixer settings and configures the BE PCMs based on routing and the FE HW params. DPCM is designed so that most ASoC component drivers will need no modification at all. It's intended that existing CODEC, DAI and platform drivers can be used in DPCM based audio devices without any changes. However, there will be some cases where minor changes are required (e.g. for very tightly coupled HW) and there are helpers to support this too. Somethimes the HW params of a FE and BE do not match or are incompatible, so in these cases the machine driver can reconfigure any hw_params and make any DSP perform sample rate / format conversion. This patch adds the core DPCM code and contains :- o The FE and BE PCM operations. o FE and BE DAI link support. o FE and BE PCM creation. o BE support API. o BE and FE link management. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: core: Remove unused variable 'min'Fabio Estevam2012-04-261-1/+0
| | | | | | | | | | | | | commit 4183eed2 (ASoC: core: Add signed multi register control) introduced the variable 'min',but it is not used. Remove it to fix the following build warning: sound/soc/soc-core.c: In function 'snd_soc_put_xr_sx': sound/soc/soc-core.c:2990: warning: unused variable 'min' Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: SSM2602: Convert to direct regmap API usageLars-Peter Clausen2012-04-251-30/+57
| | | | | | | | | Mostly a one to one converion. On one occasion the patch replaces a snd_soc_read-snd_soc_write sequence with regmap_update_bits though as it helps to keep the conversion simple. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: SSM2602: Remove driver specific versionLars-Peter Clausen2012-04-251-4/+0
| | | | | | | | We have never really updated that version number and probably never will, so just remove it. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: SSM2602: Add sysclk based rate constraintsLars-Peter Clausen2012-04-251-4/+34
| | | | | | | | Not all advertised rates are available for all sysclk frequencies. Add additional sysclk based rate constraints. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: bf5xx-ssm2602: Setup sysclock in init callbackLars-Peter Clausen2012-04-251-33/+4
| | | | | | | | The sysclock is fixed, so just set it up once in the init callback instead of setting it repeatably in the hw_params callback. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm8994: Make sure we disable FLL bypass when stopping the FLLKyung-Kwee Ryu2012-04-251-1/+1
| | | | | | | | | If FLL bypass is left enabled when we disable the CODEC then the output clock will be left running which consumes a small amount of additional current. Only enable bypass when there is an output. Signed-off-by: Kyung-Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: core: Add strobe controlKristoffer KARLSSON2012-04-231-0/+63
| | | | | | | | | | | | | | | | | | | | | | | Added support for a control that strobes a bit in a register to high then back to low (or the inverse). This is typically useful for hardware that requires strobing a singe bit to trigger some functionality and where exposing the bit in a normal single control would require the user to first manually set then again unset the bit again for the strobe to trigger. Added convenience macro. SOC_SINGLE_STROBE Added accessor implementations. snd_soc_get_strobe snd_soc_put_strobe Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson@stericsson.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: core: Add signed multi register controlKristoffer KARLSSON2012-04-231-0/+118
| | | | | | | | | | | | | | | | | | | | Added control type that can span multiple consecutive codec registers forming a single signed value in a MSB/LSB manner. The control dynamically adjusts to the register word size configured in driver. Added convenience macro. SOC_SINGLE_XR_SX Added accessor implementations. snd_soc_info_xr_sx snd_soc_get_xr_sx snd_soc_put_xr_sx Signed-off-by: Kristoffer KARLSSON <kristoffer.karlsson@stericsson.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm8994: Delete trailing whitespace from sound/soc/codecs/wm8994.cJesper Juhl2012-04-231-7/+6
| | | | | | | | While reading through sound/soc/codecs/wm8994.c I noticed a fair amount of trailing whitespace. This patch gets rid of it. Signed-off-by: Jesper Juhl <jj@chaosbits.net> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: Update regmap access for WM5100 DSP control registersMark Brown2012-04-232-2/+282
| | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm1250-ev1: Support sample rate configurationMark Brown2012-04-191-0/+43
| | | | | | | The Springbank module can support a range of sample rates, selected at runtime via GPIO configuration. Allow these to be configured at runtime. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm1250-ev1: Support stereoMark Brown2012-04-191-2/+2
| | | | | | | Springbank can support stereo, though it is primarily intended for mono use cases. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: dapm: Add API call to query valid DAPM pathsLiam Girdwood2012-04-181-10/+112
| | | | | | | | | | | | | | In preparation for ASoC DSP support. Add a DAPM API call to determine whether a DAPM audio path is valid between source and sink widgets. This also takes into account all kcontrol mux and mixer settings in between the source and sink widgets to validate the audio path. This will be used by the DSP core to determine the runtime DAI mappings between FE and BE DAIs in order to run PCM operations. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: samsung: Hook up AIF2 to the CODEC on LittlemillMark Brown2012-04-181-12/+70
| | | | | | | | Connect the WM1250-EV1 baseband simulator on Littlemill systems up to the CODEC AIF2 using the new CODEC<->CODEC link support, allowing a wider range of use cases to be represented. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: wm8994: Keep AIF3 tristated when not in useMark Brown2012-04-171-5/+4
| | | | | | | | Since AIF3 shares clock signals with other audio interfaces in order to ensure it doesn't drive undesirable clocks we need to tristate it. Rather than forcing the machine driver to do so have the driver do this. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: da7210: Minor update for PLL and SRMAshish Chavan2012-04-171-13/+9
| | | | | | | | This patch converts multiple if conditions in to single if with "&&"s. Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com> Signed-off-by: David Dajun Chen <dchen@diasemi.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: da7210: Add support for PLL and SRMAshish Chavan2012-04-171-38/+187
| | | | | | | | | | | | | | | | | | | | | | | | | | | Current DA7210 driver does support PLL mode fully. It uses fixed value of input master clock and PLL mode is enabled and disabled based on the sampling frequency being used for playback or recording. It also doesn't support Sample Rate Measurement feature of DA7210 hardware. This patch adds full support for PLL and SRM. Basically following three modes of operation are possible for DA7210 hardware, (1) I2S SLAVE mode with PLL bypassed (2) I2S SLAVE mode with PLL enabled (3) I2S Master mode with PLL enabled This patch adds support for all three modes. Also, in case of SLAVE mode with PLL, it supports SRM (Sample Rate Measurement) feature of the chip. Actually this patch was submitted earlier and received some review comments, but after that the driver got update by other patches. Because of that, I am considering this as new patch and not versioning it based of previous patches. This version tries to take care of all review comments received for earlier submissions. Signed-off-by: Ashish Chavan <ashish.chavan@kpitcummins.com> Signed-off-by: David Dajun Chen <dchen@diasemi.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: Use dai_fmt in SpeysideMark Brown2012-04-161-29/+4
| | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: Merge tag 'v3.4-rc3' into for-3.5Mark Brown2012-04-1616-115/+127
|\ | | | | | | | | | | | | | | | | | | | | Linux 3.4-rc3 contains a bunch of Tegra changes which are conflicting annoyingly with the new development that's going on for Tegra so merge it up to resolve those conflicts. Conflicts: sound/soc/soc-core.c sound/soc/tegra/tegra_i2s.c sound/soc/tegra/tegra_spdif.c
| * Merge tag 'sound-3.4' of ↵Linus Torvalds2012-04-151-10/+26
| |\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull another round of sound fixes from Takashi Iwai: "A few regression fixes for Realtek HD-audio codecs, mainly specific to some laptop models." * tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: hda/realtek - Fix mem leak (and rid us of trailing whitespace). ALSA: hda/realtek - Add quirk for Mac Pro 5,1 machines ALSA: hda/realtek - Add a fixup entry for Acer Aspire 8940G ALSA: hda/realtek - Fix GPIO1 setup for Acer Aspire 4930 & co ALSA: hda/realtek - Add a few ALC882 model strings back
| | * ALSA: hda/realtek - Fix mem leak (and rid us of trailing whitespace).Jesper Juhl2012-04-131-7/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In sound/pci/hda/patch_realtek.c::alc_auto_fill_dac_nids(), in the 'for (;;)' loop, if the 'badness' value returned from fill_and_eval_dacs() is negative, then we'll return from the function without freeing the memory we allocated for 'best_cfg', thus leaking. Fix the leak by kfree()'ing the memory when badness is negative. While I was there I also noticed some trailing whitespace in the function that I removed (along with all other trailing whitespace in the file) - it didn't seem worth-while to do that as two patches, so I hope it's OK that I just did it all as one patch. Signed-off-by: Jesper Juhl <jj@chaosbits.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * ALSA: hda/realtek - Add quirk for Mac Pro 5,1 machinesJosh Boyer2012-04-121-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | A user reported that setting model=imac24 used to allow sound to work on their Mac Pro 5,1 machine. Commit 5671087ffa "Move ALC885 macpro and imac24 models to auto-parser" removed this model option. All Mac machines are now explicitly handled with a quirk and the auto-parser. This adds a quirk for the device found on the Mac Pro 5,1 machines. This (partially) fixes https://bugzilla.redhat.com/show_bug.cgi?id=808559 [sorted the new entry in the ID number order by tiwai] Reported-by: Gabriel Somlo <somlo@cmu.edu> Signed-off-by: Josh Boyer <jwboyer@redhat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * ALSA: hda/realtek - Add a fixup entry for Acer Aspire 8940GTakashi Iwai2012-04-121-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | It's compatible with 8930G. Using the same fixup gives the proper 5.1 sound back. Reported-and-tested-by: Dany Martineau <dany.luc.martineau@gmail.com> Cc: <stable@kernel.org> [v3.3+] Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * ALSA: hda/realtek - Fix GPIO1 setup for Acer Aspire 4930 & coTakashi Iwai2012-04-121-2/+6
| | | | | | | | | | | | | | | | | | | | | | | | Add GPIO1 setup explicitly for Acer Aspire 493x & co. This could be set by alc_auto_init_amp(), but it's safer to set it more explicitly in the fixup table. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * ALSA: hda/realtek - Add a few ALC882 model strings backTakashi Iwai2012-04-111-1/+9
| | | | | | | | | | | | | | | | | | | | | | | | Since there are still many Acer models that might not be covered by the current fixup table, let's add back a few typical model names so that user can test the fixup without recompiling. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge tag 'sound-3.4' of ↵Linus Torvalds2012-04-1116-104/+108
| |\| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: - A series of fixes for Conexant 20549 HD-audio codec chip - A workaround for HDMI hotplug debug prints that annoyed people - A fix for the new support of platform DAPM contexts - Many driver-specific minor fixes * tag 'sound-3.4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: hda - hide HDMI/ELD printks unless snd.debug=2 ALSA: sound/isa/sscape.c: add missing resource-release code sound: sound/oss/msnd_pinnacle.c: add vfrees ALSA: hda - clean up CX20549 test mixer setup ALSA: hda - CX20549 doesn't need pin_amp_workaround. ALSA: hda - Remove CD control from model=benq for CX20549 ALSA: hda - fix record volume controls of CX20459 ("Venice") ALSA: hda - Rename capture sources of CX20549 to match common conventions ALSA: hda - Fix proc output for ADC amp values of CX20549 ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FS ASoC: set idle_bias_off=1 for all platform DAPM contexts ASoC: imx-audmux: Check for NULL pointer ASoC: imx-audmux: Fix ssi port numbers in sysfs ASoC: ak4642: fixup: mute needs +1 step MAINTAINERS: Don't list everyone working on Wolfson drivers MAINTAINERS: Add missing ASoC OMAP co-maintainer ASoC: pxa: pxa2xx-i2s: add io.h for IOMEM macro ASoC: tegra: ensure clocks are enabled when touching registers ASoC: sgtl5000: Enable VAG when DAC/ADC up ALSA: asihpi - fix return value of hpios_locked_mem_alloc()
| | * ALSA: hda - hide HDMI/ELD printks unless snd.debug=2Fengguang Wu2012-04-102-8/+7
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Also remove two warnings when CONFIG_SND_DEBUG is not set: sound/pci/hda/patch_hdmi.c: In function ‘hdmi_intrinsic_event’: sound/pci/hda/patch_hdmi.c:761:6: warning: unused variable ‘eldv’ [-Wunused-variable] sound/pci/hda/patch_hdmi.c:760:6: warning: unused variable ‘pd’ [-Wunused-variable] Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * ALSA: sound/isa/sscape.c: add missing resource-release codeJulia Lawall2012-04-101-2/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | At the point of this error-handling code, both regions and the dma have been allocated, so free it as done in previous and subsequent error-handling code. Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * sound: sound/oss/msnd_pinnacle.c: add vfreesJulia Lawall2012-04-101-2/+6
| | | | | | | | | | | | | | | | | | | | | | | | | | | At the point of this error-handling code, HAVE_DSPCODEH may be undefined, so free INITCODE and PERMCODE as done elsewhere. A jump and label are introduced to avoid code duplication. Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * Merge tag 'asoc-3.4' of ↵Takashi Iwai2012-04-0738-185/+257
| | |\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus ASoC: fixes for 3.4 A bunch of driver-specific fixes and one generic fix for the new support for platform DAPM contexts - we were picking the wrong default for the idle_bias_off setting which was meaning we weren't actually achieving any useful runtime PM on platform devices.
generated by cgit v1.2.3 (git 2.39.1) at 2025-01-13 13:26:11 +0100