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* 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 Brown2012-04-165-14/+21
|\ | | | | | | | | | | | | | | | | | | | | 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 Torvalds2012-04-117-15/+30
| |\ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | 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()
| | * ASoC: tegra: fix i2s compilation when !CONFIG_DEBUG_FSStephen Warren2012-04-061-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit d4a2eca "ASoC: Tegra I2S: Remove dependency on pdev->id" changed the prototype of tegra_i2s_debug_add, but didn't update the dummy inline used when !CONFIG_DEBUG_FS. Fix that. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: <stable@vger.kernel.org> # 3.3
| | * ASoC: set idle_bias_off=1 for all platform DAPM contextsStephen Warren2012-04-051-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The ASoC core currently defaults to using STANDBY rather than OFF for idle ASoC platform devices, which causes a permanent pm_runtime_get() on them. This keeps the device active unnecessarily. This can be especially problematic when the ASoC platform device and DAI device are the same device. The distinction between OFF and STANDBY is likely not relevant for ASoC platform drivers, since they aren't analog devices. So, solve this issue by hard-coding idle_bias_off = 1 for all ASoC platform devices. If this turns out to be a problem, this value could be sourced from the snd_soc_platform_driver, similarly to soc_probe_codec(). Note: Prior to this change, this caused a large (10) runtime_active count for the Tegra I2S controller even when not in use, and a leak in that value as streams were started and stopped. This change probably hides a bug. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: imx-audmux: Check for NULL pointerFabio Estevam2012-04-051-0/+3
| | | | | | | | | | | | | | | | | | | | | Check for NULL pointer before accessing it. Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: imx-audmux: Fix ssi port numbers in sysfsFabio Estevam2012-04-051-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Doing a 'cat /sys/kernel/debug/audmux/ssi7' causes the following oops to be printed by the kernel: Uhandled fault: external abort on non-linefetch (0x008) at 0xf53b003c Internal error: : 8 [#1] PREEMPT Modules linked in: CPU: 0 Not tainted (3.3.0-00033-gecc726e-dirty #307) PC is at audmux_read_file+0x68/0x2f4 LR is at clk_enable+0x3c/0x48 pc : [<c001b8c8>] lr : [<c00190a0>] psr: a0000013 sp : c3ad3f38 ip : c30a4000 fp : 00000003 r10: 00001000 r9 : be83fb00 r8 : c3ad3f80 r7 : c3ad3f80 r6 : 00000007 r5 : 00031010 r4 : c30a5000 r3 : f53b0000 r2 : 0000003c r1 : 380fa100 r0 : c068dda0 Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment user Control: 0005317f Table: 83034000 DAC: 00000015 Process cat (pid: 1042, stack limit = 0xc3ad2270) Stack: (0xc3ad3f38 to 0xc3ad4000) 3f20: c3139180 00000000 3f40: c3bc6500 00001000 be83fb00 c3ad3f80 00001000 c3ad2000 00000000 c0095f3c 3f60: 00000003 c3bc6508 c3bc6500 be83fb00 00000000 00000000 00001000 c0096010 3f80: 00000000 00000000 b6fe2050 00000000 00001000 be83fb00 00000003 00000003 3fa0: c000eb88 c000e9e0 00001000 be83fb00 00000003 be83fb00 00001000 00000000 3fc0: 00001000 be83fb00 00000003 00000003 00000001 00000001 00000000 00000003 3fe0: 000bec8c be83fae0 0000f808 b6ea8d5c 60000010 00000003 7dff7ede 749bedf1 [<c001b8c8>] (audmux_read_file+0x68/0x2f4) from [<c0095f3c>] (vfs_read+0xb0/0x144) [<c0095f3c>] (vfs_read+0xb0/0x144) from [<c0096010>] (sys_read+0x40/0x70) [<c0096010>] (sys_read+0x40/0x70) from [<c000e9e0>] (ret_fast_syscall+0x0/0x2c) Code: e1a02186 e2822004 e3500000 e7935186 (e7937002) ---[ end trace 4d046e31309023de ]--- Fix the ssi port numbers in sysfs to fix this problem. Reported-by: Joan Carles <joancarles@fqingenieria.es> Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: ak4642: fixup: mute needs +1 stepKuninori Morimoto2012-04-051-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ak4642 out_tlv is +12.0dB to -115.0 dB, and it supports mute. But current settings didn't care +1 step for mute. This patch adds it Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@vger.kernel.org
| | * ASoC: pxa: pxa2xx-i2s: add io.h for IOMEM macroMartin Jansa2012-04-021-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * fixes sound/soc/pxa/pxa2xx-i2s.c:86:2: error: implicit declaration of function 'IOMEM' [-Werror=implicit-function-declaration] sound/soc/pxa/pxa2xx-i2s.c:86:2: error: initializer element is not constant after 23019a733bb83c8499f192fb428b7e6e81c95a34 removed IOMEM definition from arch/arm/mach-pxa/include/mach/hardware.h Signed-off-by: Martin Jansa <Martin.Jansa@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: tegra: ensure clocks are enabled when touching registersStephen Warren2012-04-022-0/+8
| | | | | | | | | | | | | | | | | | | | | | | | Debugfs files could be accessed any time, so explicitly enable clocks when reading registers to generate debugfs file content. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: sgtl5000: Enable VAG when DAC/ADC upZeng Zhaoming2012-04-021-12/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | As manual described, VAG is an internal voltage reference of DAC/ADC, So enabled it before DAC/ADC up. One more thing should care about is VAG fully ramped down requires 400ms, wait it to avoid pop. Signed-off-by: Zeng Zhaoming <zengzm.kernel@gmail.com> Signed-off-by: Shawn Guo <shawn.guo@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: soc-dapm: Use '%llx' with 'u64' type.Fabio Estevam2012-04-161-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fix the following build warning: sound/soc/soc-dapm.c: In function 'snd_soc_dai_link_event': sound/soc/soc-dapm.c:2913: warning: format '%lx' expects type 'long unsigned int', but argument 3 has type 'u64' '%llx' should be used with 'u64' type. Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: core: Support transparent CODEC<->CODEC DAI linksMark Brown2012-04-162-10/+188
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Rather than having the user half start a stream but avoid any DMA to trigger data flow on links which don't pass through the CPU create a DAPM route between the two DAI widgets using a hw_params configuration provided by the machine driver with the new 'params' member of the dai_link struct. If no configuration is provided in the dai_link then use the old style even for CODEC<->CODEC links to avoid breaking systems. This greatly simplifies the userspace usage of such links, making them as simple as analogue connections with the stream configuration being completely transparent to them. This is achieved by defining a new dai_link widget type which is created when CODECs are linked and triggering the configuration of the link via the normal PCM operations from there. It is expected that the bias level callbacks will be used for clock configuration. Currently only the DAI format, rate and channel count can be configured and currently the only DAI operations which can be called are hw_params and digital_mute(). This corresponds well to the majority of CODEC drivers which only use other callbacks for constraint setting but there is obviously much room for extension here. We can't simply call hw_params() on startup as things like the system clocking configuration may change at runtime and in future it will be desirable to offer some configurability of the link parameters. At present we are also restricted to a single DAPM link for the entire DAI. Once we have better support for channel mapping it would also be desirable to extend this feature so that we can propagate per-channel power state over the link. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* | | ASoC: core: Bind DAIs to CODECs at registration timeMark Brown2012-04-161-6/+30
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | We should always have a CODEC already there when registering a CODEC DAI and for CODEC<->CODEC links a dai_link will have two CODECs so it's much simpler to do things at registration time. This results in a slight change in the error handling for failed CODEC DAI registrations but practically speaking these are never supposed to fail so there shouldn't be much issue. The change is that we don't fail the overall CODEC registration if the DAI registration fails; this seems more robust anyway as we may not need to use a given DAI in a particular system. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: core: Flip master for CODECs in the CPU slot of a CODEC<->CODEC linkMark Brown2012-04-161-4/+33
| | | | | | | | | | | | | | | | | | | | | | | | | | | When two CODEC DAIs are linked directly to each other then if we give the same master mode settings to both devices things won't work as either neither will drive or they'll drive against each other. Flip the settings for the DAI in the CPU slot of the DAI link. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: dapm: Allow DAI widgets to be routed throughMark Brown2012-04-161-1/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | In order to allow CODEC<->CODEC links to function we will need to allow DAPM paths to be created that pass through DAIs rather than only ones that are source or sunk at the DAI. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* | | ASoC: core: Return -ENOTSUPP instead of -EINVAL if mute is not supportedMark Brown2012-04-161-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | This helps us ignore errors in callers if the operation failed due to not being available as opposed to an error. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
* | | ASoC: dapm: release lock on error pathsDan Carpenter2012-04-151-5/+7
| | | | | | | | | | | | | | | | | | | | | | | | We added locking here but there were a couple error paths where we forgot to drop the lock before returning. Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: convert Tegra20 DAS driver to regmapStephen Warren2012-04-132-65/+37
| | | | | | | | | | | | | | | Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | ASoC: convert Tegra20 SPDIF driver to regmapStephen Warren2012-04-132-84/+92
| | | | | | | | | | | | | | | Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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