- 01 5月, 2012 2 次提交
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由 Mark Brown 提交于
In the conversion to module_init_i2c() the original open coded module exit function was left. Remove it. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Brian Austin 提交于
This patch adds support for Cirrus Logic CS42L52 Low Power Stereo Codec Signed-off-by: NBrian Austin <brian.austin@cirrus.com> Signed-off-by: NGeorgi Vlaev <joe@nucleusys.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 30 4月, 2012 1 次提交
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由 Liam Girdwood 提交于
We should check dailess before dereferencing. Reported-by: NDan Carpenter <dan.carpenter@oracle.com> Signed-off-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 28 4月, 2012 8 次提交
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由 Richard Zhao 提交于
It tries to clk_get the clock. And if it failed, it assumes the clock by default enabled. Signed-off-by: NRichard Zhao <richard.zhao@freescale.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Richard Zhao 提交于
Signed-off-by: NRichard Zhao <richard.zhao@freescale.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Liam Girdwood 提交于
Remove writable debugFS permission, use simple_open() and fix indentation. Signed-off-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Ashish Chavan 提交于
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: NAshish Chavan <ashish.chavan@kpitcummins.com> Signed-off-by: NDavid Dajun Chen <dchen@diasemi.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com> Ackeded-by: NLiam Girdwood <lrg@ti.com>
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- 27 4月, 2012 10 次提交
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由 Mark Brown 提交于
This can be helpful to users when tuning their systems. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Liam Girdwood 提交于
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: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Liam Girdwood 提交于
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: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Liam Girdwood 提交于
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: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Liam Girdwood 提交于
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: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Liam Girdwood 提交于
Add debugFS files for DPCM link management information. Signed-off-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Liam Girdwood 提交于
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: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 26 4月, 2012 1 次提交
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由 Fabio Estevam 提交于
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: NFabio Estevam <fabio.estevam@freescale.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 25 4月, 2012 5 次提交
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由 Lars-Peter Clausen 提交于
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: NLars-Peter Clausen <lars@metafoo.de> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Lars-Peter Clausen 提交于
We have never really updated that version number and probably never will, so just remove it. Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Lars-Peter Clausen 提交于
Not all advertised rates are available for all sysclk frequencies. Add additional sysclk based rate constraints. Signed-off-by: NLars-Peter Clausen <lars@metafoo.de> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Lars-Peter Clausen 提交于
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: NLars-Peter Clausen <lars@metafoo.de> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Kyung-Kwee Ryu 提交于
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: NKyung-Kwee Ryu <Kyung-Kwee.Ryu@wolfsonmicro.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 24 4月, 2012 4 次提交
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由 Kristoffer KARLSSON 提交于
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: NKristoffer KARLSSON <kristoffer.karlsson@stericsson.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Kristoffer KARLSSON 提交于
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: NKristoffer KARLSSON <kristoffer.karlsson@stericsson.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Jesper Juhl 提交于
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: NJesper Juhl <jj@chaosbits.net> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 23 4月, 2012 1 次提交
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由 Liam Girdwood 提交于
Fixes the following build warning on x86_64. In file included from include/trace/ftrace.h:567:0, from include/trace/define_trace.h:86, from include/trace/events/asoc.h:410, from sound/soc/soc-core.c:45: include/trace/events/asoc.h: In function 'ftrace_raw_event_snd_soc_dapm_output_path': include/trace/events/asoc.h:246:1: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast] include/trace/events/asoc.h: In function 'ftrace_raw_event_snd_soc_dapm_input_path': include/trace/events/asoc.h:275:1: warning: cast from pointer to integer of different size [-Wpointer-to-int-cast] Signed-off-by: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 19 4月, 2012 4 次提交
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Springbank can support stereo, though it is primarily intended for mono use cases. Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Liam Girdwood 提交于
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: NLiam Girdwood <lrg@ti.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 18 4月, 2012 2 次提交
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由 Mark Brown 提交于
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: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Ashish Chavan 提交于
This patch converts multiple if conditions in to single if with "&&"s. Signed-off-by: NAshish Chavan <ashish.chavan@kpitcummins.com> Signed-off-by: NDavid Dajun Chen <dchen@diasemi.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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- 17 4月, 2012 2 次提交
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由 Ashish Chavan 提交于
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: NAshish Chavan <ashish.chavan@kpitcummins.com> Signed-off-by: NDavid Dajun Chen <dchen@diasemi.com> Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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由 Mark Brown 提交于
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
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