- May 10, 2012
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Brian Austin authored
We need to read the real register values Signed-off-by:
Brian Austin <brian.austin@cirrus.com> Signed-off-by:
Mark Brown <broonie@opensource.wolfsonmicro.com>
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Brian Austin authored
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- May 09, 2012
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Mark Brown authored
We're trying to remove all usage of the ASoc level cache and I/O code and for a device like this with a pretty sparse register map the rbtree cache is a better idea anyway. Signed-off-by: -
Mark Brown authored
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Ola Lilja authored
Add driver for running I2S with the MSP-block. Signed-off-by:
Ola Lilja <ola.o.lilja@stericsson.com> [Fixed trailing whitespace -- broonie] Signed-off-by:
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Axel Lin authored
Signed-off-by:
Axel Lin <axel.lin@gmail.com> Signed-off-by:
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Axel Lin authored
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Axel Lin authored
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- May 08, 2012
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Mark Brown authored
More current API usage. Signed-off-by: -
Mark Brown authored
The WM9081 on Lowland is connected to AIF3 on the WM5100. Signed-off-by: -
Mark Brown authored
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Mark Brown authored
When we instantiate an aux_dev we use a fake rtd as part of the process which doesn't have a dai_link associated with it. Fix the dpcm startup code to cope with this. Signed-off-by:
Liam Girdwood <lrg@ti.com>
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Peter Ujfalusi authored
None of the machines uses the gain ramp possibility for HS/HF. This code path is mostly unused and it does not reduces the pop noise on the output (it alters it to sound a bit different). The preferred method to reduce pop noise is to use ABE. Remove the gain ramp, and related features form the driver. Signed-off-by:
Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by:
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- May 07, 2012
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guoyh authored
Allocating the SSP DMA parameters in startup, freeing it in shutdown instead of freeing and re-allocating it in hw_params. After doing that, the logic is clear and more safe. Signed-off-by:
guoyh <guoyh@marvell.com> Signed-off-by:
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- May 04, 2012
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Ashish Chavan authored
This patch improves playback quality for few sample rates like 8000 and 11025 Hz. This also fixes an issue observed during testing of pll slave mode. Due to the issue, on some rare occasions there was no sound output for first time playback after system boot, though all subsequent playbacks were fine. It was mainly because of the sequence in which SRM bit was enabled. Signed-off-by:
Ashish Chavan <ashish.chavan@kpitcummins.com> Signed-off-by:
David Dajun Chen <dchen@diasemi.com> Signed-off-by:
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- May 02, 2012
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Mark Brown authored
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Mark Brown authored
Devices with many DAIs are becoming more and more common, and generally the more modern devices have consistent register layouts between DAIs. Rather than have drivers open code lookups based on the DAI ID or cause uglification in UI by having register addresses for IDs provide a base address field they can use. Signed-off-by:
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Mark Brown authored
Rather than invalidating the cached DCS value every time the headphone gain changes store multiple values, indexed by gain. This allows the optimisation we get from the cache to take effect more often. Signed-off-by:
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- May 01, 2012
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Stephen Warren authored
This binding doesn't include the nvidia,model or nvidia,audio-routing properties the other Tegra audio DT bindings have, because this binding is targetted at a single machine, rather than for any machine using the tlv320aic23 codec. Signed-off-by:
Stephen Warren <swarren@nvidia.com> Signed-off-by:
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Mark Brown authored
Makes the code more standard and prepares for better framework usage. Signed-off-by: -
Mark Brown authored
This is now very standard behaviour for CODECs so shouldn't be device specific and we shouldn't really be trying to peer into the register cache from atomic context anyway. Signed-off-by: -
Mark Brown authored
In the conversion to module_init_i2c() the original open coded module exit function was left. Remove it. Signed-off-by: -
Brian Austin authored
This patch adds support for Cirrus Logic CS42L52 Low Power Stereo Codec Signed-off-by:
Georgi Vlaev <joe@nucleusys.com> Signed-off-by:
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- Apr 30, 2012
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Liam Girdwood authored
We should check dailess before dereferencing. Reported-by:
Dan Carpenter <dan.carpenter@oracle.com> Signed-off-by:
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- Apr 28, 2012
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Richard Zhao authored
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:
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Richard Zhao authored
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Mark Brown authored
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 authored
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 authored
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: -
Liam Girdwood authored
Remove writable debugFS permission, use simple_open() and fix indentation. Signed-off-by:
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Ashish Chavan authored
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:
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Mark Brown authored
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:
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- Apr 27, 2012
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Mark Brown authored
This can be helpful to users when tuning their systems. Signed-off-by: -
Mark Brown authored
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 authored
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 authored
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: -
Liam Girdwood authored
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:
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Liam Girdwood authored
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:
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Liam Girdwood authored
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:
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Liam Girdwood authored
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:
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