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linux/include/sound/soc.h

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ASoC: soc-core: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-02 06:24:18 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: soc-core: convert to SPDX identifiers Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-02 06:24:18 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: add .of_xlate_dai_name on snd_soc_component_driver ASoC sound driver requires CPU/CODEC drivers for probing, and each CPU/CODEC has some DAI on it. Then, "dai name matching" have been used to identify CPU-CODEC DAI pair on ASoC. But, the "dai port number matching" is now required from DeviceTree. The solution of this issue is to replace the dai port number into dai name. Now, CPU/CODEC are based on struct snd_soc_component, and it can care above as common issue. This patch adds .of_xlate_dai_name callback interface on struct snd_soc_component_driver, and snd_soc_of_get_dai_name() which is using .of_xlate_dai_name. Then, #sound-dai-cells which enables DAI specifier is required on CPU/CODEC device tree properties. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-09-10 17:39:56 -07:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: Add a notifier for jack status changes Some systems provide both mechanical and electrical detection of jack status changes. On such systems power savings can be achieved by only enabling the electrical detection methods when physical insertion has been detected. Begin supporting such systems by providing a notifier for jack status changes which can be used to trigger any reconfiguration. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-22 12:06:30 +00:00
[ALSA] Fix the soc code after dhowells workqueue changes. From: Andrew Morton <akpm@osdl.org> I converted the workqueues to per-device while I was there. It seems strange to create a new kernel thread (on each CPU!) and to then only have a single global work to ever be queued upon it. Plus without this, I'd have to use the _NAR stuff, gawd help me. Does that workqueue really need to be per-cpu? Does that workqueue really need to exist? Why not use keventd? Cc: Takashi Iwai <tiwai@suse.de> Cc: David Howells <dhowells@redhat.com> Signed-off-by: Andrew Morton <akpm@osdl.org> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-12-15 09:30:07 +01:00
ASoC: Add GPIO support for jack reporting interface Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpios, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:25:04 -06:00
ASoC: Use new register map API for ASoC generic physical I/O Remove all the ASoC specific physical I/O code and replace it with calls into the regmap API. The bulk write code can only be used safely if all regmap calls are locked with the CODEC lock, we need to add bulk support to the regmap API or replace the code with an open coded loop (though currently it has no users...). Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-13 19:35:29 +01:00
ASoC: Avoid recalculating the bitmask for SOC_ENUM controls For ENUM controls the bitmask is calculated based on the number of items. Currently this is done each time the control is accessed. And while the performance impact of this should be negligible we can easily do better. The roundup_pow_of_two macro performs the same calculation which is currently done manually, but it is also possible to use this macro with compile time constants and so it can be used to initialize static data. So we can use it to initialize the mask field of a ENUM control during its declaration. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-09-14 13:57:27 +02:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: add definations for compressed operations Here we update the asoc structures to add compress stream definations First the struct snd_soc_dai_driver adds a new member to indicate if the dai is compressed or pcm. Next we add a new structre the struct snd_soc_compr_ops in the struct snd_soc_dai_link. This is to be used for machine driver to perform any opertaions required for setting up compressed audio streams next is the compressed data operations, they are added using struct snd_compr_ops in the struct snd_soc_platform_driver. Signed-off-by: Namarta Kohli <namartax.kohli@intel.com> Signed-off-by: Ramesh Babu K V <ramesh.babu@intel.com> Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-16 17:10:40 +05:30
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: dapm: Implement mixer input auto-disable Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-05 11:27:31 +02:00
ALSA: ASoC: Convert bitfields in ASoC into full int width Convert bitfields in ASoC into full int width. This is a simple mechanical conversion. Two places in the DAPM code were fixed to properly use mask. Signed-off-by: Jon Smirl <jonsmirl@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-07-29 11:42:26 +01:00
ASoC: core: Change SOC_SINGLE/DOUBLE_VALUE representation SOC_SINGLE/DOUBLE_VALUE is used for mixer controls, where the bits are within one register. Assign .rreg to be the same as .reg for these types. With this change we can tell if the mixer in question: is mono: mc->reg == mc->rreg && mc->shift == mc->rshift is stereo, within single register: mc->reg == mc->rreg && mc->shift != mc->rshift is stereo, in two registers: mc->reg != mc->rreg The patch provide a small inline function to query, if the mixer is stereo, or mono. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-05 10:29:22 +03:00
ASoC: dapm: Implement mixer input auto-disable Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-05 11:27:31 +02:00
ASoC: Use generic control handlers for S8 control Commit f227b88f0fce ("ASoC: core: Add signed register volume control logic") added support for signed control to the generic volsw control handler. This makes it possible to use them for the S8 control as well, rather than having to use a custom control handler implementation. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2014-10-25 20:25:56 +02:00
ASoC: dapm: Implement mixer input auto-disable Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-05 11:27:31 +02:00
ALSA: ASoC: Convert bitfields in ASoC into full int width Convert bitfields in ASoC into full int width. This is a simple mechanical conversion. Two places in the DAPM code were fixed to properly use mask. Signed-off-by: Jon Smirl <jonsmirl@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-07-29 11:42:26 +01:00
ASoC: core: Fix for the volume limiting when invert is in use If the register for the volume needs invert, than the inversion need to be done from the chip maximum, and not from the platform dependent limit. Introduce soc_mixer_control.platform_max value, which initially equals to chip maximum. The snd_soc_limit_volume function only modify the platform_max, all volsw_info call returns this as well. The .max value holds the chip default (maximum), and it is used for the inversion, if it is needed. Additional check in the volsw_info call has been added to check the validity of the platform_max in case, when custom macros used by codec drivers are not initializing it correctly. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-10 14:39:24 +03:00
ASoC: core: Introduce SOC_DOUBLE_R_VALUE macro With the new macro we can remove duplicated code for the SOC_DOUBLE_R type of controls. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 14:39:39 +03:00
ASoC: core: Add SOC_DOUBLE_R_S_TLV Add a macro for signed mixer with two registers and tlv array. Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-01-16 16:02:11 +01:00
ASoC: core: Add DOUBLE_R variants of the _RANGE controls The code handles this fine already, we just need new macros in the header for drivers to create the controls. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2012-06-22 11:40:55 +01:00
[ALSA] ASoC TLV support Add TLV support to ASoC. Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-10 14:37:42 +01:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: dapm: Implement mixer input auto-disable Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-05 11:27:31 +02:00
ASoC: core: Add single controls with specified range of values Control type added for cases where a specific range of values within a register are required for control. Added convenience macros: SOC_SINGLE_RANGE SOC_SINGLE_RANGE_TLV Added accessor implementations: snd_soc_info_volsw_range snd_soc_put_volsw_range snd_soc_get_volsw_range Signed-off-by: Michal Hajduk <Michal.Hajduk@diasemi.com> Signed-off-by: Adam Thomson <Adam.Thomson@diasemi.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-31 15:18:01 +01:00
ASoC: core: Fix SOC_DOUBLE_RANGE() macros Although we've had macros defining double _RANGE controls for a while now they've not actually been backed up properly by the implementation, it's treated everything as mono. Fix that by implementing the handling in the stereo controls, ensuring that the mono controls don't mistakenly get treated as stereo. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2012-12-19 16:05:00 +00:00
[ALSA] ASoC TLV support Add TLV support to ASoC. Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-10 14:37:42 +01:00
ASoC: dapm: Implement mixer input auto-disable Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-05 11:27:31 +02:00
ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV Some codecs namely Cirrus Logic Codecs have a way of wrapping the dB scale around 0dB without 0dB being in the middle. Rework of SOC_DOUBLE_R_SX_TLV to be more consistent with other asoc tlv macros. Add single register macro : SOC_SINGLE_SX_TLV. Use snd_soc_info_volsw for .info Use snd_soc_get_volsw_sx, snd_soc_put_volsw_sx for single and double. kcontrols for CS42L51 and CS42L73 are adjusted to these new TLV Macros. The max value is determined by: (number of steps) +1 for 0dB +max from codec datasheet. Signed-off-by: Brian Austin <brian.austin@cirrus.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-03-30 10:43:55 -05:00
ASoC: Add info callback for SX_TLV controls SX_TLV controls are intended for situations where the register behind the control has some non-zero value indicating the minimum gain and then gains increasing from there and eventually overflowing through zero. Currently every CODEC implementing these controls specifies the minimum as the non-zero value for the minimum and the maximum as the number of gain settings available. This means when the info callback subtracts the minimum value from the maximum value to calculate the number of gain levels available it is actually under reporting the available levels. This patch fixes this issue by adding a new snd_soc_info_volsw_sx callback that does not subtract the minimum value. Fixes: 1d99f2436d0d ("ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV") Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Acked-by: Brian Austin <brian.austin@cirrus.com> Tested-by: Brian Austin <brian.austin@cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org
2015-10-14 13:31:24 +01:00
ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV Some codecs namely Cirrus Logic Codecs have a way of wrapping the dB scale around 0dB without 0dB being in the middle. Rework of SOC_DOUBLE_R_SX_TLV to be more consistent with other asoc tlv macros. Add single register macro : SOC_SINGLE_SX_TLV. Use snd_soc_info_volsw for .info Use snd_soc_get_volsw_sx, snd_soc_put_volsw_sx for single and double. kcontrols for CS42L51 and CS42L73 are adjusted to these new TLV Macros. The max value is determined by: (number of steps) +1 for 0dB +max from codec datasheet. Signed-off-by: Brian Austin <brian.austin@cirrus.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-03-30 10:43:55 -05:00
ASoC: core: Add single controls with specified range of values Control type added for cases where a specific range of values within a register are required for control. Added convenience macros: SOC_SINGLE_RANGE SOC_SINGLE_RANGE_TLV Added accessor implementations: snd_soc_info_volsw_range snd_soc_put_volsw_range snd_soc_get_volsw_range Signed-off-by: Michal Hajduk <Michal.Hajduk@diasemi.com> Signed-off-by: Adam Thomson <Adam.Thomson@diasemi.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-31 15:18:01 +01:00
ASoC: core: Fix SOC_DOUBLE_RANGE() macros Although we've had macros defining double _RANGE controls for a while now they've not actually been backed up properly by the implementation, it's treated everything as mono. Fix that by implementing the handling in the stereo controls, ensuring that the mono controls don't mistakenly get treated as stereo. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2012-12-19 16:05:00 +00:00
ASoC: core: Introduce SOC_DOUBLE_VALUE macro With the new macro we can remove duplicated code for the SOC_DOUBLE type of controls. We can also remap the SOC_SINGLE_VALUE macro to SOC_DOUBLE_VALUE Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 14:39:38 +03:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: core: Introduce SOC_DOUBLE_VALUE macro With the new macro we can remove duplicated code for the SOC_DOUBLE type of controls. We can also remap the SOC_SINGLE_VALUE macro to SOC_DOUBLE_VALUE Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 14:39:38 +03:00
ASoC: dapm: Implement mixer input auto-disable Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-05 11:27:31 +02:00
ASoC: Add SOC_DOUBLE_STS macro Add SOC_DOUBLE_STS macro for read-only volatile status controls Signed-off-by: Damien.Horsley <Damien.Horsley@imgtec.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-12-08 15:58:58 +00:00
ALSA: ASoC: Convert bitfields in ASoC into full int width Convert bitfields in ASoC into full int width. This is a simple mechanical conversion. Two places in the DAPM code were fixed to properly use mask. Signed-off-by: Jon Smirl <jonsmirl@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-07-29 11:42:26 +01:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: core: Combine snd_soc_info_volsw/info_volsw_2r functions Handle the info_volsw/info_volsw_2r in one function. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-05 10:29:23 +03:00
ASoC: core: Combine snd_soc_put_volsw/put_volsw_2r functions Handle the put_volsw/put_volsw_2r in one function. To avoid build breakage in twl6040 keep the snd_soc_put_volsw_2r as define, and map it snd_soc_put_volsw. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-05 10:29:25 +03:00
ASoC: core: Introduce SOC_DOUBLE_R_VALUE macro With the new macro we can remove duplicated code for the SOC_DOUBLE_R type of controls. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 14:39:39 +03:00
ASoC: core: Add DOUBLE_R variants of the _RANGE controls The code handles this fine already, we just need new macros in the header for drivers to create the controls. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2012-06-22 11:40:55 +01:00
ASoC: core: Introduce SOC_DOUBLE_VALUE macro With the new macro we can remove duplicated code for the SOC_DOUBLE type of controls. We can also remap the SOC_SINGLE_VALUE macro to SOC_DOUBLE_VALUE Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 14:39:38 +03:00
[ALSA] ASoC TLV support Add TLV support to ASoC. Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-10 14:37:42 +01:00
ASoC: core: Introduce SOC_DOUBLE_VALUE macro With the new macro we can remove duplicated code for the SOC_DOUBLE type of controls. We can also remap the SOC_SINGLE_VALUE macro to SOC_DOUBLE_VALUE Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 14:39:38 +03:00
ASoC: dapm: Implement mixer input auto-disable Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-05 11:27:31 +02:00
ALSA: ASoC: Convert bitfields in ASoC into full int width Convert bitfields in ASoC into full int width. This is a simple mechanical conversion. Two places in the DAPM code were fixed to properly use mask. Signed-off-by: Jon Smirl <jonsmirl@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-07-29 11:42:26 +01:00
[ALSA] ASoC TLV support Add TLV support to ASoC. Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-10 14:37:42 +01:00
ASoC: core: Combine snd_soc_info_volsw/info_volsw_2r functions Handle the info_volsw/info_volsw_2r in one function. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-05 10:29:23 +03:00
ASoC: core: Combine snd_soc_put_volsw/put_volsw_2r functions Handle the put_volsw/put_volsw_2r in one function. To avoid build breakage in twl6040 keep the snd_soc_put_volsw_2r as define, and map it snd_soc_put_volsw. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-05 10:29:25 +03:00
ASoC: core: Introduce SOC_DOUBLE_R_VALUE macro With the new macro we can remove duplicated code for the SOC_DOUBLE_R type of controls. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 14:39:39 +03:00
ASoC: core: Add DOUBLE_R variants of the _RANGE controls The code handles this fine already, we just need new macros in the header for drivers to create the controls. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2012-06-22 11:40:55 +01:00
ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV Some codecs namely Cirrus Logic Codecs have a way of wrapping the dB scale around 0dB without 0dB being in the middle. Rework of SOC_DOUBLE_R_SX_TLV to be more consistent with other asoc tlv macros. Add single register macro : SOC_SINGLE_SX_TLV. Use snd_soc_info_volsw for .info Use snd_soc_get_volsw_sx, snd_soc_put_volsw_sx for single and double. kcontrols for CS42L51 and CS42L73 are adjusted to these new TLV Macros. The max value is determined by: (number of steps) +1 for 0dB +max from codec datasheet. Signed-off-by: Brian Austin <brian.austin@cirrus.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-03-30 10:43:55 -05:00
ASoC: Add info callback for SX_TLV controls SX_TLV controls are intended for situations where the register behind the control has some non-zero value indicating the minimum gain and then gains increasing from there and eventually overflowing through zero. Currently every CODEC implementing these controls specifies the minimum as the non-zero value for the minimum and the maximum as the number of gain settings available. This means when the info callback subtracts the minimum value from the maximum value to calculate the number of gain levels available it is actually under reporting the available levels. This patch fixes this issue by adding a new snd_soc_info_volsw_sx callback that does not subtract the minimum value. Fixes: 1d99f2436d0d ("ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV") Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Acked-by: Brian Austin <brian.austin@cirrus.com> Tested-by: Brian Austin <brian.austin@cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org
2015-10-14 13:31:24 +01:00
ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV Some codecs namely Cirrus Logic Codecs have a way of wrapping the dB scale around 0dB without 0dB being in the middle. Rework of SOC_DOUBLE_R_SX_TLV to be more consistent with other asoc tlv macros. Add single register macro : SOC_SINGLE_SX_TLV. Use snd_soc_info_volsw for .info Use snd_soc_get_volsw_sx, snd_soc_put_volsw_sx for single and double. kcontrols for CS42L51 and CS42L73 are adjusted to these new TLV Macros. The max value is determined by: (number of steps) +1 for 0dB +max from codec datasheet. Signed-off-by: Brian Austin <brian.austin@cirrus.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-03-30 10:43:55 -05:00
ASoC: core: Add SOC_DOUBLE_R_S_TLV Add a macro for signed mixer with two registers and tlv array. Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-01-16 16:02:11 +01:00
ASoC: Introduce SOC_SINGLE_S8_TLV() macro This patch introduces SOC_SINGLE_S8_TLV() macro for volume control on chips which supports both negative and positive gains with sign bit on a 8 bit register, Gain ranges from -128 to +127 with a predefined step size. Currently we only have support to DOUBLE_S8_TLV() which does not fit for cases where we just have separate gain control register for each channel. One of the Qualcomm SOC msm8916 has such gain control register whose gain range is from -38.4dB to +38.4dB with step size of 0.3dB. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-05-27 14:45:45 +01:00
ALSA: ASoC: Convert bitfields in ASoC into full int width Convert bitfields in ASoC into full int width. This is a simple mechanical conversion. Two places in the DAPM code were fixed to properly use mask. Signed-off-by: Jon Smirl <jonsmirl@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-07-29 11:42:26 +01:00
[ALSA] ASoC: Add SOC_DOUBLE_S8_TLV control type The SOC_DOUBLE_S8_TLV control type was originally implemented in the UDA1380 driver by Philipp Zabel and was moved into the core by me. Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-05-28 17:58:05 +01:00
ASoC: Use generic control handlers for S8 control Commit f227b88f0fce ("ASoC: core: Add signed register volume control logic") added support for signed control to the generic volsw control handler. This makes it possible to use them for the S8 control as well, rather than having to use a custom control handler implementation. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2014-10-25 20:25:56 +02:00
ASoC: Rename soc_enum.max field with items The name "max" in struct soc_enum is rather confusing since it actually takes the number of items. With "max", one might try to assign (nitems - 1) value. Rename the field to a more appropriate one, "items", which is also used in struct snd_ctl_elem_info, too. This patch also rewrites some code like "if (x > e->nitems - 1)" with "if (x >= e->nitems)". Not only the latter improves the readability, it also fixes a potential bug when e->items is zero. Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-18 08:11:42 +01:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: Rename soc_enum.max field with items The name "max" in struct soc_enum is rather confusing since it actually takes the number of items. With "max", one might try to assign (nitems - 1) value. Rename the field to a more appropriate one, "items", which is also used in struct snd_ctl_elem_info, too. This patch also rewrites some code like "if (x > e->nitems - 1)" with "if (x >= e->nitems)". Not only the latter improves the readability, it also fixes a potential bug when e->items is zero. Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-18 08:11:42 +01:00
ASoC: New enum type: value_enum This patch introduces a new enum type. In this enum type each enumerated items referred with a value. This new enum type can handle enums encoded in bitfield, or any other weird ways. twl4030 codec has several mux selection register, where the input/output mux is coded in a bitfield. With the normal enum type this type of mux can not be handled in a clean way. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-05 09:54:57 +02:00
ASoC: Rename soc_enum.max field with items The name "max" in struct soc_enum is rather confusing since it actually takes the number of items. With "max", one might try to assign (nitems - 1) value. Rename the field to a more appropriate one, "items", which is also used in struct snd_ctl_elem_info, too. This patch also rewrites some code like "if (x > e->nitems - 1)" with "if (x >= e->nitems)". Not only the latter improves the readability, it also fixes a potential bug when e->items is zero. Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-18 08:11:42 +01:00
ASoC: Correct typo in SOC_VALUE_ENUM_SINGLE macro xnitmes is clearly intended to be xnitems, but all other macros just refer to this as xitems, so change it to that. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Tested-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-05-01 12:37:23 +01:00
ASoC: dapm: Add support for autodisable mux controls Commit 57295073b6ac ("ASoC: dapm: Implement mixer input auto-disable") added support for autodisable controls, controls whose values are only written to the hardware when their respective widgets are powered up. But it only added support for controls based on the mixer abstraction. This patch add support for mux controls (DAPM controls based on the enum abstraction) to be auto-disabled as well. As each mux can only have a single control, there is no need to tie the autodisable widget to the inputs (as is done for the mixer controls) it can be tided directly to the mux widget itself. Note that it is assumed that the first entry in a autodisable mux control will always represent the off state for the mux and is what the mux will be set to whilst it is disabled. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Tested-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-05-01 12:37:26 +01:00
ASoC: Add macros for defining virtual enums With the upcoming consolidation of normal MUXs and virtual MUXs we need to be able to distinguish between enums with and without a backing register at the enum level. Use the same approach as used for virtual mixer controls by setting the reg field of the enum to SND_SOC_NOPM for enums without a backing register. This patch adds a set of helper macros that can be used to define such enums. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-28 08:31:05 +01:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ALSA: ASoC: Rename mask to max to reflect usage Most of the ASoC controls refer to the maximum value that can be set for a control as mask but there is no actual requirement for all bits to be set at the highest possible value making the name mask misleading. Change the code to use max instead. Signed-off-by: Jon Smirl <jonsmirl@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-07-29 11:42:27 +01:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
[ALSA] soc - 0.13 ASoC headers This patch updates the API's to include the new DAI configuration and clocking architecture. Changes:- o Removed DAI automatic matching and capabilities structure (struct snd_soc_dai_mode) and macros. o Added DAI operations for codec and CPU interfaces. o Removed config_sysclk() function and struct snd_soc_clock_info. No longer needed as clocking is now configured manually in the machine drivers. Also removed other clocking data from structures. o Updated version to 0.13 o Added shift to SOC_SINGLE_EXT kcontrol macro. Signed-off-by: Graeme Gregory <gg@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-02 17:13:05 +01:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: dapm: Implement mixer input auto-disable Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-05 11:27:31 +02:00
ASoC: core: Introduce SOC_DOUBLE_VALUE macro With the new macro we can remove duplicated code for the SOC_DOUBLE type of controls. We can also remap the SOC_SINGLE_VALUE macro to SOC_DOUBLE_VALUE Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 14:39:38 +03:00
ASoC: add SOC_DOUBLE_EXT macro Add a macro for double controls with special callback functions. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-24 16:37:44 +02:00
ASoC: core: Introduce SOC_DOUBLE_VALUE macro With the new macro we can remove duplicated code for the SOC_DOUBLE type of controls. We can also remap the SOC_SINGLE_VALUE macro to SOC_DOUBLE_VALUE Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 14:39:38 +03:00
ASoC: dapm: Implement mixer input auto-disable Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-05 11:27:31 +02:00
ASoC: Add SOC_DOUBLE_R_EXT _EXT version of SOC_DOUBLE_R required to allow for custom handlers. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-29 16:43:56 +01:00
ALSA: ASoC: Rename mask to max to reflect usage Most of the ASoC controls refer to the maximum value that can be set for a control as mask but there is no actual requirement for all bits to be set at the highest possible value making the name mask misleading. Change the code to use max instead. Signed-off-by: Jon Smirl <jonsmirl@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-07-29 11:42:27 +01:00
ALSA: ASoC: Add SOC_SINGLE_EXT_TLV control type Signed-off-by: Mike Montour <mail@mmontour.net> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-06-11 13:47:13 +01:00
ASoC: dapm: Implement mixer input auto-disable Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-05 11:27:31 +02:00
ASoC: Add SOC_SINGLE_RANGE_EXT_TLV macro Add a version of the SOC_SINGLE_RANGE_TLV macro that allows a custom get and put to be specified. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-16 13:59:42 +01:00
ASoC: add SOC_DOUBLE_EXT_TLV control type This is a macro for double controls with special callback function and TLV. The SOC_DOUBLE_EXT_TLV needs one register and two shifts for double controls. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-15 20:33:47 +09:00
ASoC: core: Introduce SOC_DOUBLE_VALUE macro With the new macro we can remove duplicated code for the SOC_DOUBLE type of controls. We can also remap the SOC_SINGLE_VALUE macro to SOC_DOUBLE_VALUE Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 14:39:38 +03:00
ASoC: dapm: Implement mixer input auto-disable Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-05 11:27:31 +02:00
ASoC: add SOC_DOUBLE_R_EXT_TLV control type This is a macro for double controls with special callback function and TLV. The SOC_DOUBLE_R_EXT_TLV needs two registers and one shift for double controls. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-15 20:33:50 +09:00
ASoC: core: Combine snd_soc_info_volsw/info_volsw_2r functions Handle the info_volsw/info_volsw_2r in one function. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-05 10:29:23 +03:00
ASoC: add SOC_DOUBLE_R_EXT_TLV control type This is a macro for double controls with special callback function and TLV. The SOC_DOUBLE_R_EXT_TLV needs two registers and one shift for double controls. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-07-15 20:33:50 +09:00
ASoC: core: Introduce SOC_DOUBLE_R_VALUE macro With the new macro we can remove duplicated code for the SOC_DOUBLE_R type of controls. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-04 14:39:39 +03:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: Use snd_soc_info_enum_double() for SOC_ENUM_EXT controls snd_soc_info_enum_ext() and snd_soc_info_enum_double() are almost identical. The only difference is that snd_soc_info_enum_double() is also able to handle stereo controls. Using snd_soc_info_enum double() instead of snd_soc_info_enum_ext() for the SOC_ENUM_EXT control's info callback allows us to remove snd_soc_info_enum_ext(). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-10 21:33:10 +02:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: add SOC_VALUE_ENUM_EXT Adds an equivalent of SOC_ENUM_EXT for value enums Strictly speaking SOC_ENUM_EXT can also be used to define a value enum since the only difference is the get and set functions. But this doesn't look good in code because it is inconsistent with the normal control definitions. Adding a specific SOC_VALUE_ENUM_EXT is better for code clarity. Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-07 13:16:54 +01:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: core: Add SND_SOC_BYTES control for coefficient blocks Allow devices to export blocks of registers to the application layer, intended for use for reading and writing coefficient data which can't usefully be worked with by the kernel at runtime (for example, due to requiring complex and expensive calculations or being the results of callibration procedures). Currently drivers are using platform data to provide configurations for coefficient blocks which isn't at all convenient for runtime management or configuration development. Currently only devices using regmap are supported, an error will be generated for any attempt to work with a byte control on a non-regmap device. There's no fundamental block to other devices so support could be added if required. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-10-10 18:31:26 +01:00
ASoC: Add SOC_DOUBLE_R_SX_TLV control This patch is adding a new control which has the following capabilities: - tlv - variable data size (for instance, 7 ou 8 bit) - double mixer - data range centered around 0 Signed-off-by: Arnaud Patard <apatard@mandriva.com> Acked-by: Liam Girdwood <lrg@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-15 17:30:01 +02:00
ASoC: core: Add support for masking out parts of coefficient blocks Chip designers frequently include things like the enable and disable controls for algorithms in the register blocks which also hold the coefficients. Since it's desirable to split out the enable/disable control from userspace the plain SND_SOC_BYTES() isn't optimal for these devices. Add a SND_SOC_BYTES_MASK() which allows a bitmask from the first word of the block to be excluded from the control. This supports the needs of devices I've looked at and lets us have a reasonably simple API. Further controls can be added in future if that's needed. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2012-02-17 16:20:33 -08:00
ASoC: core: mark SND_SOC_BYTES_EXT as deprecated Since we have SND_SOC_BYTES_TLV control to lets devices have larger size data sent, we do not need SND_SOC_BYTES_EXT with 512 byte limitation so mark it deprecated Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-11-23 21:22:29 +05:30
ASoC: add SND_SOC_BYTES_EXT we need _EXT version for SND_SOC_BYTES so that DSPs can use this to pass data for DSP modules Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02 10:58:11 +05:30
ASoC: core: add a helper for extended byte controls using TLV ALSA supports arbitrary length TLVs for each kcontrol that can be used to pass metadata about the control (e.g. volumes, enum information). The same transport mechanism is now used for arbitrary length data by defining a new helper. Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah@intel.com> Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-15 21:34:48 +05:30
ASoC: core: fix .info for SND_SOC_BYTES_TLV Commit 7523a271 - "ASoC: core: add a helper for extended byte controls using TLV" introduced support for TLV byte controls but had a typo for the info function, so fix the same Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty@intel.com> Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-18 14:53:03 +05:30
ASoC: core: add a helper for extended byte controls using TLV ALSA supports arbitrary length TLVs for each kcontrol that can be used to pass metadata about the control (e.g. volumes, enum information). The same transport mechanism is now used for arbitrary length data by defining a new helper. Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah@intel.com> Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-15 21:34:48 +05:30
ASoC: core: Add signed multi register control 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>
2012-04-20 11:32:13 +02:00
ASoC: core: Add strobe control 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>
2012-04-20 11:32:44 +02:00
ASoC: add helper macros to declare struct soc_enum instances Several shortcuts for popular uses of some of SOC_ENUM_* and SOC_VALUE_ENUM_* macros. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-21 22:04:03 +01:00
ASoC: Add const to SOC_ENUM_*_DECL() macros Since these macros are supposed to be used for decalring const objects, let's add the const modifier there. The doubled const appearing in usages will be cleaned by later patches. Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-18 09:08:51 +01:00
ASoC: add helper macros to declare struct soc_enum instances Several shortcuts for popular uses of some of SOC_ENUM_* and SOC_VALUE_ENUM_* macros. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-21 22:04:03 +01:00
ASoC: Add const to SOC_ENUM_*_DECL() macros Since these macros are supposed to be used for decalring const objects, let's add the const modifier there. The doubled const appearing in usages will be cleaned by later patches. Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-18 09:08:51 +01:00
ASoC: add helper macros to declare struct soc_enum instances Several shortcuts for popular uses of some of SOC_ENUM_* and SOC_VALUE_ENUM_* macros. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-21 22:04:03 +01:00
ASoC: Add const to SOC_ENUM_*_DECL() macros Since these macros are supposed to be used for decalring const objects, let's add the const modifier there. The doubled const appearing in usages will be cleaned by later patches. Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-18 09:08:51 +01:00
ASoC: add helper macros to declare struct soc_enum instances Several shortcuts for popular uses of some of SOC_ENUM_* and SOC_VALUE_ENUM_* macros. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-21 22:04:03 +01:00
ASoC: dapm: Add support for autodisable mux controls Commit 57295073b6ac ("ASoC: dapm: Implement mixer input auto-disable") added support for autodisable controls, controls whose values are only written to the hardware when their respective widgets are powered up. But it only added support for controls based on the mixer abstraction. This patch add support for mux controls (DAPM controls based on the enum abstraction) to be auto-disabled as well. As each mux can only have a single control, there is no need to tie the autodisable widget to the inputs (as is done for the mixer controls) it can be tided directly to the mux widget itself. Note that it is assumed that the first entry in a autodisable mux control will always represent the off state for the mux and is what the mux will be set to whilst it is disabled. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Tested-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-05-01 12:37:26 +01:00
ASoC: Add macros for defining virtual enums With the upcoming consolidation of normal MUXs and virtual MUXs we need to be able to distinguish between enums with and without a backing register at the enum level. Use the same approach as used for virtual mixer controls by setting the reg field of the enum to SND_SOC_NOPM for enums without a backing register. This patch adds a set of helper macros that can be used to define such enums. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-28 08:31:05 +01:00
ASoC: add helper macros to declare struct soc_enum instances Several shortcuts for popular uses of some of SOC_ENUM_* and SOC_VALUE_ENUM_* macros. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-21 22:04:03 +01:00
ASoC: core - Allow components to probe/remove in sequence. Some ASoC components depend on other ASoC components to provide clocks and power resources in order to probe() and vice versa for remove(). Allow components to be ordered so that components can be probed() and removed() in sequences that conform to their dependencies. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-07 16:08:05 +01:00
ASoC: add for_each_comp_order() macro To be more readable code, this patch adds new for_each_comp_order() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-18 01:30:41 +00:00
[ALSA] ASoC: Clarify API for bias configuration Currently the ASoC core configures the bias levels in the system using a callback on codecs and machines called 'dapm_event', passing it PCI style power levels as SNDRV_CTL_POWER_ constants. This is more obscure than it needs to be and has caused confusion to driver authors, especially given that DAPM is also performing power management. Address this by renaming the callback function to 'set_bias_level' and using constants explicitly representing the off, standby, pre-on and on states which DAPM transitions through. Also unexport the API for setting bias level: there are currently no in-tree users of this API other than the core itself and it is likely that the core would need to be extended to cater for any users. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-05-19 12:31:28 +02:00
ASoC: Specify target bias state directly as a bias state Rather than a simple flag to say if we want the DAPM context to be at full power specify the target bias state. This should have no current effect but is a bit more direct and so makes it easier to change our decisions about the which bias state to go into in future. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-04 11:25:10 +01:00
[ALSA] ASoC: Clarify API for bias configuration Currently the ASoC core configures the bias levels in the system using a callback on codecs and machines called 'dapm_event', passing it PCI style power levels as SNDRV_CTL_POWER_ constants. This is more obscure than it needs to be and has caused confusion to driver authors, especially given that DAPM is also performing power management. Address this by renaming the callback function to 'set_bias_level' and using constants explicitly representing the off, standby, pre-on and on states which DAPM transitions through. Also unexport the API for setting bias level: there are currently no in-tree users of this API other than the core itself and it is likely that the core would need to be extended to cater for any users. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-05-19 12:31:28 +02:00
ASoC: Allow DAI links to be specified using device tree nodes DAI link endpoints and platform (DMA) devices are currently specified by name. When instantiating sound cards from device tree, it may be more convenient to refer to these devices by phandle in the device tree, and for code to describe DAI links using the "struct device_node *" ("of_node") those phandles map to. This change adds new fields to snd_soc_dai_link which can "name" devices using of_node, enhances soc_bind_dai_link() to allow binding based on of_node, and enhances snd_soc_register_card() to ensure that illegal combinations of name and of_node are not used. Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-12-21 10:40:59 -07:00
ASoC: Add jack reporting interface This patch adds a jack reporting interface to ASoC. This wraps the ALSA core jack detection functionality and provides integration with DAPM to automatically update the power state of pins based on the jack state. Since embedded platforms can have multiple detecton methods used for a single jack (eg, separate microphone and headphone detection) the report function allows specification of which bits are being updated on a given report. The expected usage is that machine drivers will create jack objects and then configure jack detection methods to update that jack. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-07 17:31:10 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ALSA: asoc: core - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch series merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai in preparation for further ASoC v2 patches. This merger removes duplication in both DAI structures and simplifies the API for other users. Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-07-07 16:07:17 +01:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: Pass dai_link as argument to platform suspend and resume Passing pointer to relevant dai_link provides easier reach to the ASoC tree in suspend/resume of snd_soc_platform. It also provides direct access to the dai at the other end of the dai_link. Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-02-22 15:58:04 +09:00
ASoC: add snd_soc_register_component() Current ASoC has register function for platform/codec/dai/card, but doesn't have for cpu. It often produces confusion and fault on ASoC. As result of ASoC community discussion, we consider new struct snd_soc_component for CPU/CODEC, and will switch over to use it. This patch adds very basic struct snd_soc_component, and register function for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Reviewed-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-03-11 18:27:21 -07:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: Add jack reporting interface This patch adds a jack reporting interface to ASoC. This wraps the ALSA core jack detection functionality and provides integration with DAPM to automatically update the power state of pins based on the jack state. Since embedded platforms can have multiple detecton methods used for a single jack (eg, separate microphone and headphone detection) the report function allows specification of which bits are being updated on a given report. The expected usage is that machine drivers will create jack objects and then configure jack detection methods to update that jack. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-07 17:31:10 +00:00
ASoC: add support for multiple jack types This patch adds soc-jack support for adding voltage zones and for detecting jack type Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Harsha Priya <priya.harsha@intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-02-09 14:44:17 +05:30
ASoC: Add jack reporting interface This patch adds a jack reporting interface to ASoC. This wraps the ALSA core jack detection functionality and provides integration with DAPM to automatically update the power state of pins based on the jack state. Since embedded platforms can have multiple detecton methods used for a single jack (eg, separate microphone and headphone detection) the report function allows specification of which bits are being updated on a given report. The expected usage is that machine drivers will create jack objects and then configure jack detection methods to update that jack. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-07 17:31:10 +00:00
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 15:53:46 +02:00
ASoC: dpcm: Add Dynamic PCM core operations. 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>
2012-04-25 12:12:49 +01:00
ASoC: topology: Add FE DAIs dynamically Topology will create FE DAIs dynamically from the PCM objects, and register them to the component. A PCM topoplogy object describes a FE DAI and DAI link. Later patch will add FE DAI links as well. Change tplg load ops for DAI: - Only process a DAI. - Pass the DAI driver pointer to the component driver for extra initialization. Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-01-15 16:13:28 +08:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: Add GPIO support for jack reporting interface Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpios, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:25:04 -06:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: core - PCM mutex per rtd In preparation for the new ASoC Dynamic PCM support (AKA DSP support). The new ASoC Dynamic PCM core allows DAIs to be dynamically re-routed at runtime between the PCM device end (or Frontend - FE) and the physical DAI (Backend - BE) using regular kcontrols (just like a hardware CODEC routes audio in the analog domain). The Dynamic PCM core therefore must be able to call PCM operations for both the Frontend and Backend(s) DAIs at the same time. Currently we have a global pcm_mutex that is used to serialise the ASoC PCM operations. This patch removes the global mutex and adds a mutex per RTD allowing the PCM operations to be reentrant and allow control of more than one DAI at at time. e.g. a frontend PCM hw_params() could configure multiple backend DAI hw_params() with similar or different hw parameters at the same time. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-09 17:04:39 +01:00
ASoC: core: Add card mutex locking subclasses This is the first part of a change that is intended to improve ASoC locking protection for DAPM and PCM operations. This part of the series adds a mutex class for the soc_card mutex. The SND_SOC_CARD_CLASS_INIT class is used for card initialisation only whilst the SND_SOC_CARD_CLASS_PCM class is used for the forth coming Dynamic PCM operations. The new mutex classes are required otherwise we will see a false positive mutex deadlock warning between the card initialisation and the PCM operations (something that would never deadlock in real life). Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-03-07 10:38:25 +00:00
ASoC: core: Rename card mutex subclass to better align with usage Change SND_SOC_CARD_CLASS_PCM to SND_SOC_CARD_CLASS_RUNTIME to better describe all uses for this mutex subclass and align with DAPM too. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-03-09 12:02:07 +00:00
ASoC: core: Add card mutex locking subclasses This is the first part of a change that is intended to improve ASoC locking protection for DAPM and PCM operations. This part of the series adds a mutex class for the soc_card mutex. The SND_SOC_CARD_CLASS_INIT class is used for card initialisation only whilst the SND_SOC_CARD_CLASS_PCM class is used for the forth coming Dynamic PCM operations. The new mutex classes are required otherwise we will see a false positive mutex deadlock warning between the card initialisation and the PCM operations (something that would never deadlock in real life). Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-03-07 10:38:25 +00:00
ASoC: soc core allow machine driver to register the card The machine driver can't register the card directly and need to do this thru soc-audio device creation This patch allows the register and unregister card to be directly called by machine drivers 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>
2011-01-14 19:22:48 +05:30
ASoC: core: Add devm_snd_soc_register_card() Simplify error handling and remove repetitive (and rarely executed) code for unregistration by providing a devm_snd_soc_register() card. Signed-off-by: Mark Brown <broonie@linaro.org> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com>
2013-09-16 18:02:05 +01:00
ASoC: add static inline funcs to fix a compiling issue When CONFIG_PM_SLEEP is not selected, calling funcs snd_soc_suspend and _resume will generate a compiling issue. Here add static inline stub functions to fix it. Signed-off-by: Jie Yang <yang.jie@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-04-19 09:45:48 +08:00
ASoC: Export card PM callbacks for use in direct registered cards 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>
2011-01-26 14:59:27 +00:00
ASoC: add static inline funcs to fix a compiling issue When CONFIG_PM_SLEEP is not selected, calling funcs snd_soc_suspend and _resume will generate a compiling issue. Here add static inline stub functions to fix it. Signed-off-by: Jie Yang <yang.jie@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-04-19 09:45:48 +08:00
ASoC: Export card PM callbacks for use in direct registered cards 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>
2011-01-26 14:59:27 +00:00
ASoC: soc-core: add snd_soc_add_component() ALSA SoC platform/codec will be replaced to component soon. But, some function exist in "platform" doesn't exist in "component". Current soc-core has snd_soc_register_component(), but doesn't have snd_soc_add_component() like snd_soc_add_platform(). This patch adds it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-10-02 05:10:17 +00:00
ASoC: add snd_soc_register_component() Current ASoC has register function for platform/codec/dai/card, but doesn't have for cpu. It often produces confusion and fault on ASoC. As result of ASoC community discussion, we consider new struct snd_soc_component for CPU/CODEC, and will switch over to use it. This patch adds very basic struct snd_soc_component, and register function for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Reviewed-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-03-11 18:27:21 -07:00
ASoC: soc-core: rename "cmpnt" to "component" To unify notation, to readable. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-08-07 02:06:23 +00:00
ASoC: add snd_soc_register_component() Current ASoC has register function for platform/codec/dai/card, but doesn't have for cpu. It often produces confusion and fault on ASoC. As result of ASoC community discussion, we consider new struct snd_soc_component for CPU/CODEC, and will switch over to use it. This patch adds very basic struct snd_soc_component, and register function for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Reviewed-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-03-11 18:27:21 -07:00
ASoC: core: Implement devm_snd_soc_register_component() Since with the wider use of devres many drivers are now only calling snd_soc_unregister_component() in their remove functions providing a managed version will save a reasonable amount of code. Signed-off-by: Mark Brown <broonie@linaro.org>
2013-09-04 20:37:34 +01:00
ASoC: soc-core: rename "cmpnt" to "component" To unify notation, to readable. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-08-07 02:06:23 +00:00
ASoC: core: Implement devm_snd_soc_register_component() Since with the wider use of devres many drivers are now only calling snd_soc_unregister_component() in their remove functions providing a managed version will save a reasonable amount of code. Signed-off-by: Mark Brown <broonie@linaro.org>
2013-09-04 20:37:34 +01:00
ASoC: add snd_soc_register_component() Current ASoC has register function for platform/codec/dai/card, but doesn't have for cpu. It often produces confusion and fault on ASoC. As result of ASoC community discussion, we consider new struct snd_soc_component for CPU/CODEC, and will switch over to use it. This patch adds very basic struct snd_soc_component, and register function for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Reviewed-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-03-11 18:27:21 -07:00
ASoC: soc-core: add component lookup functions ALSA SoC platform/codec will be replaced to component soon. This means 1 device might have multiple components. But current unregister component function only checks "dev" to find it. This means, unexpected component might be unregistered by current function. But, it is no problem if driver registered only 1 component. To prepare avoid this issue, this patch adds new component lookup function. it finds component by "dev" and "driver name". Here, the reason why it uses "driver name" is that "component name" was created by fmt_single_name() and difficult to use it from driver. Driver of course knows its "driver name", thus, using it is more easy. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-10-02 05:09:52 +00:00
ASoC: Remove snd_soc_cache_sync() implementation This function has no more non regmap user, which means we can remove the implementation of the function and associated functions and structure fields. For convenience we keep a static inline version of the function that forwards calls to regcache_sync() unconditionally. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2014-11-08 16:38:07 +01:00
ASoC: Declare soc_new_pcm() properly Ensure that everything is seeing the same declaration by moving it to a header file rather than putting the declaration in soc-core.c Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-12-22 12:16:39 +00:00
ASoC: compress: add config item for soc-compress to make it compiled only when needed We don't always need soc-compress in soc, here add a config item SND_SOC_COMPRESS, when nobody select it, the soc-compress will not be compiled. Here also change Kconfig to 'select SND_SOC_COMPRESS' for drivers that needed soc-compress. Signed-off-by: Jie Yang <yang.jie@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-13 23:41:00 +08:00
ASoC: fix 0-day warnings with snd_soc_new_compress() All conditionally-defined routines in include/sound/soc.h expose a static inline fallback to avoid 0-day warnings and compilation issues, except snd_soc_new_compress(). Fixes: 5db6aab6f36f ('ASoC: topology: Add support for compressed PCMs') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-05-29 18:30:02 -05:00
ASoC: compress: add config item for soc-compress to make it compiled only when needed We don't always need soc-compress in soc, here add a config item SND_SOC_COMPRESS, when nobody select it, the soc-compress will not be compiled. Here also change Kconfig to 'select SND_SOC_COMPRESS' for drivers that needed soc-compress. Signed-off-by: Jie Yang <yang.jie@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-13 23:41:00 +08:00
ASoC: Add platform registration API ASoC v2 allows platform drivers to instantiate independantly of the overall ASoC card. This API allows drivers to notify the core when they are registered. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-03 19:40:30 +00:00
ASoC: add snd_soc_disconnect_sync() Now, we have snd_card_disconnect_sync() on ALSA framework. snd_soc_disconnect_sync() is ASoC version of it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-11-28 06:27:09 +00:00
ASoC: dpcm: Add API for DAI link substream and runtime lookup 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>
2012-04-25 12:12:53 +01:00
ASoC: Handle ignore_pmdown_time for CODEC to CODEC links For CODEC to CODEC links we should only immediately power down if both CODECs are configured to ignore the power down delay. Factor the logic for this into a helper function that can be used for both compressed and normal PCMs. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-03-05 13:17:42 +01:00
ASoC: Add helper functions for PCM runtime 'active' management We have the same code that increments and decrements the active field of the various PCM runtime components (all with the same bugs). Factor this out into common helper functions. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-03-05 13:17:43 +01:00
ASoC: Handle ignore_pmdown_time for CODEC to CODEC links For CODEC to CODEC links we should only immediately power down if both CODECs are configured to ignore the power down delay. Factor the logic for this into a helper function that can be used for both compressed and normal PCMs. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-03-05 13:17:42 +01:00
ASoC: Add helper function for changing the DAI link format For some setups it is necessary to change the DAI link format at runtime. This patch factors out the code that does the initial static DAI link format configuration into a separate helper function which can be used board drivers as well. This allows board drivers that have to change the DAI link format at runtime to reuse it instead of having to manually change the format on all DAIs. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-01-06 15:17:20 +01:00
ASoC: Provide a dummy wrapper of snd_soc_set_dmi_name() For systems without DMI, it makes no sense to have the code. Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Acked-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-24 08:54:41 +02:00
ASoC: core: Add API to use DMI name in sound card long name Intel DSP platform drivers are used by many different devices but are difficult for userspace to differentiate. This patch adds an API to allow the DMI name to be used in the sound card long name, thereby helping userspace load the correct UCM configuration. Usually machine drivers uses their own name as the sound card name (short name), and leave the long name and driver name blank. This API will use the DMI info like vendor, product and board to make up the card long name. If the machine driver has already explicitly set the long name, this API will do nothing. This patch also allows for further differentiation as many devices that share the same DMI name i.e. Minnowboards, UP boards may be configured with different codecs or firmwares. The API supports flavoring the DMI name into the card longname to provide the extra differentiation required for these devices. For Use Case Manager (UCM) in the user space, changing card long name by this API is backward compatible, since the card name does not change. For a given sound card, even if there is no device-specific UCM configuration file that uses the card long name, UCM will fall back to load the default configuration file that uses the card name. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-01-14 16:13:02 +08:00
ASoC: Provide a dummy wrapper of snd_soc_set_dmi_name() For systems without DMI, it makes no sense to have the code. Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Acked-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-04-24 08:54:41 +02:00
ASoC: core: Add API to use DMI name in sound card long name Intel DSP platform drivers are used by many different devices but are difficult for userspace to differentiate. This patch adds an API to allow the DMI name to be used in the sound card long name, thereby helping userspace load the correct UCM configuration. Usually machine drivers uses their own name as the sound card name (short name), and leave the long name and driver name blank. This API will use the DMI info like vendor, product and board to make up the card long name. If the machine driver has already explicitly set the long name, this API will do nothing. This patch also allows for further differentiation as many devices that share the same DMI name i.e. Minnowboards, UP boards may be configured with different codecs or firmwares. The API supports flavoring the DMI name into the card longname to provide the extra differentiation required for these devices. For Use Case Manager (UCM) in the user space, changing card long name by this API is backward compatible, since the card name does not change. For a given sound card, even if there is no device-specific UCM configuration file that uses the card long name, UCM will fall back to load the default configuration file that uses the card name. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-01-14 16:13:02 +08:00
ASoC: Add bit clock rate calculator utility functions Many devices need to calculate the bit clock rate desired to work out the clock configuration required for the device. Provide utility functions to do this using both hw_params structures and raw numbers. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2009-11-10 16:08:04 +00:00
ASoC: Add BCLK calculation utility for TDM mode too Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-11-19 11:36:10 +00:00
ASoC: Add bit clock rate calculator utility functions Many devices need to calculate the bit clock rate desired to work out the clock configuration required for the device. Provide utility functions to do this using both hw_params structures and raw numbers. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2009-11-10 16:08:04 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: pcm: Add soc_dai_hw_params helper Add a function helper to factorize the hw_params code. Suggested by Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Benoit Cousson <bcousson@baylibre.com> Tested-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-08 23:19:38 +02:00
ASoC: Add jack reporting interface This patch adds a jack reporting interface to ASoC. This wraps the ALSA core jack detection functionality and provides integration with DAPM to automatically update the power state of pins based on the jack state. Since embedded platforms can have multiple detecton methods used for a single jack (eg, separate microphone and headphone detection) the report function allows specification of which bits are being updated on a given report. The expected usage is that machine drivers will create jack objects and then configure jack detection methods to update that jack. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-07 17:31:10 +00:00
ASoC: Allow to register jacks at the card level Jacks are typically card level elements, but are currently registered with a CODEC. When it was originally introduced snd_soc_jack_new() took a snd_soc_card as its parameter, but at that time DAPM was only implemented at the CODEC level and there was only one CODEC per card. This made it clear which CODEC to use for the jack DAPM operations. But the multi-component patchset added support for having multiple CODECs per card and with it the API was updated to register jacks with a specific CODEC instance instead. Subsequently DAPM support at the card level has been introduced, but the snd_soc_jack_new() API has so remained unchanged. This leaves us with the issue that the DAPM pins that are managed by the jack detection logic usually are part of the card DAPM context but are accessed through a CODEC DAPM context. Currently this works fine, but might break in the future if we take a more hierarchical approach to DAPM contexts. Furthermore with componentization progressing systems that do not register a snd_soc_codec might appear, while these system may still want to able to register a jack. This patch addresses these issues by adding a new function called snd_soc_card_jack_new() that can be used to register jacks with the card rather than a CODEC. This new function is mostly identical to snd_soc_jack_new() except that it additionally allows to directly specify the DAPM pins associated with the jack. This was done since most users of snd_soc_jack_new() typically call snd_soc_jack_add_pins() right after it, which is not necessary with the new API and allows to reduce the amount of boiler plate code. The old snd_soc_jack_new() is re-implemented as a wrapper around snd_soc_card_jack_new(). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-03-04 10:33:17 +01:00
ASoC: Add jack reporting interface This patch adds a jack reporting interface to ASoC. This wraps the ALSA core jack detection functionality and provides integration with DAPM to automatically update the power state of pins based on the jack state. Since embedded platforms can have multiple detecton methods used for a single jack (eg, separate microphone and headphone detection) the report function allows specification of which bits are being updated on a given report. The expected usage is that machine drivers will create jack objects and then configure jack detection methods to update that jack. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-07 17:31:10 +00:00
ASoC: Add a notifier for jack status changes Some systems provide both mechanical and electrical detection of jack status changes. On such systems power savings can be achieved by only enabling the electrical detection methods when physical insertion has been detected. Begin supporting such systems by providing a notifier for jack status changes which can be used to trigger any reconfiguration. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-22 12:06:30 +00:00
ASoC: add support for multiple jack types This patch adds soc-jack support for adding voltage zones and for detecting jack type Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Harsha Priya <priya.harsha@intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-02-09 14:44:17 +05:30
ASoC: Add GPIO support for jack reporting interface Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpios, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:25:04 -06:00
ASoC: jack: Add support for GPIO descriptor defined jack pins Allow jack GPIO pins be defined also using GPIO descriptor-based interface in addition to legacy GPIO numbers. This is done by adding two new fields to struct snd_soc_jack_gpio: idx and gpiod_dev. Legacy GPIO numbers are used only when GPIO consumer device gpiod_dev is NULL and otherwise idx is the descriptor index within the GPIO consumer device. New function snd_soc_jack_add_gpiods() is added for typical cases where all GPIO descriptor jack pins belong to same GPIO consumer device. For other cases the caller must set the gpiod_dev in struct snd_soc_jack_gpio before calling snd_soc_jack_add_gpios(). Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-26 14:34:37 +03:00
ASoC: Add GPIO support for jack reporting interface Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpios, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:25:04 -06:00
ASoC: Fix build without CONFIG_GPIOLIB snd_soc_jack_gpio stuff is currently enabled for CONFIG_GPIOLIB explicitly with ifdef, and this causes build errors on some drivers such as: sound/soc/omap/rx51.c:220:33: error: array type has incomplete element type Remove ifdef and provide dummy functions for CONFIG_GPIOLIB=n case instead. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-21 15:55:06 +01:00
ASoC: jack: Fix multiple definition of `snd_soc_jack_add_gpiods' Commit f025d3b9c64e ("ASoC: jack: Add support for GPIO descriptor defined jack pins") caused build error when CONFIG_GPIOLIB is not set: sound/include/sound/soc.h:470: multiple definition of `snd_soc_jack_add_gpiods' sound/soc/soc-core.o:sound/include/sound/soc.h:470: first defined here make[2]: *** [sound/soc/snd-soc-core.o] Error 1 make[2]: *** Waiting for unfinished jobs.... make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Fix this by marking snd_soc_jack_add_gpiods() as static inline in soc.h. Reported-by: kbuild test robot <fengguang.wu@intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-27 10:39:57 +03:00
ASoC: jack: Add support for GPIO descriptor defined jack pins Allow jack GPIO pins be defined also using GPIO descriptor-based interface in addition to legacy GPIO numbers. This is done by adding two new fields to struct snd_soc_jack_gpio: idx and gpiod_dev. Legacy GPIO numbers are used only when GPIO consumer device gpiod_dev is NULL and otherwise idx is the descriptor index within the GPIO consumer device. New function snd_soc_jack_add_gpiods() is added for typical cases where all GPIO descriptor jack pins belong to same GPIO consumer device. For other cases the caller must set the gpiod_dev in struct snd_soc_jack_gpio before calling snd_soc_jack_add_gpios(). Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-26 14:34:37 +03:00
ASoC: Fix build without CONFIG_GPIOLIB snd_soc_jack_gpio stuff is currently enabled for CONFIG_GPIOLIB explicitly with ifdef, and this causes build errors on some drivers such as: sound/soc/omap/rx51.c:220:33: error: array type has incomplete element type Remove ifdef and provide dummy functions for CONFIG_GPIOLIB=n case instead. Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-11-21 15:55:06 +01:00
ASoC: Add GPIO support for jack reporting interface Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpios, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:25:04 -06:00
ASoC: Add jack reporting interface This patch adds a jack reporting interface to ASoC. This wraps the ALSA core jack detection functionality and provides integration with DAPM to automatically update the power state of pins based on the jack state. Since embedded platforms can have multiple detecton methods used for a single jack (eg, separate microphone and headphone detection) the report function allows specification of which bits are being updated on a given report. The expected usage is that machine drivers will create jack objects and then configure jack detection methods to update that jack. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-07 17:31:10 +00:00
ASoC: ac97: replace codec to component Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 0 -> .idle_bias_on = 1 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 To keep compatibilty, this patch adds snd_soc_xxx_ac97_codec() macro. These will be removed when all codec code was removed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-01-29 02:58:25 +00:00
ASoC: ac97: Add support for resetting device before registration AC97 devices need to be initially reset before they can be used. Currently each driver does this on its own. Add support for resetting the device to core in snd_soc_new_ac97_codec(). If the caller supplies a device ID and device ID mask the function will reset the device and verify that it has the correct ID, if it does not a error is returned. This will allow to remove custom code with similar functionality from individual drivers. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-07-21 21:53:01 +02:00
ASoC: ac97: replace codec to component Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 0 -> .idle_bias_on = 1 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 To keep compatibilty, this patch adds snd_soc_xxx_ac97_codec() macro. These will be removed when all codec code was removed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-01-29 02:58:25 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: wm97xx: fix uninitialized regmap pointer problem gcc notices that without either the ac97 bus or the pdata, we never initialize the regmap pointer, which leads to an uninitialized variable access: sound/soc/codecs/wm9712.c: In function 'wm9712_soc_probe': sound/soc/codecs/wm9712.c:666:2: error: 'regmap' may be used uninitialized in this function [-Werror=maybe-uninitialized] Since that configuration is invalid, it's better to return an error here. I tried to avoid adding complexity to the conditions, and turned the #ifdef into a regular if(IS_ENABLED()) check for readability. This in turn requires moving some header file declarations out of an #ifdef. The same code is used in three drivers, all of which I'm changing the same way. Fixes: 2ed1a8e0ce8d ("ASoC: wm9712: add ac97 new bus support") Signed-off-by: Arnd Bergmann <arnd@arndb.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-11-02 16:18:21 +01:00
ASoC: ac97: Support multi-platform AC'97 Currently we can only have a single platform built in with AC'97 support due to the use of a global variable to provide the bus operations. Fix this by making that variable a pointer and having the bus drivers set the operations prior to registering. This is not a particularly good or nice approach but it avoids blocking multiplatform and a real fix involves fixing the fairly deep problems with AC'97 support - we should be converting it to a real bus. Acked-by: Arnd Bergmann <arnd@arndb.de> Reviewed-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-06-26 12:45:59 +01:00
ASoC: core: Generic ac97 link reset functions This patch adds generic ac97 reset functions using pincontrol and gpio parsed from devicetree. Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-19 17:05:55 +02:00
ASoC: ac97: Support multi-platform AC'97 Currently we can only have a single platform built in with AC'97 support due to the use of a global variable to provide the bus operations. Fix this by making that variable a pointer and having the bus drivers set the operations prior to registering. This is not a particularly good or nice approach but it avoids blocking multiplatform and a real fix involves fixing the fairly deep problems with AC'97 support - we should be converting it to a real bus. Acked-by: Arnd Bergmann <arnd@arndb.de> Reviewed-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-06-26 12:45:59 +01:00
ASoC: Move AC'97 support to its own file Currently the AC'97 support is splattered all throughout soc-core.c. Some parts are #ifdef'd some parts are not. This patch moves the AC'97 support to its own file, this should make the code a bit more clearer and also makes it possible to easily not compile it into the kernel when not needed. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2014-11-10 22:41:46 +01:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: dapm: Constify lots of names that are never modified Neater and avoids warnings when used in other places where const strings are desired. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2012-02-16 17:07:42 -08:00
ASoC: Fix prefixing of DAPM controls by factoring prefix into snd_soc_cnew() Currently will ignore prefixes when creating DAPM controls. Since currently all control creation goes through snd_soc_cnew() we can fix this by factoring the prefixing into that function. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-03-08 17:23:24 +00:00
ASoC: core: Add snd_soc_card_get_kcontrol() This is useful for drivers who want to grab a pointer to snd_kcontrol outside of the kcontrol callbacks. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-07-29 13:51:58 +01:00
ASoC: Add function to register component controls We have now everything in place to actual let a component register controls. Add a function which allows to do so. Also update snd_soc_add_codec_controls() and snd_soc_platform_controls() to use this new function internally. And while we are at it also change the num_controls parameter of those two functions from int to unsigned int. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-17 22:01:08 +02:00
ASoC: core: Add support for DAI and machine kcontrols. Currently ASoC can only add kcontrols using codec and platform component device handles. It's also desirable to add kcontrols for DAIs (i.e. McBSP) and for SoC card machine drivers too. This allows the kcontrol to have a direct handle to the parent ASoC component DAI/SoC Card/Platform/Codec device and hence easily get it's private data. This change makes snd_soc_add_controls() static and wraps it in the folowing calls (card and dai are new) :- snd_soc_add_card_controls() snd_soc_add_codec_controls() snd_soc_add_dai_controls() snd_soc_add_platform_controls() This patch also does a lot of small mechanical changes in individual codec drivers to replace snd_soc_add_controls() with snd_soc_add_codec_controls(). It also updates the McBSP DAI driver to use snd_soc_add_dai_controls(). Finally, it updates the existing machine drivers that register controls to either :- 1) Use snd_soc_add_card_controls() where no direct codec control is required. 2) Use snd_soc_add_codec_controls() where there is direct codec control. In the case of 1) above we also update the machine drivers to get the correct component data pointers from the kcontrol (rather than getting the machine pointer via the codec pointer). Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-02-03 17:43:09 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: Add info callback for SX_TLV controls SX_TLV controls are intended for situations where the register behind the control has some non-zero value indicating the minimum gain and then gains increasing from there and eventually overflowing through zero. Currently every CODEC implementing these controls specifies the minimum as the non-zero value for the minimum and the maximum as the number of gain settings available. This means when the info callback subtracts the minimum value from the maximum value to calculate the number of gain levels available it is actually under reporting the available levels. This patch fixes this issue by adding a new snd_soc_info_volsw_sx callback that does not subtract the minimum value. Fixes: 1d99f2436d0d ("ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV") Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Acked-by: Brian Austin <brian.austin@cirrus.com> Tested-by: Brian Austin <brian.austin@cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org> Cc: stable@vger.kernel.org
2015-10-14 13:31:24 +01:00
[ALSA] ASoC: build fix for snd_soc_info_bool_ext I suspect that snd_ctl_boolean_mono should have been snd_ctl_boolean_mono_info instead. This fixes the build for magician. Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-05-13 14:03:40 +02:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: fix codec breakage caused by the volsw/volsw_2r merger By accident few places still uses the _2r calls from the core. This is a quick fix, the drivers using the old callbacks going to be changed. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-06 07:43:21 +03:00
ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV Some codecs namely Cirrus Logic Codecs have a way of wrapping the dB scale around 0dB without 0dB being in the middle. Rework of SOC_DOUBLE_R_SX_TLV to be more consistent with other asoc tlv macros. Add single register macro : SOC_SINGLE_SX_TLV. Use snd_soc_info_volsw for .info Use snd_soc_get_volsw_sx, snd_soc_put_volsw_sx for single and double. kcontrols for CS42L51 and CS42L73 are adjusted to these new TLV Macros. The max value is determined by: (number of steps) +1 for 0dB +max from codec datasheet. Signed-off-by: Brian Austin <brian.austin@cirrus.com> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-03-30 10:43:55 -05:00
ASoC: core: Add single controls with specified range of values Control type added for cases where a specific range of values within a register are required for control. Added convenience macros: SOC_SINGLE_RANGE SOC_SINGLE_RANGE_TLV Added accessor implementations: snd_soc_info_volsw_range snd_soc_put_volsw_range snd_soc_get_volsw_range Signed-off-by: Michal Hajduk <Michal.Hajduk@diasemi.com> Signed-off-by: Adam Thomson <Adam.Thomson@diasemi.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-31 15:18:01 +01:00
ASoC: Let snd_soc_limit_volume() take a snd_soc_card snd_soc_limit_volume() operates on a card and the CODEC that is passed in is only used to look up the card. Let it directly take the card instead. This makes it possible to use it when no snd_soc_codec is available. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Tested-by: Jarkko Nikula <jarkko.nikula@bitmer.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-10-18 17:04:33 +02:00
ASoC: core: Support for limiting the volume Add support for the core to limit the maximum volume on an existing control. The function will modify the soc_mixer_control.max value of the given control. The new value must be lower than the original one (chip maximum) If there is a need for limiting a gain on a given control, than machine drivers can do the following in their snd_soc_dai_link.init function: snd_soc_limit_volume(codec, "TPA6140A2 Headphone Playback Volume", 21); This will modify the original 31 (chip maximum) to 21, so user space will not be able to set the gain higher than this. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-07 14:05:49 +03:00
ASoC: core: Add SND_SOC_BYTES control for coefficient blocks Allow devices to export blocks of registers to the application layer, intended for use for reading and writing coefficient data which can't usefully be worked with by the kernel at runtime (for example, due to requiring complex and expensive calculations or being the results of callibration procedures). Currently drivers are using platform data to provide configurations for coefficient blocks which isn't at all convenient for runtime management or configuration development. Currently only devices using regmap are supported, an error will be generated for any attempt to work with a byte control on a non-regmap device. There's no fundamental block to other devices so support could be added if required. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-10-10 18:31:26 +01:00
ASoC: add SND_SOC_BYTES_EXT we need _EXT version for SND_SOC_BYTES so that DSPs can use this to pass data for DSP modules Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02 10:58:11 +05:30
ASoC: core: add a helper for extended byte controls using TLV ALSA supports arbitrary length TLVs for each kcontrol that can be used to pass metadata about the control (e.g. volumes, enum information). The same transport mechanism is now used for arbitrary length data by defining a new helper. Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah@intel.com> Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-15 21:34:48 +05:30
ASoC: core: Add signed multi register control 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>
2012-04-20 11:32:13 +02:00
ASoC: core: Add strobe control 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>
2012-04-20 11:32:44 +02:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: Add jack reporting interface This patch adds a jack reporting interface to ASoC. This wraps the ALSA core jack detection functionality and provides integration with DAPM to automatically update the power state of pins based on the jack state. Since embedded platforms can have multiple detecton methods used for a single jack (eg, separate microphone and headphone detection) the report function allows specification of which bits are being updated on a given report. The expected usage is that machine drivers will create jack objects and then configure jack detection methods to update that jack. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-07 17:31:10 +00:00
ASoC: Clean up docbook warnings A number of functions and structures in the sound subsystem had incomplete and/or obsolete DocBook comments, leading to warnings when the docs were built. Correct those comments so that we can enjoy our audio in the absence of warning noise. Signed-off-by: Jonathan Corbet <corbet@lwn.net> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-08-25 01:14:48 -06:00
ASoC: Add jack reporting interface This patch adds a jack reporting interface to ASoC. This wraps the ALSA core jack detection functionality and provides integration with DAPM to automatically update the power state of pins based on the jack state. Since embedded platforms can have multiple detecton methods used for a single jack (eg, separate microphone and headphone detection) the report function allows specification of which bits are being updated on a given report. The expected usage is that machine drivers will create jack objects and then configure jack detection methods to update that jack. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-07 17:31:10 +00:00
ASoC: add support for multiple jack types This patch adds soc-jack support for adding voltage zones and for detecting jack type Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Harsha Priya <priya.harsha@intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-02-09 14:44:17 +05:30
ASoC: Clean up docbook warnings A number of functions and structures in the sound subsystem had incomplete and/or obsolete DocBook comments, leading to warnings when the docs were built. Correct those comments so that we can enjoy our audio in the absence of warning noise. Signed-off-by: Jonathan Corbet <corbet@lwn.net> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-08-25 01:14:48 -06:00
ASoC: add support for multiple jack types This patch adds soc-jack support for adding voltage zones and for detecting jack type Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Harsha Priya <priya.harsha@intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-02-09 14:44:17 +05:30
ASoC: Add GPIO support for jack reporting interface Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpios, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:25:04 -06:00
ASoC: jack: Add support for GPIO descriptor defined jack pins Allow jack GPIO pins be defined also using GPIO descriptor-based interface in addition to legacy GPIO numbers. This is done by adding two new fields to struct snd_soc_jack_gpio: idx and gpiod_dev. Legacy GPIO numbers are used only when GPIO consumer device gpiod_dev is NULL and otherwise idx is the descriptor index within the GPIO consumer device. New function snd_soc_jack_add_gpiods() is added for typical cases where all GPIO descriptor jack pins belong to same GPIO consumer device. For other cases the caller must set the gpiod_dev in struct snd_soc_jack_gpio before calling snd_soc_jack_add_gpios(). Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-26 14:34:37 +03:00
ASoC: jack: Clarify GPIO descriptor lookup in struct snd_soc_jack_gpio doc Clarify struct snd_soc_jack_gpio documentation for the idx and name fields. Because name is passed as connection ID to gpiod_get_index() when using GPIO descriptor defined jack pins it is not only used as a label in debugfs but also as function name lookup in systems that support functions names for GPIOs. Clarify also idx since the index is within the function of the GPIO consumer device and not within the device itself only. Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-27 13:54:18 +03:00
ASoC: Clean up docbook warnings A number of functions and structures in the sound subsystem had incomplete and/or obsolete DocBook comments, leading to warnings when the docs were built. Correct those comments so that we can enjoy our audio in the absence of warning noise. Signed-off-by: Jonathan Corbet <corbet@lwn.net> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-08-25 01:14:48 -06:00
ASoC: jack: Clarify GPIO descriptor lookup in struct snd_soc_jack_gpio doc Clarify struct snd_soc_jack_gpio documentation for the idx and name fields. Because name is passed as connection ID to gpiod_get_index() when using GPIO descriptor defined jack pins it is not only used as a label in debugfs but also as function name lookup in systems that support functions names for GPIOs. Clarify also idx since the index is within the function of the GPIO consumer device and not within the device itself only. Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-27 13:54:18 +03:00
ASoC: Add GPIO support for jack reporting interface Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpios, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:25:04 -06:00
ASoC: Clean up docbook warnings A number of functions and structures in the sound subsystem had incomplete and/or obsolete DocBook comments, leading to warnings when the docs were built. Correct those comments so that we can enjoy our audio in the absence of warning noise. Signed-off-by: Jonathan Corbet <corbet@lwn.net> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-08-25 01:14:48 -06:00
ASoC: Allow GPIO jack detection to be configured as a wake source Some systems wish to use jacks as wake sources. Provide a wake flag in the GPIO configuration which causes the driver to enable the IRQ as a wake source. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-02-17 16:35:55 -08:00
ASoC: Add kerneldoc for jack_status_check callback Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-02-17 16:41:42 -08:00
ASoC: Add GPIO support for jack reporting interface Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpios, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:25:04 -06:00
ASoC: jack: Add support for GPIO descriptor defined jack pins Allow jack GPIO pins be defined also using GPIO descriptor-based interface in addition to legacy GPIO numbers. This is done by adding two new fields to struct snd_soc_jack_gpio: idx and gpiod_dev. Legacy GPIO numbers are used only when GPIO consumer device gpiod_dev is NULL and otherwise idx is the descriptor index within the GPIO consumer device. New function snd_soc_jack_add_gpiods() is added for typical cases where all GPIO descriptor jack pins belong to same GPIO consumer device. For other cases the caller must set the gpiod_dev in struct snd_soc_jack_gpio before calling snd_soc_jack_add_gpios(). Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-26 14:34:37 +03:00
ASoC: Add GPIO support for jack reporting interface Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpios, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:25:04 -06:00
ASoC: Allow GPIO jack detection to be configured as a wake source Some systems wish to use jacks as wake sources. Provide a wake flag in the GPIO configuration which causes the driver to enable the IRQ as a wake source. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-02-17 16:35:55 -08:00
ASoC: Clean up docbook warnings A number of functions and structures in the sound subsystem had incomplete and/or obsolete DocBook comments, leading to warnings when the docs were built. Correct those comments so that we can enjoy our audio in the absence of warning noise. Signed-off-by: Jonathan Corbet <corbet@lwn.net> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-08-25 01:14:48 -06:00
ASoC: Add GPIO support for jack reporting interface Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpios, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:25:04 -06:00
ASoC: Use delayed work for debounce of GPIO based jacks Rather than block the workqueue by sleeping to do the debounce use delayed work to implement the debounce time. This should also means that we extend the debounce time on each new bounce, potentially allowing shorter debounce times for clean insertions. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-10-06 15:54:28 -07:00
ASoC: jack - check status of GPIO-based pins on resume For GPIO-backed pins that are not configured as wakeup sources, we may miss change in their state that happens while system is suspended. Let's use PM notifier to refresh their state upon resume. Signed-off-by: Dmitry Torokhov <dtor@chromium.org> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-03-21 16:50:43 -07:00
ASoC: jack: Basic GPIO descriptor conversion This patch does basic GPIO descriptor conversion to soc-jack. Even the GPIOs are still passed and requested using legacy GPIO numbers the driver internals are converted to use GPIO descriptor API. Motivation for this is to prepare soc-jack so that it will allow registering jack GPIO pins using both GPIO descriptors and legacy GPIO numbers. Signed-off-by: Jarkko Nikula <jarkko.nikula@linux.intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-26 14:34:36 +03:00
ASoC: Add jack_status_check callback function for GPIO jacks The jack_status_check callback function is the interface to check the status of the jack. Some target provides the method to distinguish what is the jack inserted - headphone jack, microphone jack, tvout jack, etc, so we can implement it using the jack_status_check function. Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-11-12 17:14:04 +09:00
ASoC: add data field into snd_soc_jack_gpio so callback could get the context data as needed Signed-off-by: xiangxiao <xiaoxiang@xiaomi.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-23 14:40:44 +08:00
ASoC: Clean up docbook warnings A number of functions and structures in the sound subsystem had incomplete and/or obsolete DocBook comments, leading to warnings when the docs were built. Correct those comments so that we can enjoy our audio in the absence of warning noise. Signed-off-by: Jonathan Corbet <corbet@lwn.net> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-08-25 01:14:48 -06:00
ASoC: add data field into snd_soc_jack_gpio so callback could get the context data as needed Signed-off-by: xiangxiao <xiaoxiang@xiaomi.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-23 14:40:44 +08:00
ASoC: Add GPIO support for jack reporting interface Add GPIO support to jack reporting framework in ASoC using gpiolib calls. The gpio support exports two new functions: snd_soc_jack_add_gpios and snd_soc_jack_free_gpios. Client drivers using gpio feature must pass an array of jack_gpio pins belonging to a specific jack to the snd_soc_jack_add_gpios function. The framework will request the gpios, set the data direction and request irq. The framework will update power status of related jack_pins when an event on the gpio pins comes according to the reporting bits defined for each gpio. All gpio resources allocated when adding jack_gpio pins can be released using snd_soc_jack_free_gpios function. Signed-off-by: Misael Lopez Cruz <x0052729@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-03 15:25:04 -06:00
ASoC: Add jack reporting interface This patch adds a jack reporting interface to ASoC. This wraps the ALSA core jack detection functionality and provides integration with DAPM to automatically update the power state of pins based on the jack state. Since embedded platforms can have multiple detecton methods used for a single jack (eg, separate microphone and headphone detection) the report function allows specification of which bits are being updated on a given report. The expected usage is that machine drivers will create jack objects and then configure jack detection methods to update that jack. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-07 17:31:10 +00:00
ASoC: jack: Push locking for jacks down to the jack Currently operations on jack reporting take the CODEC mutex both to protect the current jack status and also to protect the DAPM run which is triggered on status updates. Since the addition of a DAPM-specific lock we no longer need to worry about locking DAPM as it has its own finer grained lock so create a per jack lock to take care of the jack status. This is both cleaner where the jack isn't specifically associated with a CODEC and clearer as it's much more obvious what the lock is protecting. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-03-12 14:07:49 +00:00
ASoC: Add jack reporting interface This patch adds a jack reporting interface to ASoC. This wraps the ALSA core jack detection functionality and provides integration with DAPM to automatically update the power state of pins based on the jack state. Since embedded platforms can have multiple detecton methods used for a single jack (eg, separate microphone and headphone detection) the report function allows specification of which bits are being updated on a given report. The expected usage is that machine drivers will create jack objects and then configure jack detection methods to update that jack. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-07 17:31:10 +00:00
ASoC: Allow to register jacks at the card level Jacks are typically card level elements, but are currently registered with a CODEC. When it was originally introduced snd_soc_jack_new() took a snd_soc_card as its parameter, but at that time DAPM was only implemented at the CODEC level and there was only one CODEC per card. This made it clear which CODEC to use for the jack DAPM operations. But the multi-component patchset added support for having multiple CODECs per card and with it the API was updated to register jacks with a specific CODEC instance instead. Subsequently DAPM support at the card level has been introduced, but the snd_soc_jack_new() API has so remained unchanged. This leaves us with the issue that the DAPM pins that are managed by the jack detection logic usually are part of the card DAPM context but are accessed through a CODEC DAPM context. Currently this works fine, but might break in the future if we take a more hierarchical approach to DAPM contexts. Furthermore with componentization progressing systems that do not register a snd_soc_codec might appear, while these system may still want to able to register a jack. This patch addresses these issues by adding a new function called snd_soc_card_jack_new() that can be used to register jacks with the card rather than a CODEC. This new function is mostly identical to snd_soc_jack_new() except that it additionally allows to directly specify the DAPM pins associated with the jack. This was done since most users of snd_soc_jack_new() typically call snd_soc_jack_add_pins() right after it, which is not necessary with the new API and allows to reduce the amount of boiler plate code. The old snd_soc_jack_new() is re-implemented as a wrapper around snd_soc_card_jack_new(). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-03-04 10:33:17 +01:00
ASoC: Add jack reporting interface This patch adds a jack reporting interface to ASoC. This wraps the ALSA core jack detection functionality and provides integration with DAPM to automatically update the power state of pins based on the jack state. Since embedded platforms can have multiple detecton methods used for a single jack (eg, separate microphone and headphone detection) the report function allows specification of which bits are being updated on a given report. The expected usage is that machine drivers will create jack objects and then configure jack detection methods to update that jack. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-07 17:31:10 +00:00
ASoC: Add a notifier for jack status changes Some systems provide both mechanical and electrical detection of jack status changes. On such systems power savings can be achieved by only enabling the electrical detection methods when physical insertion has been detected. Begin supporting such systems by providing a notifier for jack status changes which can be used to trigger any reconfiguration. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-22 12:06:30 +00:00
ASoC: add support for multiple jack types This patch adds soc-jack support for adding voltage zones and for detecting jack type Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Harsha Priya <priya.harsha@intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-02-09 14:44:17 +05:30
ASoC: Add jack reporting interface This patch adds a jack reporting interface to ASoC. This wraps the ALSA core jack detection functionality and provides integration with DAPM to automatically update the power state of pins based on the jack state. Since embedded platforms can have multiple detecton methods used for a single jack (eg, separate microphone and headphone detection) the report function allows specification of which bits are being updated on a given report. The expected usage is that machine drivers will create jack objects and then configure jack detection methods to update that jack. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-07 17:31:10 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
[ALSA] soc - 0.13 ASoC headers This patch updates the API's to include the new DAI configuration and clocking architecture. Changes:- o Removed DAI automatic matching and capabilities structure (struct snd_soc_dai_mode) and macros. o Added DAI operations for codec and CPU interfaces. o Removed config_sysclk() function and struct snd_soc_clock_info. No longer needed as clocking is now configured manually in the machine drivers. Also removed other clocking data from structures. o Updated version to 0.13 o Added shift to SOC_SINGLE_EXT kcontrol macro. Signed-off-by: Graeme Gregory <gg@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2007-02-02 17:13:05 +01:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: Allow drivers to specify how many bits are significant on a DAI Most devices accept data in formats that don't correspond directly to their internal format. ALSA allows us to set a msbits constraint which tells userspace about this in case it finds it useful (for example, in order to avoid wasting effort dithering bits that will be ignored when raising the sample size of data) so provide a mechanism for drivers to specify the number of bits that are actually significant on a DAI and add the appropriate constraints along with all the others. This is done slightly awkwardly as the constraint is specified per sample size - we loop over every possible sample size, including ones that the device doesn't support and including ones that have fewer bits than are actually used, but this is harmless as the upper layers do the right thing in these cases. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2012-01-16 18:38:51 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: add definations for compressed operations Here we update the asoc structures to add compress stream definations First the struct snd_soc_dai_driver adds a new member to indicate if the dai is compressed or pcm. Next we add a new structre the struct snd_soc_compr_ops in the struct snd_soc_dai_link. This is to be used for machine driver to perform any opertaions required for setting up compressed audio streams next is the compressed data operations, they are added using struct snd_compr_ops in the struct snd_soc_platform_driver. Signed-off-by: Namarta Kohli <namartax.kohli@intel.com> Signed-off-by: Ramesh Babu K V <ramesh.babu@intel.com> Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-16 17:10:40 +05:30
ASoC: snd_soc_codec includes snd_soc_component Codec includes component by this patch, and component moved to upside of codec to avoid extra declaration. Codec dai will be registered via component by this patch. Current component register function is used for cpu, and it is using dai/dais functions properly to keep existing cpu dai name. And now, it will be used from codec also. But codec driver had been used dais function only even though it was single dai. This patch adds new flag which can selects dai/dais function on component register function to keep existing codec dai name. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-09-04 19:39:03 -07:00
ASoC: add .of_xlate_dai_name on snd_soc_component_driver ASoC sound driver requires CPU/CODEC drivers for probing, and each CPU/CODEC has some DAI on it. Then, "dai name matching" have been used to identify CPU-CODEC DAI pair on ASoC. But, the "dai port number matching" is now required from DeviceTree. The solution of this issue is to replace the dai port number into dai name. Now, CPU/CODEC are based on struct snd_soc_component, and it can care above as common issue. This patch adds .of_xlate_dai_name callback interface on struct snd_soc_component_driver, and snd_soc_of_get_dai_name() which is using .of_xlate_dai_name. Then, #sound-dai-cells which enables DAI specifier is required on CPU/CODEC device tree properties. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-09-10 17:39:56 -07:00
ASoC: Add component level probe/remove support Now that we have a unified probe and remove path make sure to call them for all components. soc_{probe,remove}_component are responsible for setting up the DAPM context for the component, initialize the component prefix, manage the debugfs entries as well as do the registration of table based controls and DAPM elements. They also call the component drivers probe and remove callbacks. This patch makes these things available for generic snd_soc_component drivers rather than only having them for snd_soc_codec and snd_soc_platform drivers. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-19 15:51:21 +02:00
ASoC: add Component level suspend/resume In current ALSA SoC, Codec only has suspend/resume feature, but it should be supported on Component level. This patch adds it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-11-30 06:23:13 +00:00
ASoC: Add component level probe/remove support Now that we have a unified probe and remove path make sure to call them for all components. soc_{probe,remove}_component are responsible for setting up the DAPM context for the component, initialize the component prefix, manage the debugfs entries as well as do the registration of table based controls and DAPM elements. They also call the component drivers probe and remove callbacks. This patch makes these things available for generic snd_soc_component drivers rather than only having them for snd_soc_codec and snd_soc_platform drivers. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-19 15:51:21 +02:00
ASoC: add Component level .read/.write In current ALSA SoC, Codec only has .read/.write callback. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-01-16 02:00:59 +00:00
ASoC: add Component level pcm_new/pcm_free v2 In current ALSA SoC, Platform only has pcm_new/pcm_free feature, but it should be supported on Component level. This patch adds it. The v1 was added commit 99b04f4c4051f7 ("ASoC: add Component level pcm_new/pcm_free") but it called all "card" connected component's pcm_new/free, it was wrong. This patch calls "rtd" connected component. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-09-26 01:00:53 +00:00
ASoC: add Component level set_sysclk In current ALSA SoC, Codec only has set_sysclk feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-08-24 00:57:35 +00:00
ASoC: add Component level set_pll In current ALSA SoC, Codec only has set_pll feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-08-24 00:57:51 +00:00
ASoC: add Component level set_jack In current ALSA SoC, Codec only has set_jack feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-08-24 00:58:07 +00:00
ASoC: add Component level set_sysclk In current ALSA SoC, Codec only has set_sysclk feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-08-24 00:57:35 +00:00
ASoC: add .of_xlate_dai_name on snd_soc_component_driver ASoC sound driver requires CPU/CODEC drivers for probing, and each CPU/CODEC has some DAI on it. Then, "dai name matching" have been used to identify CPU-CODEC DAI pair on ASoC. But, the "dai port number matching" is now required from DeviceTree. The solution of this issue is to replace the dai port number into dai name. Now, CPU/CODEC are based on struct snd_soc_component, and it can care above as common issue. This patch adds .of_xlate_dai_name callback interface on struct snd_soc_component_driver, and snd_soc_of_get_dai_name() which is using .of_xlate_dai_name. Then, #sound-dai-cells which enables DAI specifier is required on CPU/CODEC device tree properties. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-09-10 17:39:56 -07:00
ASoC: add snd_soc_get_dai_id() function ALSA SoC needs to know connected DAI ID for detecting. It is not a big problem if device/driver was only for sound, but getting DAI ID will be difficult if device includes both Video/Sound, like HDMI. To solve this issue, this patch adds new snd_soc_get_dai_id() and its related .of_xlate_dai_id callback on component driver. In below case, we can handle Sound port (= port@2) as ID = 0 if .of_xlate_dai_id has its support. hdmi { port@0 { /* VIDEO */ }; port@1 { /* VIDEO */ }; port@2 { /* SOUND */ }; }; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-05-18 01:39:25 +00:00
ASoC: Add component level stream_event() and seq_notifier() support This patch adds stream_event() and seq_notifier() callbacks similar to those found in the snd_soc_codec_driver and snd_soc_platform driver struct to the snd_soc_component_driver struct. This is meant to unify the handling of these callbacks across different types of components and will eventually allow their removal from the CODEC and platfrom driver structs. The new callbacks are slightly different from the old ones in that they take a snd_soc_component as a parameter rather than a snd_soc_dapm_context. This was done since otherwise casting from the DAPM context to the component would typically be the first thing to do in the callback. And the interface becomes slightly cleaner by passing a snd_soc_component to all callbacks in the snd_soc_component_driver struct. The patch also already removes the stream_event() callback from the snd_soc_codec_driver and snd_soc_platform_driver structs as it is currently unused. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-16 18:13:08 +02:00
ASoC: add Component level set_bias_level In current ALSA SoC, Codec only has set_bias_level feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Codec driver has .idle_bias_off for dapm bias. But Component driver doesn't have it, and dapm->idle_bias_off is set as "true". To keep compatibility, this patch adds "idle_bias_on" instead of ".idle_bias_off" on Component driver. dapm->idle_bias_off will be set by inverted idle_bias_on. When we replace Codec to Component, the driver which has ".idle_bias_off = true" is just remove it, and the driver which doesn't have it will have new ".idle_bias_on = true". Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-09-26 01:01:10 +00:00
ASoC: Consolidate platform and CODEC probe/remove The platform and CODEC probe and remove code is now largely identical. This patch consolidates it at the component level. The resulting code is slightly larger due to all the boiler plate code setting up the indirection for the table based control and DAPM registration. Once all drivers have been update to no longer use the snd_soc_codec_driver and snd_soc_platform_driver specific fields for this the indirection can be removed again. This patch contains two noteworthy hacks that are only meant to be temporary to be able to update drivers and the core in separate incremental patches. The first hack is related to that some DPCM platforms expect that the DAPM widgets for the DAIs of a snd_soc_component are created in the DAPM context of the snd_soc_platform that has the same parent device. For handling this the steal_sibling_dai_widgets attribute is introduced. It gets set for snd_soc_platforms that register DAPM elements. When creating the DAI widgets for a component this flag is checked and if it is found on one of the siblings the component will not create any DAI widgets in its own DAPM context. If the attribute is set on a platform it will look for siblings components and create DAI widgets for them in its own context. The fix for this will be to update the offending drivers to only register a single component rather than two. The second hack deals with the fact that the ASoC card suspend and resume code still needs a list of CODECs that have been registered for the card. To handle this the generic probe and remove path have a check to see if the component is CODEC and if yes add/remove it to the card's CODEC list. While it is possible to clean up the suspend/resume code to not need the CODEC list anymore this is a bit of a chicken and egg problem since it will become easier to clean up the suspend/resume code once there is a unified component layer. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-19 15:51:19 +02:00
ASoC: snd_soc_component_driver has snd_pcm_ops Platform will be replaced into Component in the future. snd_soc_platform_driver has snd_pcm_ops, but snd_soc_component_driver doesn't have it. To prepare for replacing, this patch adds snd_pcm_ops on component driver. platform will be replaced into component, and its code will be removed. But during replacing, both platform and component process code exists. To keep compatibility, to avoid platform NULL access and to avoid platform/component duplicate operation during replacing process, this patch has such code. Some of this code will be removed when platform was removed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-10-11 01:37:23 +00:00
ASoC: snd_soc_component_driver has snd_compr_ops Platform will be replaced into Component in the future. snd_soc_platform_driver has snd_compr_ops, but snd_soc_component_driver doesn't have. To prepare for replacing, this patch adds snd_compr_ops on component driver. platform will be replaced into component, and its code will be removed. But during replacing, both platform and component process code exists. To keep compatibility, to avoid platform NULL access and to avoid platform/component duplicate operation during replacing process, this patch has such code. Some of this code will be removed when platform was removed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-10-11 01:37:45 +00:00
ASoC: snd_soc_component_driver has snd_pcm_ops Platform will be replaced into Component in the future. snd_soc_platform_driver has snd_pcm_ops, but snd_soc_component_driver doesn't have it. To prepare for replacing, this patch adds snd_pcm_ops on component driver. platform will be replaced into component, and its code will be removed. But during replacing, both platform and component process code exists. To keep compatibility, to avoid platform NULL access and to avoid platform/component duplicate operation during replacing process, this patch has such code. Some of this code will be removed when platform was removed. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-10-11 01:37:23 +00:00
ASoC: Consolidate platform and CODEC probe/remove The platform and CODEC probe and remove code is now largely identical. This patch consolidates it at the component level. The resulting code is slightly larger due to all the boiler plate code setting up the indirection for the table based control and DAPM registration. Once all drivers have been update to no longer use the snd_soc_codec_driver and snd_soc_platform_driver specific fields for this the indirection can be removed again. This patch contains two noteworthy hacks that are only meant to be temporary to be able to update drivers and the core in separate incremental patches. The first hack is related to that some DPCM platforms expect that the DAPM widgets for the DAIs of a snd_soc_component are created in the DAPM context of the snd_soc_platform that has the same parent device. For handling this the steal_sibling_dai_widgets attribute is introduced. It gets set for snd_soc_platforms that register DAPM elements. When creating the DAI widgets for a component this flag is checked and if it is found on one of the siblings the component will not create any DAI widgets in its own DAPM context. If the attribute is set on a platform it will look for siblings components and create DAI widgets for them in its own context. The fix for this will be to update the offending drivers to only register a single component rather than two. The second hack deals with the fact that the ASoC card suspend and resume code still needs a list of CODECs that have been registered for the card. To handle this the generic probe and remove path have a check to see if the component is CODEC and if yes add/remove it to the card's CODEC list. While it is possible to clean up the suspend/resume code to not need the CODEC list anymore this is a bit of a chicken and egg problem since it will become easier to clean up the suspend/resume code once there is a unified component layer. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-19 15:51:19 +02:00
ASoC: add Component level set_bias_level In current ALSA SoC, Codec only has set_bias_level feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Codec driver has .idle_bias_off for dapm bias. But Component driver doesn't have it, and dapm->idle_bias_off is set as "true". To keep compatibility, this patch adds "idle_bias_on" instead of ".idle_bias_off" on Component driver. dapm->idle_bias_off will be set by inverted idle_bias_on. When we replace Codec to Component, the driver which has ".idle_bias_off = true" is just remove it, and the driver which doesn't have it will have new ".idle_bias_on = true". Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-09-26 01:01:10 +00:00
ASoC: core: conditionally increase module refcount on component open Recently, for Intel platforms the "ignore_module_refcount" field was introduced for the component driver. In order to avoid a deadlock preventing the PCI modules from being removed even when the card was idle, the refcounts were not incremented for the device driver module during component probe. However, this change introduced a nasty side effect: the device driver module can be unloaded while a pcm stream is open. This patch proposes to change the field to be renamed as "module_get_upon_open". When this field is set, the module refcount should be incremented on pcm open amd decremented upon pcm close. This will enable modules to be removed when no PCM playback/capture happens and prevent removal when the component is actually in use. Also, align with the skylake component driver with the new name. Fixes: b450b878('ASoC: core: don't increase component module refcount unconditionally' Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-04-05 09:57:08 -07:00
ASoC: core: don't increase component module refcount unconditionally The ASoC core has for the longest time increased the module reference counts, even before the transition to the component model. This is probably fine on most platforms, but it introduces a deadlock case on Intel devices with the Skylake and SOF drivers which cannot be removed due to their reference counts being modified by the core. In these 2 cases, the PCI or ACPI driver .probe creates a platform device to let the machine driver .probe register the audio card. Conversely the PCI or ACPI driver .remove will unregister the platform device which results in the card being removed by the machine driver .remove. With ascii art, this can be represented as modprobe snd_soc_skl/ soc-pci-dev/sof-acpci-dev ----------> pci/acpi probe ^ | | ---------------| | | | | V V increase register register machine refcount component platform_device ^ | | | | V component <---- register card <---- probe probe The issue is that by playing with the component's module reference counts during the card registration, it's no longer possible to remove the module which controls the component. This can be shown, e.g. with the following error: root@plb-XPS-13-9350:~# lsmod | grep snd_soc_skl snd_soc_skl 110592 1 root@plb-XPS-13-9350:~# rmmod snd_soc_skl rmmod: ERROR: Module snd_soc_skl is in use Increasing the reference count during the component probe is not useful. If the PCI/ACPI module is removed, the card will be removed anyway. To avoid breaking existing platforms and allowing Intel platforms to safely deal with module load/unload cases, this patch introduces a flag which needs to be set during the component initialization. This is a strictly opt-in capability that should only be used when the handling of the component module does not require a reference count increase to prevent removal during use. Note that this solution is not directly applicable to the legacy Atom/SST driver, which uses a different device hierarchy. There are however additional refcount issues which prevent the ACPI driver from being removed. This is a different issue which would need a different patch. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-02-01 11:22:23 -06:00
ASoC: add Component level set_bias_level In current ALSA SoC, Codec only has set_bias_level feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Codec driver has .idle_bias_off for dapm bias. But Component driver doesn't have it, and dapm->idle_bias_off is set as "true". To keep compatibility, this patch adds "idle_bias_on" instead of ".idle_bias_off" on Component driver. dapm->idle_bias_off will be set by inverted idle_bias_on. When we replace Codec to Component, the driver which has ".idle_bias_off = true" is just remove it, and the driver which doesn't have it will have new ".idle_bias_on = true". Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-09-26 01:01:10 +00:00
ASoC: soc-pcm: rename .pmdown_time to .use_pmdown_time for Component commit fbb16563c6c2 ("ASoC: snd_soc_component_driver has pmdown_time") added new .pmdown_time which is for inverted version of current .ignore_pmdown_time But it is confusable name. Let's rename it to .use_pmdown_time Reported-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-01-19 05:21:19 +00:00
ASoC: snd_soc_component_driver has endianness Codec will be replaced into Component, then Codec side only needs to call fixup_codec_formats() at this point. This patch adds new endianness flag on Component driver and call convert_endianness_formats() (= was fixup_codec_format()) if endianness was true. When Codec is replaced into Component, Codec driver needs to have endianness = 1 flags. Existing CPU side of course doesn't have this flag, thus CPU doesn't call it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-10-11 01:38:29 +00:00
ASoC: snd_soc_component_driver has non_legacy_dai_naming Codec will be replaced into Component, then Codec side doesn't use legacy_dai_naming on snd_soc_register_dais(). This patch adds new non_legacy_dai_naming flag on Component driver and use converted its value for snd_soc_register_dais(). When Codec is replaced into Component, Codec driver needs to have non_legacy_dai_naming = 1 flags. Existing CPU side of course doesn't have this flag, thus CPU calls it as true. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-10-11 01:38:51 +00:00
ASoC: core: Allow topology to override machine driver FE DAI link config. Machine drivers statically define a number of DAI links that currently cannot be changed or removed by topology. This means PCMs and platform components cannot be changed by topology at runtime AND machine drivers are tightly coupled to topology. This patch allows topology to override the machine driver DAI link config in order to reuse machine drivers with different topologies and platform components. The patch supports :- 1) create new FE PCMs with a topology defined PCM ID. 2) destroy existing static FE PCMs 3) change the platform component driver. 4) assign any new HW params fixups. 5) assign a new card name prefix to differentiate this topology to userspace. The patch requires no changes to the machine drivers, but does add some platform component flags that the platform component driver can assign before loading topologies. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-02 16:59:54 +01:00
ASoC: snd_soc_codec includes snd_soc_component Codec includes component by this patch, and component moved to upside of codec to avoid extra declaration. Codec dai will be registered via component by this patch. Current component register function is used for cpu, and it is using dai/dais functions properly to keep existing cpu dai name. And now, it will be used from codec also. But codec driver had been used dais function only even though it was single dai. This patch adds new flag which can selects dai/dais function on component register function to keep existing codec dai name. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-09-04 19:39:03 -07:00
ASoC: Move name_prefix from CODEC to component Move the name_prefix from the CODEC struct to the component struct. This will eventually allow to specify prefixes for all types of components. It is also necessary to make the DAPM code component type independent (i.e. a DAPM context does not need to know whether it belongs to a CODEC or a platform or something else). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-16 18:13:01 +02:00
ASoC: snd_soc_codec includes snd_soc_component Codec includes component by this patch, and component moved to upside of codec to avoid extra declaration. Codec dai will be registered via component by this patch. Current component register function is used for cpu, and it is using dai/dais functions properly to keep existing cpu dai name. And now, it will be used from codec also. But codec driver had been used dais function only even though it was single dai. This patch adds new flag which can selects dai/dais function on component register function to keep existing codec dai name. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-09-04 19:39:03 -07:00
ASoC: Move card field form platform/codec to component Both the snd_soc_codec and snd_soc_platform struct do have a pointer to the parent card and both handle this pointer in mostly the same way. This patch moves the card field to the component level which will allow further code consolidation between platforms and CODECS. Since there are only a handful of users of the snd_soc_codec struct's card field (and none of the snd_soc_platform's) these are update in this patch as well, which allows it to be removed from the snd_soc_codec struct. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-17 22:01:07 +02:00
ASoC: Move active count from CODEC to component There is no reason why active count tracking should only be done for CODECs but not for other components. Moving the active count from the snd_soc_codec struct to the snd_soc_component struct reduces the differences between CODECs and other components and will eventually allow component to component DAI links (Which is a prerequisite for converting CODECs to components). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-03-05 13:17:47 +01:00
ASoC: add Component level suspend/resume In current ALSA SoC, Codec only has suspend/resume feature, but it should be supported on Component level. This patch adds it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-11-30 06:23:13 +00:00
ASoC: Move ignore_pmdown_time from CODEC to component In preparation for componentization move the ignore_pmdown_time field from the snd_soc_codec struct to the snd_soc_component struct. Set it to true for non CODEC components for now. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-03-05 13:17:48 +01:00
ASoC: snd_soc_codec includes snd_soc_component Codec includes component by this patch, and component moved to upside of codec to avoid extra declaration. Codec dai will be registered via component by this patch. Current component register function is used for cpu, and it is using dai/dais functions properly to keep existing cpu dai name. And now, it will be used from codec also. But codec driver had been used dais function only even though it was single dai. This patch adds new flag which can selects dai/dais function on component register function to keep existing codec dai name. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-09-04 19:39:03 -07:00
ASoC: Fix binding and probing of auxiliary components Currently binding of auxiliary devices doesn't work as in soc_bind_aux_dev() function a bound component is not being added to any list and in soc_probe_aux_devices() we are trying to walk the component_dev_list list to probe auxiliary components but at that time this list doesn't contain any auxiliary components since they are being added to the card only in soc_probe_component(). This patch adds a list to the card where are stored bound but not probed auxiliary devices, so that all aux devices can be probed. Fixes: 1a653aa44725 "ASoC: core: replace aux_comp_list to component_dev_list" Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-12-29 14:11:21 +01:00
ASoC: core: replace codec_dev_list to component_dev_list on Card Current Card has Codec list (= codec_dev_list), but Codec will be removed in the future. Because of this reason, this patch adds new Component list in Card, and replace Codec list. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-11-30 06:22:36 +00:00
ASoC: snd_soc_codec includes snd_soc_component Codec includes component by this patch, and component moved to upside of codec to avoid extra declaration. Codec dai will be registered via component by this patch. Current component register function is used for cpu, and it is using dai/dais functions properly to keep existing cpu dai name. And now, it will be used from codec also. But codec driver had been used dais function only even though it was single dai. This patch adds new flag which can selects dai/dais function on component register function to keep existing codec dai name. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-09-04 19:39:03 -07:00
ASoC: Add a per component dai list Now that every DAI has a component we can track the DAIs on a per component basis. This simplifies the DAI lookup when we are only interested in DAIs of a specific component and also makes it possible to have multiple components with the same parent device and also register DAIs. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-03-09 17:41:47 +01:00
ASoC: soc.h: Arrange DAI related parameter "dai_list" and "num_dai" on snd_soc_component are related parameter. Let's arrange these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-12-20 01:48:44 +00:00
ASoC: Move IO abstraction to the component level We currently have two very similar IO abstractions in ASoC, one for CODECs, the other for platforms. Moving this to the component level will allow us to unify those two. It will also enable us to move the standard kcontrol helpers as well as DAPM support to the component level. The new component level abstraction layer is primarily build around regmap. There is a per component pointer for the regmap instance for the underlying device. There are four new function snd_soc_component_read(), snd_soc_component_write(), snd_soc_component_update_bits() and snd_soc_component_update_bits_async(). They have the same signature as their regmap counter-part and will internally forward the call one-to-one to regmap. If the component it not using regmap it will fallback to using the custom IO callbacks. This is done to be able to support drivers that haven't been converted to regmap yet, but it is expected that this will eventually be removed in the future once all component drivers have been converted to regmap. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-22 13:23:13 +02:00
ASoC: Add DAPM support at the component level This patch adds full DAPM support at the component level. Previously there was only full DAPM support for CODECs and partial DAPM support (e.g. no Mixers nor MUXs) for platforms. Having DAPM support at the component level will allow all types of components to use DAPM and also help in consolidating the DAPM support between CODECs and platforms. Since the DAPM context is directly embedded into the snd_soc_codec and snd_soc_platform struct and the 'dapm' field is directly referenced in a lot of drivers moving the field just right now is not possible without causing code churn. The approach this patch takes is to add two new fields to the component struct. One field which is the pointer to the actual DAPM context used by the component and one DAPM context that will be used as the default if no other context was specified. For CODECs and platforms the pointer is initialized to point to the CODEC or platform DAPM context. All generic code when referencing a component's DAPM struct will go via the pointer. This will make it possible to eventually seamlessly move the DAPM context from snd_soc_codec and snd_soc_platform struct over once all direct references have been eliminated. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-16 18:13:06 +02:00
ASoC: topology: Add topology core The topology core parses the FW topology file for known block types and instanciates any common ALSA/ASoC objects that it discovers. The core also passes any block that is does not understand to client component drivers for enumeration. The core exports some APIs to client drivers in order to load and unload firmware topology data as use case require. Currently the core deals with the following object types :- o kcontrols. This includes TLV, enumerated and bytes controls. o DAPM widgets. All types with any associated kcontrol. o DAPM graph. o FE PCM. FE PCM capabilities and configuration can be defined. o BE DAI Link. BE DAI link capabilities and configuration can be defined. o Codec <-> codec style links capabilities and configuration. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-05-29 19:06:14 +01:00
ASoC: Move debugfs registration to the component level The debugfs registration is mostly identical between platforms and CODECs. This patches consolidates the two implementations at the component level. Unfortunately there are still a couple of CODEC specific debugfs files that are related to legacy ASoC IO that need to be registered. For this a new callback is added to the component struct that will be initialized when a CODEC is registered and will be used to register the CODEC specific files. Once there are no drivers left using legacy IO this can be removed again. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-19 15:51:18 +02:00
ASoC: Add DAPM support at the component level This patch adds full DAPM support at the component level. Previously there was only full DAPM support for CODECs and partial DAPM support (e.g. no Mixers nor MUXs) for platforms. Having DAPM support at the component level will allow all types of components to use DAPM and also help in consolidating the DAPM support between CODECs and platforms. Since the DAPM context is directly embedded into the snd_soc_codec and snd_soc_platform struct and the 'dapm' field is directly referenced in a lot of drivers moving the field just right now is not possible without causing code churn. The approach this patch takes is to add two new fields to the component struct. One field which is the pointer to the actual DAPM context used by the component and one DAPM context that will be used as the default if no other context was specified. For CODECs and platforms the pointer is initialized to point to the CODEC or platform DAPM context. All generic code when referencing a component's DAPM struct will go via the pointer. This will make it possible to eventually seamlessly move the DAPM context from snd_soc_codec and snd_soc_platform struct over once all direct references have been eliminated. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-16 18:13:06 +02:00
ASoC: Move debugfs registration to the component level The debugfs registration is mostly identical between platforms and CODECs. This patches consolidates the two implementations at the component level. Unfortunately there are still a couple of CODEC specific debugfs files that are related to legacy ASoC IO that need to be registered. For this a new callback is added to the component struct that will be initialized when a CODEC is registered and will be used to register the CODEC specific files. Once there are no drivers left using legacy IO this can be removed again. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-19 15:51:18 +02:00
ASoC: Make aux_dev more like a generic component aux_dev is mainly used by the machine driver to specify analog devices, which are registered as codecs. Making it more like a generic component can help the machine driver to use it to specify any component with topology info by name. Details: - Remove the stub 'rtd_aux' array from the soc card. - Add a list 'aux_comp_list' to store the components of aux_devs. And add a list head 'list_aux' to struct snd_soc_component, for adding such components to the above list. - Add a 'init' ops to a component for machine specific init. soc_bind_aux_dev() will set it to be aux_dev's init. And it will be called when probing the component. - soc_bind_aux_dev() will also search components by name of an aux_dev, since it may not be a codec. - Move probing of aux_devs before checking new DAI links brought by topology. - Move removal of aux_devs later than removal of links. Because topology of aux components may register DAIs and the DAI drivers will go with removal of the aux components, we want soc_remove_link_dais() to remove the DAIs at first. Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-01-06 13:29:31 +08:00
ASoC: Move debugfs registration to the component level The debugfs registration is mostly identical between platforms and CODECs. This patches consolidates the two implementations at the component level. Unfortunately there are still a couple of CODEC specific debugfs files that are related to legacy ASoC IO that need to be registered. For this a new callback is added to the component struct that will be initialized when a CODEC is registered and will be used to register the CODEC specific files. Once there are no drivers left using legacy IO this can be removed again. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-19 15:51:18 +02:00
ASoC: soc.h: merge CONFIG_DEBUG_FS Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Reviewed-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-05-08 03:21:46 +00:00
ASoC: Move debugfs registration to the component level The debugfs registration is mostly identical between platforms and CODECs. This patches consolidates the two implementations at the component level. Unfortunately there are still a couple of CODEC specific debugfs files that are related to legacy ASoC IO that need to be registered. For this a new callback is added to the component struct that will be initialized when a CODEC is registered and will be used to register the CODEC specific files. Once there are no drivers left using legacy IO this can be removed again. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-19 15:51:18 +02:00
ASoC: snd_soc_codec includes snd_soc_component Codec includes component by this patch, and component moved to upside of codec to avoid extra declaration. Codec dai will be registered via component by this patch. Current component register function is used for cpu, and it is using dai/dais functions properly to keep existing cpu dai name. And now, it will be used from codec also. But codec driver had been used dais function only even though it was single dai. This patch adds new flag which can selects dai/dais function on component register function to keep existing codec dai name. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-09-04 19:39:03 -07:00
ASoC: add for_each_component_dais() macro To be more readable code, this patch adds new for_each_component_dais() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-21 05:23:17 +00:00
ASoC: soc-core: add snd_soc_rtdcom_xxx() Current snd_soc_pcm_runtime has platform / codec pointers, and we could use these specific pointer. But these will be replaced to more generic "component" soon, and will need more generic method to get each connected component pointer from rtd. This patch adds new snd_soc_rtdcom_xxx() to connect/disconnect component to rtd. It means same as previous "platform" / "codec" pointer style, but more generic. We can find necessary component pointer from rtd by using component driver name on snd_soc_rtdcom_lookup(). Here, the reason why it uses "driver name" is that "component name" was created by fmt_single_name() and difficult to use it from driver. Driver of course knows its "driver name", thus, using it is more easy. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Arnaud Pouliquen <arnaud.pouliquen@st.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-08-08 06:17:47 +00:00
ASoC: core: Add initial support for DAI multicodec DAI link assumes a one to one mapping between CPU DAI and CODEC. In some cases, the same CPU DAI can be connected to several codecs. This is the case for example, if you connect two mono codecs to the same I2S link in order to have a stereo card. The current ASoC implementation does not allow such setup. Add support for DAI links composed of a single CPU DAI and multiple CODECs. Sound cards have to pass the CODECs array in the corresponding DAI link through a new 'snd_soc_dai_link_component' struct. Each CODEC in this array is described in the same manner single CODEC DAIs are (either DT/OF node or codec_name). Multi-codec links are not supported in the case of CODEC to CODEC links. Just print a warning if it happens. Based on an original code done by Misael. Signed-off-by: Benoit Cousson <bcousson@baylibre.com> Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Fabien Parent <fparent@baylibre.com> Tested-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-08 23:19:34 +02:00
ASoC: Remove 'const' from the device_node pointers As Russell King's explained it, there should not be pointers to struct device_node: "struct device_node is a ref-counted structure. That means if you store a reference to it, you should "get" it, and you should "put" it once you've done. The act of "put"ing the pointed-to structure involves writing to that structure, so it is totally unappropriate to store a device_node structure as a const pointer. It forces you to have to cast it back to a non-const pointer at various points in time to use various OF function calls." [This isn't quite the application here, we're not geting or putting the pointer though we did add some other users who call non-const OF functions -- broonie] Signed-off-by: Jean-Francois Moine <moinejf@free.fr> Signed-off-by: Mark Brown <broonie@kernel.org>
2014-11-25 12:14:48 +01:00
ASoC: core: Add initial support for DAI multicodec DAI link assumes a one to one mapping between CPU DAI and CODEC. In some cases, the same CPU DAI can be connected to several codecs. This is the case for example, if you connect two mono codecs to the same I2S link in order to have a stereo card. The current ASoC implementation does not allow such setup. Add support for DAI links composed of a single CPU DAI and multiple CODECs. Sound cards have to pass the CODECs array in the corresponding DAI link through a new 'snd_soc_dai_link_component' struct. Each CODEC in this array is described in the same manner single CODEC DAIs are (either DT/OF node or codec_name). Multi-codec links are not supported in the case of CODEC to CODEC links. Just print a warning if it happens. Based on an original code done by Misael. Signed-off-by: Benoit Cousson <bcousson@baylibre.com> Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Fabien Parent <fparent@baylibre.com> Tested-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-08 23:19:34 +02:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: soc.h: add explanation of legacy/modern style of dai_link Current ALSA SoC is assuming 1 CPU 1 Platform (= DMA) style system. Because of this background, it is directly using xxx_name / xxx_of_node / xxx_dai_name on dai_link. Let's call it as legacy style here. More complex style system like multi CPU multi Platform (= DMA) will coming. To supporting it, we can use snd_soc_dai_link_component on dai_link. Let's call it as modern style here. But current ALSA SoC can't support it so far. Thus, we need to have multi CPU / multi Codec / multi Platform style in the future on ALSA SoC. Currently we already have multi Codec support. Platform is starting to use modern style on dai_link, but still style only. Multi Platform is not yet implemented. And we still don't have multi CPU support on ALSA SoC, and not have modern style either. Currently, if driver is using legacy style Codec/Platform, it will be converted to modern style on soc-core. This means, we are using glue code for legacy vs modern style so far on ALSA SoC. We can fully switch to modern style on all drivers if ALSA SoC supported modern style for CPU, and then, legacy style code will be removed from ALSA SoC. Untile then, we need to keep both legacy/modern style and its glue code. This patch adds such future plan and background on soc.h Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-01-18 11:20:41 +09:00
ASoC: make snd_soc_dai_link more symmetrical Prior to this patch, the CPU side of a DAI link was specified using a single name. Often, this was the result of calling dev_name() on the device providing the DAI, but in the case of a CPU DAI driver that provided multiple DAIs, it needed to mix together both the device name and some device-relative name, in order to form a single globally unique name. However, the CODEC side of the DAI link was specified using separate fields for device (name or OF node) and device-relative DAI name. This patch allows the CPU side of a DAI link to be specified in the same way as the CODEC side, separating concepts of device and device-relative DAI name. I believe this will be important in multi-codec and/or dynamic PCM scenarios, where a single CPU driver provides multiple DAIs, while also booting using device tree, with accompanying desire not to hard-code the CPU side device's name into the original .cpu_dai_name field. Ideally, both the CPU DAI and CODEC DAI loops in soc_bind_dai_link() would now be identical. However, two things prevent that at present: 1) The need to save rtd->codec for the CODEC side, which means we have to search for the CODEC explicitly, and not just the CODEC side DAI. 2) Since we know the CODEC side DAI is part of a codec, and not just a standalone DAI, it's slightly more efficient to convert .codec_name/ .codec_of_node into a codec first, and then compare each DAI's .codec field, since this avoids strcmp() on each DAI's CODEC's name within the loop. However, the two loops are essentially semantically equivalent. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-25 18:22:11 -06:00
ASoC: soc-core: use snd_soc_dai_link_component for CPU current ALSA SoC is starting to support modern style dai_linke (= struct snd_soc_dai_link_component) which is mainly used for multipul DAI/component connection. Now Codec has full multi-codec support, Platform is using modern style but still for single Platform. Only CPU is not yet supporting modern style yet. If we could support it for CPU, we can switch to modern style dai_link on all CPU/Codec/Platform, and remove legacy style from ALSA SoC. Multi-CPU will be supported in the future. This patch is initial support for modern style for CPU Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-06-06 13:07:22 +09:00
ASoC: make snd_soc_dai_link more symmetrical Prior to this patch, the CPU side of a DAI link was specified using a single name. Often, this was the result of calling dev_name() on the device providing the DAI, but in the case of a CPU DAI driver that provided multiple DAIs, it needed to mix together both the device name and some device-relative name, in order to form a single globally unique name. However, the CODEC side of the DAI link was specified using separate fields for device (name or OF node) and device-relative DAI name. This patch allows the CPU side of a DAI link to be specified in the same way as the CODEC side, separating concepts of device and device-relative DAI name. I believe this will be important in multi-codec and/or dynamic PCM scenarios, where a single CPU driver provides multiple DAIs, while also booting using device tree, with accompanying desire not to hard-code the CPU side device's name into the original .cpu_dai_name field. Ideally, both the CPU DAI and CODEC DAI loops in soc_bind_dai_link() would now be identical. However, two things prevent that at present: 1) The need to save rtd->codec for the CODEC side, which means we have to search for the CODEC explicitly, and not just the CODEC side DAI. 2) Since we know the CODEC side DAI is part of a codec, and not just a standalone DAI, it's slightly more efficient to convert .codec_name/ .codec_of_node into a codec first, and then compare each DAI's .codec field, since this avoids strcmp() on each DAI's CODEC's name within the loop. However, the two loops are essentially semantically equivalent. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-25 18:22:11 -06:00
ASoC: core: Add initial support for DAI multicodec DAI link assumes a one to one mapping between CPU DAI and CODEC. In some cases, the same CPU DAI can be connected to several codecs. This is the case for example, if you connect two mono codecs to the same I2S link in order to have a stereo card. The current ASoC implementation does not allow such setup. Add support for DAI links composed of a single CPU DAI and multiple CODECs. Sound cards have to pass the CODECs array in the corresponding DAI link through a new 'snd_soc_dai_link_component' struct. Each CODEC in this array is described in the same manner single CODEC DAIs are (either DT/OF node or codec_name). Multi-codec links are not supported in the case of CODEC to CODEC links. Just print a warning if it happens. Based on an original code done by Misael. Signed-off-by: Benoit Cousson <bcousson@baylibre.com> Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Fabien Parent <fparent@baylibre.com> Tested-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-08 23:19:34 +02:00
ASoC: make snd_soc_dai_link more symmetrical Prior to this patch, the CPU side of a DAI link was specified using a single name. Often, this was the result of calling dev_name() on the device providing the DAI, but in the case of a CPU DAI driver that provided multiple DAIs, it needed to mix together both the device name and some device-relative name, in order to form a single globally unique name. However, the CODEC side of the DAI link was specified using separate fields for device (name or OF node) and device-relative DAI name. This patch allows the CPU side of a DAI link to be specified in the same way as the CODEC side, separating concepts of device and device-relative DAI name. I believe this will be important in multi-codec and/or dynamic PCM scenarios, where a single CPU driver provides multiple DAIs, while also booting using device tree, with accompanying desire not to hard-code the CPU side device's name into the original .cpu_dai_name field. Ideally, both the CPU DAI and CODEC DAI loops in soc_bind_dai_link() would now be identical. However, two things prevent that at present: 1) The need to save rtd->codec for the CODEC side, which means we have to search for the CODEC explicitly, and not just the CODEC side DAI. 2) Since we know the CODEC side DAI is part of a codec, and not just a standalone DAI, it's slightly more efficient to convert .codec_name/ .codec_of_node into a codec first, and then compare each DAI's .codec field, since this avoids strcmp() on each DAI's CODEC's name within the loop. However, the two loops are essentially semantically equivalent. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-25 18:22:11 -06:00
ASoC: soc-core: allow no Platform on dai_link dai_link is used to selecting Component (= CPU/Codec/Platform) and DAI (= CPU/Codec). And selected CPU/Codec/Platform components are *listed* on Card. Many drivers don't need special Platform component, but was mandatory at legacy style ALSA SoC. Thus, there is this kind of settings on many drivers. dai_link->platform_of_node = dai_link->cpu_of_node; In this case, soc_bind_dai_link() will pick-up "CPU component" as "Platform component", and try to add it to snd_soc_pcm_runtime. But it will be ignored, because it is already added when CPU bindings. Historically, this kind of "CPU component" is used/selected as "Platform" on many ALSA SoC drivers. OTOH, Dummy Platform will be selected automatically by ALSA SoC if driver doesn't have Platform settings. These indicates that there are 2 type of Platforms exist at current ALSA SoC if driver doesn't need special Platform. 1) use Dummy Platform as Platform component 2) use CPU component as Platform component ALSA SoC will call Dummy Platform callback function if it is using Dummy Platform, but it is completely pointless. Because it is the sound card which doesn't need special Platform. Thus, the behavior we request to ALSA SoC is selecting 2) automatically instead of 1) if sound card doesn't need special Platform. And, 2) means "do nothing" as above explain. These were needed at legacy style dai_link, but is no longer needed at modern style dai_link anymore. This patch allows "no Platform" settings on dai_link, and will do nothing for it if there was no platform settings. This is same as 2). By this patch, all drivers which is selecting "CPU component" as "Platform" can remove such settings. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-06-19 10:14:07 +09:00
ASoC: make snd_soc_dai_link more symmetrical Prior to this patch, the CPU side of a DAI link was specified using a single name. Often, this was the result of calling dev_name() on the device providing the DAI, but in the case of a CPU DAI driver that provided multiple DAIs, it needed to mix together both the device name and some device-relative name, in order to form a single globally unique name. However, the CODEC side of the DAI link was specified using separate fields for device (name or OF node) and device-relative DAI name. This patch allows the CPU side of a DAI link to be specified in the same way as the CODEC side, separating concepts of device and device-relative DAI name. I believe this will be important in multi-codec and/or dynamic PCM scenarios, where a single CPU driver provides multiple DAIs, while also booting using device tree, with accompanying desire not to hard-code the CPU side device's name into the original .cpu_dai_name field. Ideally, both the CPU DAI and CODEC DAI loops in soc_bind_dai_link() would now be identical. However, two things prevent that at present: 1) The need to save rtd->codec for the CODEC side, which means we have to search for the CODEC explicitly, and not just the CODEC side DAI. 2) Since we know the CODEC side DAI is part of a codec, and not just a standalone DAI, it's slightly more efficient to convert .codec_name/ .codec_of_node into a codec first, and then compare each DAI's .codec field, since this avoids strcmp() on each DAI's CODEC's name within the loop. However, the two loops are essentially semantically equivalent. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-05-25 18:22:11 -06:00
ASoC: soc-core: add .num_platform for dai_link Current snd_soc_dai_link is starting to use snd_soc_dai_link_component (= modern) style for Platform, but it is still assuming single Platform so far. We will need to have multi Platform support in the not far future. Currently only simple card is using it as sound card driver, and other drivers are converted to it from legacy style by snd_soc_init_platform(). To avoid future problem of multi Platform support, let's add num_platforms before it is too late. In the same time, to make it same naming mothed, "platform" should be "platforms". This patch fixup it too. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-01-21 09:32:32 +09:00
ASoC: soc-core: use snd_soc_dai_link_component for platform Current struct snd_soc_dai_link is supporting multicodec, and it is supporting legacy style of codec_name codec_of_node code_dai_name This is handled as single entry of multicodec. We don't have multicpu support yet, but in the future we will. In such case, we can use snd_soc_dai_link_component for both cpu/codec. Then the code will be more simple and readble. As next step, we want to use it for platform, too. This patch adds snd_soc_dai_link_component style for platform. We might have multiplatform support in the future, but we don't know yet. To avoid un-known issue / complex code, this patch supports just single-platform as 1st step. If we could use snd_soc_dai_link_component for all CPU/Codec/Platform, we will switch to new style, and remove legacy code. This is prepare for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-08-31 03:10:08 +00:00
ASoC: Change DAI link's be_id to a generic id The generic ID can be used by topology: - Toplogy can create FE links and set their ID, machine drivers will be notified and check this ID for machine-specific init. - Toplogy can use the ID to find existing BE & CC links and further configure them. Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-04-19 13:12:35 +08:00
[ALSA] soc - Ensure PCMs are suspended This fixes a bug whereby PCMs were not being suspended when the rest of the audio subsystem was suspended. Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-10 14:39:01 +01:00
ASoC: core: Support transparent CODEC<->CODEC DAI links 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>
2012-04-04 22:12:09 +01:00
ASoC: dapm: add code to configure dai link parameters dai-link params for codec-codec links were fixed. The fixed link between codec and another chip which may be another codec, baseband, bluetooth codec etc may require run time configuaration changes. This change provides an optional alsa control to select one of the params from a list of params. Signed-off-by: Nikesh Oswal <nikesh@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-02-02 17:06:44 +00:00
ASoC: core: Support transparent CODEC<->CODEC DAI links 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>
2012-04-04 22:12:09 +01:00
ASoC: Allow DAI formats to be specified in the dai_link For almost all machines the DAI format is a constant, always set to the same thing. This means that not only should we normally set it on init rather than in hw_params() (where it has been for historical reasons) we should also allow users to configure this by setting a variable in the dai_link structure. The combination of these two will make many machine drivers even more data driven. Implement a new dai_fmt field in the dai_link doing just that. Since 0 is a valid value for many format flags and we need to be able to tell if the field is actually set also add one to all the values used to configure formats. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-27 16:41:01 +01:00
ASoC: dpcm: Add Dynamic PCM core operations. 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>
2012-04-25 12:12:49 +01:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: Provide core support for symmetric sample rates Many devices require symmetric configurations of capture and playback data formats, often due to shared clocking but sometimes also due to other shared playback and record configuration in the device. Start providing core support for this by allowing the DAIs or the machine to specify that the sample rates used should be kept symmetric. A flag symmetric_rates is provided in the snd_soc_dai and snd_soc_dai_link structures. If this is set in either of the DAIs or in the machine then a constraint will be applied when a stream is already open preventing any changes in sample rate. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-04-07 18:10:13 +01:00
ASoC: dpcm: Add Dynamic PCM core operations. 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>
2012-04-25 12:12:49 +01:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: Constify ops and compr_ops fields of snd_soc_dai_link The core does not modify these fields, so they can be made const. This allows drivers to declare their op tables as const. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-10 17:06:15 +01:00
ASoC: soc-pcm: Allow to specify unidirectional dai_link Add 'playback_only' and 'capture_only' fields that can be used for specifying that a dai_link has a unidirectional capability. The motivation for this is for the cases of systems, such as Freescale MX28, that has two unidirectional DAIs. Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-29 10:32:13 -03:00
ASoC: soc.h: tidyup struct snd_soc_dai_link definition order Current struct snd_soc_dai_link has many members, but definition order was random. Especially, bool / bit field are defined randomly. This patch tidyups these definition order to calculate data alignment easy. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-04-21 07:02:34 +00:00
ASoC: soc.h: use bit field for playback/capture_only Current snd_soc_dai_link is already using many bit fields. Let's use it for playback_only/capture_only too. We can reduce struct size in certain environment. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-10-24 00:44:10 +00:00
ASoC: soc.h: tidyup struct snd_soc_dai_link definition order Current struct snd_soc_dai_link has many members, but definition order was random. Especially, bool / bit field are defined randomly. This patch tidyups these definition order to calculate data alignment easy. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-04-21 07:02:34 +00:00
ASoC: soc-pcm: DPCM cares BE format Current DPCM is caring only FE format. but it will be no sound if FE/BE was below style, and user selects S24_LE format. FE: S16_LE/S24_LE BE: S16_LE DPCM can rewrite the format, so basically we don't want to constrain with the BE constraints. But sometimes it will be trouble. This patch adds new .dpcm_merged_format on struct snd_soc_dai_link. DPCM will use FE / BE merged format if .struct snd_soc_dai_link has it. We can have other .dpcm_merged_xxx in the future .dpcm_merged_foramt .dpcm_merged_rate .dpcm_merged_chan Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-05-12 02:03:33 +00:00
ASoC: soc-pcm: DPCM cares BE channel constraint Current DPCM is caring only FE channel configuration. Sometimes it will be trouble if user selects channel which isn't supported by BE. This patch adds new .dpcm_merged_chan on struct snd_soc_dai_link. DPCM will use FE / BE merged channel if struct snd_soc_dai_link has it. Signed-off-by: Jiada Wang <jiada_wang@mentor.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-06-20 18:25:20 +09:00
ASoC: dpcm: add rate merge to the BE stream merge As done for format and channels, add the possibility to merge the backend rates on the frontend rates. This useful if the backend does not support all rates supported by the frontend, or if several backends (cpu and codecs) with different capabilities are connected to the same frontend. Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-05 12:13:49 +02:00
ASoC: soc-pcm: DPCM cares BE format Current DPCM is caring only FE format. but it will be no sound if FE/BE was below style, and user selects S24_LE format. FE: S16_LE/S24_LE BE: S16_LE DPCM can rewrite the format, so basically we don't want to constrain with the BE constraints. But sometimes it will be trouble. This patch adds new .dpcm_merged_format on struct snd_soc_dai_link. DPCM will use FE / BE merged format if .struct snd_soc_dai_link has it. We can have other .dpcm_merged_xxx in the future .dpcm_merged_foramt .dpcm_merged_rate .dpcm_merged_chan Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-05-12 02:03:33 +00:00
ASoC: soc.h: tidyup struct snd_soc_dai_link definition order Current struct snd_soc_dai_link has many members, but definition order was random. Especially, bool / bit field are defined randomly. This patch tidyups these definition order to calculate data alignment easy. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-04-21 07:02:34 +00:00
ASoC: Implement DAI links in a list & define API to add/remove a link Implement a dai link list for the soc card. Add APIs to add/remove a DAI links dynamically, e.g. by topology. And a dobj is embedded into the struct snd_soc_dai_link. Topology can use the dobj to find the links created by it and remove them when the topology component is unloaded. The predefined DAI links are reserved to keep backward compatibility. And they will also be added to the list. Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-12-02 14:11:22 +08:00
ASoC: core: Allow topology to override machine driver FE DAI link config. Machine drivers statically define a number of DAI links that currently cannot be changed or removed by topology. This means PCMs and platform components cannot be changed by topology at runtime AND machine drivers are tightly coupled to topology. This patch allows topology to override the machine driver DAI link config in order to reuse machine drivers with different topologies and platform components. The patch supports :- 1) create new FE PCMs with a topology defined PCM ID. 2) destroy existing static FE PCMs 3) change the platform component driver. 4) assign any new HW params fixups. 5) assign a new card name prefix to differentiate this topology to userspace. The patch requires no changes to the machine drivers, but does add some platform component flags that the platform component driver can assign before loading topologies. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-02 16:59:54 +01:00
ASoC: Implement DAI links in a list & define API to add/remove a link Implement a dai link list for the soc card. Add APIs to add/remove a DAI links dynamically, e.g. by topology. And a dobj is embedded into the struct snd_soc_dai_link. Topology can use the dobj to find the links created by it and remove them when the topology component is unloaded. The predefined DAI links are reserved to keep backward compatibility. And they will also be added to the list. Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-12-02 14:11:22 +08:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: add for_each_link_codecs() macro ALSA SoC snd_soc_dai_link has snd_soc_dai_link_component array for codecs. To be more readable code, this patch adds new for_each_link_codecs() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-03 02:12:40 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: soc-core: support dai_link with platforms_num != 1 Add support platforms_num != 1 in dai_link. Initially, the main purpose of this change was to make the platform optional in the dai_link, instead of inserting the dummy platform driver. This particular case had just been solved by Kuninori Morimoto with commit 1d7689892878 ("ASoC: soc-core: allow no Platform on dai_link"). However, this change may still be useful for those who need multiple platform components on a single dai_link (it solves one of the FIXME note in soc-core) Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-06-27 14:13:50 +02:00
ASoC: soc.h: add sound dai_link connection macro Modern style dai_link requests CPU/Codec/Platform component pointer array and its size, but it will be very verbose code. To avoid such scene, this patch adds dai_link connection macro. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-06-06 13:07:42 +09:00
ASoC: soc-core: Generalize snd_soc_prefix_map and rename to snd_soc_codec_conf The snd_soc_codec_conf struct now holds codec specific configuration information. A new configuration option has been added to allow machine drivers to override the compression type set by the codec driver. In the absence of providing an snd_soc_codec_conf struct or when providing one but not setting the compress_type member to anything, the one supplied by the codec driver will be used instead. In all other cases the one set in the snd_soc_codec_conf struct takes effect. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-12-02 14:53:03 +00:00
ASoC: dt: Allow Aux Codecs to be specified using DT This patch adds support for specifying auxiliary codecs and codec configuration via device tree phandles. This change adds new fields to snd_soc_aux_dev and snd_soc_codec_conf and adds support for the changes to SoC core methods. Signed-off-by: Pavel Machek <pavel@ucw.cz> Signed-off-by: Sebastian Reichel <sre@kernel.org> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-28 16:07:22 +02:00
ASoC: Add optional name_prefix for codec kcontrol, widget and route names There is a need to prefix codec kcontrol, widget and internal route names in an ASoC machine that has multiple codecs with conflicting names. The name collision would occur when codec drivers try to registering kcontrols with the same name or when building audio paths. This patch introduces optional prefix_map into struct snd_soc_card. With it machine drivers can specify a unique name prefix to each codec that have conflicting names with anothers. Prefix to codec is matched with codec name. Following example illustrates a machine that has two same codec instances. Name collision from kcontrol registration is avoided by specifying a name prefix "foo" for the second codec. As the codec widget names are prefixed then second audio map for that codec shows a prefixed widget name. static const struct snd_soc_dapm_route map0[] = { {"Spk", NULL, "MONO"}, }; static const struct snd_soc_dapm_route map1[] = { {"Vibra", NULL, "foo MONO"}, }; static struct snd_soc_prefix_map codec_prefix[] = { { .dev_name = "codec.2", .name_prefix = "foo", }, }; static struct snd_soc_card card = { ... .prefix_map = codec_prefix, .num_prefixes = ARRAY_SIZE(codec_prefix), }; 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>
2010-11-13 20:40:44 +02:00
ASoC: Remove 'const' from the device_node pointers As Russell King's explained it, there should not be pointers to struct device_node: "struct device_node is a ref-counted structure. That means if you store a reference to it, you should "get" it, and you should "put" it once you've done. The act of "put"ing the pointed-to structure involves writing to that structure, so it is totally unappropriate to store a device_node structure as a const pointer. It forces you to have to cast it back to a non-const pointer at various points in time to use various OF function calls." [This isn't quite the application here, we're not geting or putting the pointer though we did add some other users who call non-const OF functions -- broonie] Signed-off-by: Jean-Francois Moine <moinejf@free.fr> Signed-off-by: Mark Brown <broonie@kernel.org>
2014-11-25 12:14:48 +01:00
ASoC: soc-core: Generalize snd_soc_prefix_map and rename to snd_soc_codec_conf The snd_soc_codec_conf struct now holds codec specific configuration information. A new configuration option has been added to allow machine drivers to override the compression type set by the codec driver. In the absence of providing an snd_soc_codec_conf struct or when providing one but not setting the compress_type member to anything, the one supplied by the codec driver will be used instead. In all other cases the one set in the snd_soc_codec_conf struct takes effect. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-12-02 14:53:03 +00:00
ASoC: Add optional name_prefix for codec kcontrol, widget and route names There is a need to prefix codec kcontrol, widget and internal route names in an ASoC machine that has multiple codecs with conflicting names. The name collision would occur when codec drivers try to registering kcontrols with the same name or when building audio paths. This patch introduces optional prefix_map into struct snd_soc_card. With it machine drivers can specify a unique name prefix to each codec that have conflicting names with anothers. Prefix to codec is matched with codec name. Following example illustrates a machine that has two same codec instances. Name collision from kcontrol registration is avoided by specifying a name prefix "foo" for the second codec. As the codec widget names are prefixed then second audio map for that codec shows a prefixed widget name. static const struct snd_soc_dapm_route map0[] = { {"Spk", NULL, "MONO"}, }; static const struct snd_soc_dapm_route map1[] = { {"Vibra", NULL, "foo MONO"}, }; static struct snd_soc_prefix_map codec_prefix[] = { { .dev_name = "codec.2", .name_prefix = "foo", }, }; static struct snd_soc_card card = { ... .prefix_map = codec_prefix, .num_prefixes = ARRAY_SIZE(codec_prefix), }; 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>
2010-11-13 20:40:44 +02:00
ASoC: Add support for optional auxiliary dailess codecs This makes possible to register auxiliary dailess codecs in a machine driver. Term dailess is used here for amplifiers and codecs without DAI or DAI being unused. Dailess auxiliary codecs are kept in struct snd_soc_aux_dev and those codecs are probed after initializing the DAI links. There are no major differences between DAI link codecs and dailess codecs in ASoC core point of view. DAPM handles them equally and sysfs and debugfs directories for dailess codecs are similar except the pmdown_time node is not created. Only suspend and resume functions are modified to traverse all probed codecs instead of DAI link codecs. Example below shows a dailess codec registration. struct snd_soc_aux_dev foo_aux_dev[] = { { .name = "Amp", .codec_name = "codec.2", .init = foo_init2, }, }; static struct snd_soc_card card = { ... .aux_dev = foo_aux_dev, .num_aux_devs = ARRAY_SIZE(foo_aux_dev), }; Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-25 17:47:38 +02:00
ASoC: dt: Allow Aux Codecs to be specified using DT This patch adds support for specifying auxiliary codecs and codec configuration via device tree phandles. This change adds new fields to snd_soc_aux_dev and snd_soc_codec_conf and adds support for the changes to SoC core methods. Signed-off-by: Pavel Machek <pavel@ucw.cz> Signed-off-by: Sebastian Reichel <sre@kernel.org> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-28 16:07:22 +02:00
ASoC: Remove 'const' from the device_node pointers As Russell King's explained it, there should not be pointers to struct device_node: "struct device_node is a ref-counted structure. That means if you store a reference to it, you should "get" it, and you should "put" it once you've done. The act of "put"ing the pointed-to structure involves writing to that structure, so it is totally unappropriate to store a device_node structure as a const pointer. It forces you to have to cast it back to a non-const pointer at various points in time to use various OF function calls." [This isn't quite the application here, we're not geting or putting the pointer though we did add some other users who call non-const OF functions -- broonie] Signed-off-by: Jean-Francois Moine <moinejf@free.fr> Signed-off-by: Mark Brown <broonie@kernel.org>
2014-11-25 12:14:48 +01:00
ASoC: Add support for optional auxiliary dailess codecs This makes possible to register auxiliary dailess codecs in a machine driver. Term dailess is used here for amplifiers and codecs without DAI or DAI being unused. Dailess auxiliary codecs are kept in struct snd_soc_aux_dev and those codecs are probed after initializing the DAI links. There are no major differences between DAI link codecs and dailess codecs in ASoC core point of view. DAPM handles them equally and sysfs and debugfs directories for dailess codecs are similar except the pmdown_time node is not created. Only suspend and resume functions are modified to traverse all probed codecs instead of DAI link codecs. Example below shows a dailess codec registration. struct snd_soc_aux_dev foo_aux_dev[] = { { .name = "Amp", .codec_name = "codec.2", .init = foo_init2, }, }; static struct snd_soc_card card = { ... .aux_dev = foo_aux_dev, .num_aux_devs = ARRAY_SIZE(foo_aux_dev), }; Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-25 17:47:38 +02:00
ASoC: Pass component instead of DAPM context to AUX dev init callback Given that the component is the containing structure it makes more sense to pass the component rather than the DAPM context to the AUX dev init callback. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-19 15:51:23 +02:00
ASoC: Add support for optional auxiliary dailess codecs This makes possible to register auxiliary dailess codecs in a machine driver. Term dailess is used here for amplifiers and codecs without DAI or DAI being unused. Dailess auxiliary codecs are kept in struct snd_soc_aux_dev and those codecs are probed after initializing the DAI links. There are no major differences between DAI link codecs and dailess codecs in ASoC core point of view. DAPM handles them equally and sysfs and debugfs directories for dailess codecs are similar except the pmdown_time node is not created. Only suspend and resume functions are modified to traverse all probed codecs instead of DAI link codecs. Example below shows a dailess codec registration. struct snd_soc_aux_dev foo_aux_dev[] = { { .name = "Amp", .codec_name = "codec.2", .init = foo_init2, }, }; static struct snd_soc_card card = { ... .aux_dev = foo_aux_dev, .num_aux_devs = ARRAY_SIZE(foo_aux_dev), }; Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-25 17:47:38 +02:00
ASoC: Rename snd_soc_card to snd_soc_machine One of the issues with the ASoC v1 API which has been addressed in the ASoC v2 work that Liam Girdwood has done is that the ALSA card provided by ASoC is distributed around the ASoC structures. For example, machine wide data such as the struct snd_card are maintained as part of the CODEC data structure, preventing the use of multiple codecs. This has been addressed by refactoring the data structures so that all the data for the ALSA card is contained in a single structure snd_soc_card which replaces the existing snd_soc_machine and snd_soc_device. Begin the process of backporting this by renaming struct snd_soc_machine to struct snd_soc_card, better reflecting its function and bringing it closer to standard ALSA terminology. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-18 20:50:34 +00:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: core - allow ASoC more flexible machine name Allow ASoC machine drivers to register a driver name and a longname. This allows user space to determine the flavour of machine driver. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-12 16:14:04 +01:00
ASoC: core: Add API to use DMI name in sound card long name Intel DSP platform drivers are used by many different devices but are difficult for userspace to differentiate. This patch adds an API to allow the DMI name to be used in the sound card long name, thereby helping userspace load the correct UCM configuration. Usually machine drivers uses their own name as the sound card name (short name), and leave the long name and driver name blank. This API will use the DMI info like vendor, product and board to make up the card long name. If the machine driver has already explicitly set the long name, this API will do nothing. This patch also allows for further differentiation as many devices that share the same DMI name i.e. Minnowboards, UP boards may be configured with different codecs or firmwares. The API supports flavoring the DMI name into the card longname to provide the extra differentiation required for these devices. For Use Case Manager (UCM) in the user space, changing card long name by this API is backward compatible, since the card name does not change. For a given sound card, even if there is no device-specific UCM configuration file that uses the card long name, UCM will fall back to load the default configuration file that uses the card name. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-01-14 16:13:02 +08:00
ASoC: core: Allow topology to override machine driver FE DAI link config. Machine drivers statically define a number of DAI links that currently cannot be changed or removed by topology. This means PCMs and platform components cannot be changed by topology at runtime AND machine drivers are tightly coupled to topology. This patch allows topology to override the machine driver DAI link config in order to reuse machine drivers with different topologies and platform components. The patch supports :- 1) create new FE PCMs with a topology defined PCM ID. 2) destroy existing static FE PCMs 3) change the platform component driver. 4) assign any new HW params fixups. 5) assign a new card name prefix to differentiate this topology to userspace. The patch requires no changes to the machine drivers, but does add some platform component flags that the platform component driver can assign before loading topologies. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-02 16:59:54 +01:00
ASoC: core: Add API to use DMI name in sound card long name Intel DSP platform drivers are used by many different devices but are difficult for userspace to differentiate. This patch adds an API to allow the DMI name to be used in the sound card long name, thereby helping userspace load the correct UCM configuration. Usually machine drivers uses their own name as the sound card name (short name), and leave the long name and driver name blank. This API will use the DMI info like vendor, product and board to make up the card long name. If the machine driver has already explicitly set the long name, this API will do nothing. This patch also allows for further differentiation as many devices that share the same DMI name i.e. Minnowboards, UP boards may be configured with different codecs or firmwares. The API supports flavoring the DMI name into the card longname to provide the extra differentiation required for these devices. For Use Case Manager (UCM) in the user space, changing card long name by this API is backward compatible, since the card name does not change. For a given sound card, even if there is no device-specific UCM configuration file that uses the card long name, UCM will fall back to load the default configuration file that uses the card name. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-01-14 16:13:02 +08:00
ASoC: Add card registration API ASoC v2 allows cards, codecs and platforms to instantiate separately, with the overall ASoC device only being instantiated once all the required components have registered. As part of backporting Liam's work introduce an initial version of the card registration functions. At present these do nothing active and are internal only, they will be exposed to machine drivers after further backporting. Adding this now allows the datastructures used for dynamic card instantiation to be built up gradually. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-28 13:29:45 +00:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: Add card registration API ASoC v2 allows cards, codecs and platforms to instantiate separately, with the overall ASoC device only being instantiated once all the required components have registered. As part of backporting Liam's work introduce an initial version of the card registration functions. At present these do nothing active and are internal only, they will be exposed to machine drivers after further backporting. Adding this now allows the datastructures used for dynamic card instantiation to be built up gradually. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-28 13:29:45 +00:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: dapm: Use DAPM mutex for DAPM ops instead of codec mutex It has now become necessary to use a DAPM mutex instead of the codec mutex to lock the DAPM operations. This is due to the recent multi component support and forth coming Dynamic PCM updates. Currently we lock DAPM operations with the codec mutex of the calling RTD context. However, DAPM operations can span the whole card context and all components. This patch updates the DAPM operations that use the codec mutex to now use the DAPM mutex PCM subclass for all DAPM ops. We also add a mutex subclass for DAPM init and PCM operations. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-03-07 10:38:26 +00:00
ASoC: Add card registration API ASoC v2 allows cards, codecs and platforms to instantiate separately, with the overall ASoC device only being instantiated once all the required components have registered. As part of backporting Liam's work introduce an initial version of the card registration functions. At present these do nothing active and are internal only, they will be exposed to machine drivers after further backporting. Adding this now allows the datastructures used for dynamic card instantiation to be built up gradually. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-28 13:29:45 +00:00
ASoC: dpcm: prevent snd_soc_dpcm use after free The dpcm get from fe_clients/be_clients may be free before use Add a spin lock at snd_soc_card level, to protect the dpcm instance. The lock may be used in atomic context, so use spin lock. Use irq spin lock version, since the lock may be used in interrupts. possible race condition between void dpcm_be_disconnect( ... list_del(&dpcm->list_be); list_del(&dpcm->list_fe); kfree(dpcm); ... and for_each_dpcm_fe() for_each_dpcm_be*() race condition example Thread 1: snd_soc_dapm_mixer_update_power() -> soc_dpcm_runtime_update() -> dpcm_be_disconnect() -> kfree(dpcm); Thread 2: dpcm_fe_dai_trigger() -> dpcm_be_dai_trigger() -> snd_soc_dpcm_can_be_free_stop() -> if (dpcm->fe == fe) Excpetion Scenario: two FE link to same BE FE1 -> BE FE2 -> Thread 1: switch of mixer between FE2 -> BE Thread 2: pcm_stop FE1 Exception: Unable to handle kernel paging request at virtual address dead0000000000e0 pc=<> [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c sound/soc/soc-pcm.c:3226 if (dpcm->fe == fe) lr=<> [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c Backtrace: [<ffffff89602dba80>] notify_die+0x68/0xb8 [<ffffff896028c7dc>] die+0x118/0x2a8 [<ffffff89602a2f84>] __do_kernel_fault+0x13c/0x14c [<ffffff89602a27f4>] do_translation_fault+0x64/0xa0 [<ffffff8960280cf8>] do_mem_abort+0x4c/0xd0 [<ffffff8960282ad0>] el1_da+0x24/0x40 [<ffffff8960e2cd10>] dpcm_be_dai_trigger+0x29c/0x47c [<ffffff8960e2f694>] dpcm_fe_dai_do_trigger+0x94/0x26c [<ffffff8960e2edec>] dpcm_fe_dai_trigger+0x3c/0x44 [<ffffff8960de5588>] snd_pcm_do_stop+0x50/0x5c [<ffffff8960dded24>] snd_pcm_action+0xb4/0x13c [<ffffff8960ddfdb4>] snd_pcm_drop+0xa0/0x128 [<ffffff8960de69bc>] snd_pcm_common_ioctl+0x9d8/0x30f0 [<ffffff8960de1cac>] snd_pcm_ioctl_compat+0x29c/0x2f14 [<ffffff89604c9d60>] compat_SyS_ioctl+0x128/0x244 [<ffffff8960283740>] el0_svc_naked+0x34/0x38 [<ffffffffffffffff>] 0xffffffffffffffff Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-03-08 13:05:53 +08:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: core: Allow topology to override machine driver FE DAI link config. Machine drivers statically define a number of DAI links that currently cannot be changed or removed by topology. This means PCMs and platform components cannot be changed by topology at runtime AND machine drivers are tightly coupled to topology. This patch allows topology to override the machine driver DAI link config in order to reuse machine drivers with different topologies and platform components. The patch supports :- 1) create new FE PCMs with a topology defined PCM ID. 2) destroy existing static FE PCMs 3) change the platform component driver. 4) assign any new HW params fixups. 5) assign a new card name prefix to differentiate this topology to userspace. The patch requires no changes to the machine drivers, but does add some platform component flags that the platform component driver can assign before loading topologies. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-02 16:59:54 +01:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: Replace pdev with card in machine driver probe and remove 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>
2011-01-26 14:17:20 +00:00
ASoC: Add a late_probe() callback to cards This is run after the DAPM widgets and routes are added, allowing setup of things like jacks using the routes. The main card probe() is run before anything else so can't be used for this purpose. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-03-02 18:36:34 +00:00
ASoC: Replace pdev with card in machine driver probe and remove 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>
2011-01-26 14:17:20 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: Use card rather than soc-audio device to card PM functions 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>
2011-01-26 14:05:25 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
[ALSA] soc - Add device level DAPM event Added a device level dapm event so that both the machine and codec are informed when dapm events occur. Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-10 14:36:20 +01:00
ASoC: Rename snd_soc_card to snd_soc_machine One of the issues with the ASoC v1 API which has been addressed in the ASoC v2 work that Liam Girdwood has done is that the ALSA card provided by ASoC is distributed around the ASoC structures. For example, machine wide data such as the struct snd_card are maintained as part of the CODEC data structure, preventing the use of multiple codecs. This has been addressed by refactoring the data structures so that all the data for the ALSA card is contained in a single structure snd_soc_card which replaces the existing snd_soc_machine and snd_soc_device. Begin the process of backporting this by renaming struct snd_soc_machine to struct snd_soc_card, better reflecting its function and bringing it closer to standard ALSA terminology. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-18 20:50:34 +00:00
ASoC: Add context parameter to card DAPM callbacks The card callback will get called for each DAPM context in the card so it can be useful for it to know which device is currently undergoing a transition. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 19:13:23 +01:00
[ALSA] ASoC: Clarify API for bias configuration Currently the ASoC core configures the bias levels in the system using a callback on codecs and machines called 'dapm_event', passing it PCI style power levels as SNDRV_CTL_POWER_ constants. This is more obscure than it needs to be and has caused confusion to driver authors, especially given that DAPM is also performing power management. Address this by renaming the callback function to 'set_bias_level' and using constants explicitly representing the off, standby, pre-on and on states which DAPM transitions through. Also unexport the API for setting bias level: there are currently no in-tree users of this API other than the core itself and it is likely that the core would need to be extended to cater for any users. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: Jarkko Nikula <jarkko.nikula@nokia.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-05-19 12:31:28 +02:00
ASoC: Add post-CODEC bias level callback for machine driver Currently the machine driver can only do bias level configuration before the CODEC bias level is brought up. This means that the machine cannot do any configuration which depends on the CODEC bias level being maintained. Provide a post-CODEC callback which allows the machine driver to do things like enable the FLL on a CODEC which is brought down to BIAS_OFF when idle. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-12-04 12:41:04 +00:00
ASoC: Add context parameter to card DAPM callbacks The card callback will get called for each DAPM context in the card so it can be useful for it to know which device is currently undergoing a transition. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 19:13:23 +01:00
ASoC: Add post-CODEC bias level callback for machine driver Currently the machine driver can only do bias level configuration before the CODEC bias level is brought up. This means that the machine cannot do any configuration which depends on the CODEC bias level being maintained. Provide a post-CODEC callback which allows the machine driver to do things like enable the FLL on a CODEC which is brought down to BIAS_OFF when idle. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-12-04 12:41:04 +00:00
[ALSA] soc - Add device level DAPM event Added a device level dapm event so that both the machine and codec are informed when dapm events occur. Signed-off-by: Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-01-10 14:36:20 +01:00
ASoC: Define add/remove_dai_link ops for a soc card A machine driver can register the two ops. When a DAI link is added or removed by a component's topology, the ASoC core can call the ops to notify the machine driver for extra intialization or destruction. E.g. topology can create FE DAI links from a cpu DAI component, and the machine driver may define an add_dai_link ops to set machine-specific .init ops for the DAI link. Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-12-02 14:11:32 +08:00
ASoC: Make pmdown_time a long Fixes a warning. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-17 14:30:44 +00:00
ASoC: Make pmdown_time a per-card setting Make the pmdown_time a per-card setting rather than a global one, initialised before the card initialisation runs. This allows cards to override the default setting if it makes sense to do so (for example, due to an unavoidable pop). Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-02-12 11:05:44 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: Implement DAI links in a list & define API to add/remove a link Implement a dai link list for the soc card. Add APIs to add/remove a DAI links dynamically, e.g. by topology. And a dobj is embedded into the struct snd_soc_dai_link. Topology can use the dobj to find the links created by it and remove them when the topology component is unloaded. The predefined DAI links are reserved to keep backward compatibility. And they will also be added to the list. Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-12-02 14:11:22 +08:00
ASoC: Change the PCM runtime array to a list Currently the number of DAI links is statically defined by the machine driver at build time using an array. This makes it difficult to shrink/ grow the number of DAI links at runtime in order to reflect any changes in topology. We can change the DAI link array in the core to a list so that PCMs and FE DAI links can be added and deleted at runtime to reflect changes in use case and DSP topology. The machine driver can still register DAI links as an array. As the 1st step, this patch change the PCM runtime array to a list. A new PCM runtime is added to the list when a DAI link is bound successfully. Later patches will further implement the DAI link list. More: - define snd_soc_new/free_pcm_runtime() to create/free a runtime. - define soc_add_pcm_runtime() to add a runtime to the rtd list. - define soc_remove_pcm_runtimes() to clean up the runtime list. - traverse the rtd list to probe the link components and dais. - Add a field "num" to PCM runtime struct, used to specify the device number when creating the pcm device, and for a soc card to access its dai_props array. - The following 3rd party machine/platform drivers iterate the rtd list to check the runtimes: sound/soc/intel/atom/sst-mfld-platform-pcm.c sound/soc/intel/boards/cht_bsw_rt5645.c sound/soc/intel/boards/cht_bsw_rt5672.c sound/soc/intel/boards/cht_bsw_max98090_ti.c Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-11-18 02:34:11 -05:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: Push workqueue data into snd_soc_card ASoC v2 does not use the struct snd_soc_device at runtime, using struct snd_soc_card as the root of the card. Begin removing data from snd_soc_device by pushing the workqueue data into snd_soc_card, using a backpointer to the snd_soc_device to keep things going for the time being. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-02 15:08:03 +00:00
ASoC: soc-core: Generalize snd_soc_prefix_map and rename to snd_soc_codec_conf The snd_soc_codec_conf struct now holds codec specific configuration information. A new configuration option has been added to allow machine drivers to override the compression type set by the codec driver. In the absence of providing an snd_soc_codec_conf struct or when providing one but not setting the compress_type member to anything, the one supplied by the codec driver will be used instead. In all other cases the one set in the snd_soc_codec_conf struct takes effect. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-12-02 14:53:03 +00:00
ASoC: Add optional name_prefix for codec kcontrol, widget and route names There is a need to prefix codec kcontrol, widget and internal route names in an ASoC machine that has multiple codecs with conflicting names. The name collision would occur when codec drivers try to registering kcontrols with the same name or when building audio paths. This patch introduces optional prefix_map into struct snd_soc_card. With it machine drivers can specify a unique name prefix to each codec that have conflicting names with anothers. Prefix to codec is matched with codec name. Following example illustrates a machine that has two same codec instances. Name collision from kcontrol registration is avoided by specifying a name prefix "foo" for the second codec. As the codec widget names are prefixed then second audio map for that codec shows a prefixed widget name. static const struct snd_soc_dapm_route map0[] = { {"Spk", NULL, "MONO"}, }; static const struct snd_soc_dapm_route map1[] = { {"Vibra", NULL, "foo MONO"}, }; static struct snd_soc_prefix_map codec_prefix[] = { { .dev_name = "codec.2", .name_prefix = "foo", }, }; static struct snd_soc_card card = { ... .prefix_map = codec_prefix, .num_prefixes = ARRAY_SIZE(codec_prefix), }; 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>
2010-11-13 20:40:44 +02:00
ASoC: Add support for optional auxiliary dailess codecs This makes possible to register auxiliary dailess codecs in a machine driver. Term dailess is used here for amplifiers and codecs without DAI or DAI being unused. Dailess auxiliary codecs are kept in struct snd_soc_aux_dev and those codecs are probed after initializing the DAI links. There are no major differences between DAI link codecs and dailess codecs in ASoC core point of view. DAPM handles them equally and sysfs and debugfs directories for dailess codecs are similar except the pmdown_time node is not created. Only suspend and resume functions are modified to traverse all probed codecs instead of DAI link codecs. Example below shows a dailess codec registration. struct snd_soc_aux_dev foo_aux_dev[] = { { .name = "Amp", .codec_name = "codec.2", .init = foo_init2, }, }; static struct snd_soc_card card = { ... .aux_dev = foo_aux_dev, .num_aux_devs = ARRAY_SIZE(foo_aux_dev), }; Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-25 17:47:38 +02:00
ASoC: Fix binding and probing of auxiliary components Currently binding of auxiliary devices doesn't work as in soc_bind_aux_dev() function a bound component is not being added to any list and in soc_probe_aux_devices() we are trying to walk the component_dev_list list to probe auxiliary components but at that time this list doesn't contain any auxiliary components since they are being added to the card only in soc_probe_component(). This patch adds a list to the card where are stored bound but not probed auxiliary devices, so that all aux devices can be probed. Fixes: 1a653aa44725 "ASoC: core: replace aux_comp_list to component_dev_list" Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-12-29 14:11:21 +01:00
ASoC: Add support for optional auxiliary dailess codecs This makes possible to register auxiliary dailess codecs in a machine driver. Term dailess is used here for amplifiers and codecs without DAI or DAI being unused. Dailess auxiliary codecs are kept in struct snd_soc_aux_dev and those codecs are probed after initializing the DAI links. There are no major differences between DAI link codecs and dailess codecs in ASoC core point of view. DAPM handles them equally and sysfs and debugfs directories for dailess codecs are similar except the pmdown_time node is not created. Only suspend and resume functions are modified to traverse all probed codecs instead of DAI link codecs. Example below shows a dailess codec registration. struct snd_soc_aux_dev foo_aux_dev[] = { { .name = "Amp", .codec_name = "codec.2", .init = foo_init2, }, }; static struct snd_soc_card card = { ... .aux_dev = foo_aux_dev, .num_aux_devs = ARRAY_SIZE(foo_aux_dev), }; Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-25 17:47:38 +02:00
ASoC: Add data based control initialisation for CODECs and cards Allow CODEC and card drivers to point to an array of controls from their driver structure rather than explicitly calling snd_soc_add_controls(). Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-07 19:18:44 +09:00
ASoC: Allow card DAPM widgets and routes to be set up at registration These will be added after all devices are registered and allow most DAI init functions in machine drivers to be replaced by simple data. Regular controls are not supported as the registration function still works in terms of CODECs. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-03-02 18:35:51 +00:00
ASoC: core: Add extra dapm properties for Device Tree The current helper functions, snd_soc_of_parse_audio_simple_widgets() and snd_soc_of_parse_audio_routing(), set dapm_widgets and dapm_routes without caring if they are already set by using build-in widgets and routes in the card driver. So there could be one of them, build-in one or Device Tree one, overrided by the other depending on which one was assigned later. This patch adds an extra pair of dapm_widgets and dapm_routes for DT use only so as to prevent unexpected overriding. Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-02-14 17:22:49 -08:00
ASoC: Allow card DAPM widgets and routes to be set up at registration These will be added after all devices are registered and allow most DAI init functions in machine drivers to be replaced by simple data. Regular controls are not supported as the registration function still works in terms of CODECs. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-03-02 18:35:51 +00:00
ASoC: Make struct snd_soc_card's dapm_widgets and dapm_routes const Those should not be modified (and are not) by the core code, so make them const. This also makes them consistent with the same members of snd_soc_codec. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-12 19:31:01 +02:00
ASoC: Allow card DAPM widgets and routes to be set up at registration These will be added after all devices are registered and allow most DAI init functions in machine drivers to be replaced by simple data. Regular controls are not supported as the registration function still works in terms of CODECs. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-03-02 18:35:51 +00:00
ASoC: Make struct snd_soc_card's dapm_widgets and dapm_routes const Those should not be modified (and are not) by the core code, so make them const. This also makes them consistent with the same members of snd_soc_codec. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-04-12 19:31:01 +02:00
ASoC: Allow card DAPM widgets and routes to be set up at registration These will be added after all devices are registered and allow most DAI init functions in machine drivers to be replaced by simple data. Regular controls are not supported as the registration function still works in terms of CODECs. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-03-02 18:35:51 +00:00
ASoC: core: Add extra dapm properties for Device Tree The current helper functions, snd_soc_of_parse_audio_simple_widgets() and snd_soc_of_parse_audio_routing(), set dapm_widgets and dapm_routes without caring if they are already set by using build-in widgets and routes in the card driver. So there could be one of them, build-in one or Device Tree one, overrided by the other depending on which one was assigned later. This patch adds an extra pair of dapm_widgets and dapm_routes for DT use only so as to prevent unexpected overriding. Signed-off-by: Nicolin Chen <nicoleotsuka@gmail.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-02-14 17:22:49 -08:00
ASoC: Implement fully_routed card property A card is fully routed if the DAPM route table describes all connections on the board. When a card is fully routed, some operations can be automated by the ASoC core. The first, and currently only, such operation is described below, and implemented by this patch. Codecs often have a large number of external pins, and not all of these pins will be connected on all board designs. Some machine drivers therefore call snd_soc_dapm_nc_pin() for all the unused pins, in order to tell the ASoC core never to activate them. However, when a card is fully routed, the information needed to derive the set of unused pins is present in card->dapm_routes. In this case, have the ASoC core automatically call snd_soc_dapm_nc_pin() for each unused codec pin. This has been tested with soc/tegra/tegra_wm8903.c and soc/tegra/trimslice.c. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-23 12:42:04 -07:00
ASoC: Allow card DAPM widgets and routes to be set up at registration These will be added after all devices are registered and allow most DAI init functions in machine drivers to be replaced by simple data. Regular controls are not supported as the registration function still works in terms of CODECs. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-03-02 18:35:51 +00:00
ASoC: Push workqueue data into snd_soc_card ASoC v2 does not use the struct snd_soc_device at runtime, using struct snd_soc_card as the root of the card. Begin removing data from snd_soc_device by pushing the workqueue data into snd_soc_card, using a backpointer to the snd_soc_device to keep things going for the time being. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-02 15:08:03 +00:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: core: replace codec_dev_list to component_dev_list on Card Current Card has Codec list (= codec_dev_list), but Codec will be removed in the future. Because of this reason, this patch adds new Component list in Card, and replace Codec list. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-11-30 06:22:36 +00:00
ASoC: core: add support to card rebind Current behaviour of ASoC core w.r.t to component removal is that it unregisters dependent sound card totally. There is no support to rebind the card if the component comes back. Typical use case is DSP restart or kernel modules itself. With this patch, core now maintains list of cards that are unbind due to any of its depended components are removed and card not unregistered yet. This list is cleared when the card is rebind successfully or when the card is unregistered from machine driver. This list of unbind cards are tried to bind once again after every new component is successfully added, giving a fair chance for card bind to be successful. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-12 10:15:00 +01:00
ASoC: Add sound card directory under debugfs/asoc/ There will be need to have sound card specific debugfs entries. This patch introduces a new debugfs/asoc/{card->name}/ directory but does not add yet any entries there. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 20:35:19 +02:00
ASoC: Move widgets from DAPM context to snd_soc_card Decoupling widgets from DAPM context is required when extending the ASoC core to cross-device paths. Even the list of widgets are now kept in struct snd_soc_card, the widget listing in sysfs and debugs remain sorted per device. This patch makes possible to build cross-device paths but does not extend yet the DAPM to handle codec bias and widget power changes of an another device. Cross-device paths are registered by listing the widgets from device A in a map for device B. In case of conflicting widget names between the devices, a uniform name prefix is needed to separate them. See commit ead9b91 "ASoC: Add optional name_prefix for kcontrol, widget and route names" for help. An example below shows a path that connects MONO out of A into Line In of B: static const struct snd_soc_dapm_route mapA[] = { {"MONO", NULL, "DAC"}, }; static const struct snd_soc_dapm_route mapB[] = { {"Line In", NULL, "MONO"}, }; 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>
2010-12-14 12:18:31 +02:00
ASoC: Move DAPM paths from DAPM context to snd_soc_card Decoupling DAPM paths from DAPM context is a first prerequisite when extending ASoC core to cross-device paths. This patch is almost a nullop and does not allow to construct cross-device setup but the path clean-up part in dapm_free_widgets is prepared to remove cross-device paths between a device being removed and others. 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>
2010-12-14 12:18:30 +02:00
ASoC: Extend DAPM to handle power changes on cross-device paths Power change event like stream start/stop or kcontrol change in a cross-device path originates from one device but codec bias and widget power changes must be populated to another devices on that path as well. This patch modifies the dapm_power_widgets so that all the widgets on a sound card are checked for a power change, not just those that are specific to originating device. Also bias management is extended to check all the devices. Only exception in bias management are widgetless codecs whose bias state is changed only if power change is originating from that context. DAPM context test is added to dapm_seq_run to take care of if power sequence extends to an another device which requires separate register writes. 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>
2010-12-14 12:18:32 +02:00
ASoC: Do DAPM power checks only for widgets changed since last run In order to reduce the number of DAPM power checks we run keep a list of widgets which have been changed since the last DAPM run and iterate over that rather than the full widget list. Whenever we change the power state for a widget we add all the source and sink widgets it has to the dirty list, ensuring that all widgets in the path are checked. This covers more widgets than we need to as some of the neighbour widgets won't be connected but it's simpler as a first step. On one system I tried this gave: Power Path Neighbour Before: 207 1939 2461 After: 114 1066 1327 which seems useful. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-03 21:06:40 +01:00
ASoC: Move DAPM paths from DAPM context to snd_soc_card Decoupling DAPM paths from DAPM context is a first prerequisite when extending ASoC core to cross-device paths. This patch is almost a nullop and does not allow to construct cross-device setup but the path clean-up part in dapm_free_widgets is prepared to remove cross-device paths between a device being removed and others. 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>
2010-12-14 12:18:30 +02:00
ASoC: topology: Add topology core The topology core parses the FW topology file for known block types and instanciates any common ALSA/ASoC objects that it discovers. The core also passes any block that is does not understand to client component drivers for enumeration. The core exports some APIs to client drivers in order to load and unload firmware topology data as use case require. Currently the core deals with the following object types :- o kcontrols. This includes TLV, enumerated and bytes controls. o DAPM widgets. All types with any associated kcontrol. o DAPM graph. o FE PCM. FE PCM capabilities and configuration can be defined. o BE DAI Link. BE DAI link capabilities and configuration can be defined. o Codec <-> codec style links capabilities and configuration. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-05-29 19:06:14 +01:00
ASoC: Add a per-card DAPM context This means that rather than adding the board specific DAPM widgets to a random CODEC DAPM context they can be added to the card itself which is a bit cleaner. Previously there only was one DAPM context and it was tied to the single supported CODEC. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-03-02 18:21:57 +00:00
ASoC: Trace and collect statistics for DAPM graph walking One of the longest standing areas for improvement in ASoC has been the DAPM algorithm - it repeats the same checks many times whenever it is run and makes no effort to limit the areas of the graph it checks meaning we do an awful lot of walks over the full graph. This has never mattered too much as the size of the graph has generally been small in relation to the size of the devices supported and the speed of CPUs but it is annoying. In preparation for work on improving this insert a trace point after the graph walk has been done. This gives us specific timing information for the walk, and in order to give quantifiable (non-benchmark) numbers also count every time we check a link or check the power for a widget and report those numbers. Substantial changes in the algorithm may require tweaks to the stats but they should be useful for simpler things. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-20 21:43:24 +01:00
ASoC: dapm: Move snd_soc_dapm_update from dapm context to card The update field of a DAPM context is only assigned while the card's dapm_mutex is locked, the field is also cleared again while the mutex is stil locked. So there will only ever be one DAPM context at a time with a non-NULL update field. So it is safe to move the update field from the DAPM context struct to the card struct. Doing so will allow further cleanups in this area. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-07-29 17:13:55 +02:00
ASoC: Add a per-card DAPM context This means that rather than adding the board specific DAPM widgets to a random CODEC DAPM context they can be added to the card itself which is a bit cleaner. Previously there only was one DAPM context and it was tied to the single supported CODEC. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-03-02 18:21:57 +00:00
ASoC: Add sound card directory under debugfs/asoc/ There will be need to have sound card specific debugfs entries. This patch introduces a new debugfs/asoc/{card->name}/ directory but does not add yet any entries there. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 20:35:19 +02:00
ASoC: Move pop time from DAPM context to sound card Based on discussion the dapm_pop_time in debugsfs should be per card rather than per device. Single pop time value for entire card is cleaner when the DAPM sequencing is extended to cross-device paths. debugfs/asoc/{card->name}/{codec dir}/dapm_pop_time -> debugfs/asoc/{card->name}/dapm_pop_time Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 20:35:21 +02:00
ASoC: Add sound card directory under debugfs/asoc/ There will be need to have sound card specific debugfs entries. This patch introduces a new debugfs/asoc/{card->name}/ directory but does not add yet any entries there. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 20:35:19 +02:00
ASoC: Move pop time from DAPM context to sound card Based on discussion the dapm_pop_time in debugsfs should be per card rather than per device. Single pop time value for entire card is cleaner when the DAPM sequencing is extended to cross-device paths. debugfs/asoc/{card->name}/{codec dir}/dapm_pop_time -> debugfs/asoc/{card->name}/dapm_pop_time Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 20:35:21 +02:00
ASoC: Add card driver data Provide driver data for cards within the card structure. To simplify the implementation of the PM operations we don't use the struct device driver data as this is used by the core to retrieve the card in callbacks from the device model and PM core. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-01-28 13:11:47 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: add for_each_card_prelinks() macro To be more readable code, this patch adds new for_each_card_prelinks() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-18 01:28:49 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: add for_each_card_links() macro To be more readable code, this patch adds new for_each_card_links() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-18 01:29:16 +00:00
ASoC: soc.h: fixup for_each_card_links() macro Macro is using "link", not "dai_link" Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-06-19 10:27:52 +09:00
ASoC: add for_each_card_links() macro To be more readable code, this patch adds new for_each_card_links() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-18 01:29:16 +00:00
ASoC: add for_each_card_rtds() macro To be more readable code, this patch adds new for_each_card_rtds() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-18 01:29:35 +00:00
ASoC: add for_each_card_links() macro To be more readable code, this patch adds new for_each_card_links() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-18 01:29:16 +00:00
ASoC: add for_each_card_components() macro To be more readable code, this patch adds new for_each_card_components() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-18 01:29:55 +00:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: Remove extra space in runtime struct definition My usual technique for finding definitions is to search for "name {" which breaks with the extra space. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-10 11:46:13 +00:00
ASoC: Dynamically allocate the rtd device for a non-empty release() The device model needs a release() function so it can free devices when they become dereferenced. Do that for rtds. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-06 17:12:45 -08:00
ASoC: Rename snd_soc_card to snd_soc_machine One of the issues with the ASoC v1 API which has been addressed in the ASoC v2 work that Liam Girdwood has done is that the ALSA card provided by ASoC is distributed around the ASoC structures. For example, machine wide data such as the struct snd_card are maintained as part of the CODEC data structure, preventing the use of multiple codecs. This has been addressed by refactoring the data structures so that all the data for the ALSA card is contained in a single structure snd_soc_card which replaces the existing snd_soc_machine and snd_soc_device. Begin the process of backporting this by renaming struct snd_soc_machine to struct snd_soc_card, better reflecting its function and bringing it closer to standard ALSA terminology. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-11-18 20:50:34 +00:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: core - PCM mutex per rtd In preparation for the new ASoC Dynamic PCM support (AKA DSP support). The new ASoC Dynamic PCM core allows DAIs to be dynamically re-routed at runtime between the PCM device end (or Frontend - FE) and the physical DAI (Backend - BE) using regular kcontrols (just like a hardware CODEC routes audio in the analog domain). The Dynamic PCM core therefore must be able to call PCM operations for both the Frontend and Backend(s) DAIs at the same time. Currently we have a global pcm_mutex that is used to serialise the ASoC PCM operations. This patch removes the global mutex and adds a mutex per RTD allowing the PCM operations to be reentrant and allow control of more than one DAI at at time. e.g. a frontend PCM hw_params() could configure multiple backend DAI hw_params() with similar or different hw parameters at the same time. Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-09 17:04:39 +01:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: dapm: Move CODEC to CODEC params from the widget to the runtime Larger CODECs may contain many several hundred widgets and which set of parameters is selected only needs to be recorded on a per DAI basis. As such move the selected CODEC to CODEC link params to be stored in the runtime rather than the DAPM widget, to save some memory. Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-05 15:21:02 +01:00
ASoC: dpcm: Add Dynamic PCM core operations. 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>
2012-04-25 12:12:49 +01:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: add definations for compressed operations Here we update the asoc structures to add compress stream definations First the struct snd_soc_dai_driver adds a new member to indicate if the dai is compressed or pcm. Next we add a new structre the struct snd_soc_compr_ops in the struct snd_soc_dai_link. This is to be used for machine driver to perform any opertaions required for setting up compressed audio streams next is the compressed data operations, they are added using struct snd_compr_ops in the struct snd_soc_platform_driver. Signed-off-by: Namarta Kohli <namartax.kohli@intel.com> Signed-off-by: Ramesh Babu K V <ramesh.babu@intel.com> Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-16 17:10:40 +05:30
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: core: Add initial support for DAI multicodec DAI link assumes a one to one mapping between CPU DAI and CODEC. In some cases, the same CPU DAI can be connected to several codecs. This is the case for example, if you connect two mono codecs to the same I2S link in order to have a stereo card. The current ASoC implementation does not allow such setup. Add support for DAI links composed of a single CPU DAI and multiple CODECs. Sound cards have to pass the CODECs array in the corresponding DAI link through a new 'snd_soc_dai_link_component' struct. Each CODEC in this array is described in the same manner single CODEC DAIs are (either DT/OF node or codec_name). Multi-codec links are not supported in the case of CODEC to CODEC links. Just print a warning if it happens. Based on an original code done by Misael. Signed-off-by: Benoit Cousson <bcousson@baylibre.com> Signed-off-by: Misael Lopez Cruz <misael.lopez@ti.com> Signed-off-by: Fabien Parent <fparent@baylibre.com> Tested-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-08 23:19:34 +02:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: dpcm: Add debugFS support for DPCM 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>
2012-04-25 12:12:50 +01:00
ASoC: Change the PCM runtime array to a list Currently the number of DAI links is statically defined by the machine driver at build time using an array. This makes it difficult to shrink/ grow the number of DAI links at runtime in order to reflect any changes in topology. We can change the DAI link array in the core to a list so that PCMs and FE DAI links can be added and deleted at runtime to reflect changes in use case and DSP topology. The machine driver can still register DAI links as an array. As the 1st step, this patch change the PCM runtime array to a list. A new PCM runtime is added to the list when a DAI link is bound successfully. Later patches will further implement the DAI link list. More: - define snd_soc_new/free_pcm_runtime() to create/free a runtime. - define soc_add_pcm_runtime() to add a runtime to the rtd list. - define soc_remove_pcm_runtimes() to clean up the runtime list. - traverse the rtd list to probe the link components and dais. - Add a field "num" to PCM runtime struct, used to specify the device number when creating the pcm device, and for a soc card to access its dai_props array. - The following 3rd party machine/platform drivers iterate the rtd list to check the runtimes: sound/soc/intel/atom/sst-mfld-platform-pcm.c sound/soc/intel/boards/cht_bsw_rt5645.c sound/soc/intel/boards/cht_bsw_rt5672.c sound/soc/intel/boards/cht_bsw_max98090_ti.c Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-11-18 02:34:11 -05:00
ASoC: soc-core: add snd_soc_rtdcom_xxx() Current snd_soc_pcm_runtime has platform / codec pointers, and we could use these specific pointer. But these will be replaced to more generic "component" soon, and will need more generic method to get each connected component pointer from rtd. This patch adds new snd_soc_rtdcom_xxx() to connect/disconnect component to rtd. It means same as previous "platform" / "codec" pointer style, but more generic. We can find necessary component pointer from rtd by using component driver name on snd_soc_rtdcom_lookup(). Here, the reason why it uses "driver name" is that "component name" was created by fmt_single_name() and difficult to use it from driver. Driver of course knows its "driver name", thus, using it is more easy. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Arnaud Pouliquen <arnaud.pouliquen@st.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-08-08 06:17:47 +00:00
ASoC: bunch up bit field for snd_soc_pcm_runtime We can reduce struct size in certain environment. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-10-24 05:24:36 +00:00
ASoC: soc.h: fe_compr can be bit field fe_compr is used at soc-compress, it can be bit field. This patch move it from int to bit field. > grep fe_compr -r sound/soc/* sound/soc/soc-compress.c: rtd->fe_compr = 1; sound/soc/soc-pcm.c: if (!fe->dpcm[stream].runtime && !fe->fe_compr) Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-05-13 16:07:27 +09:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: add for_each_rtd_codec_dai() macro ALSA SoC snd_soc_pcm_runtime has snd_soc_dai array for codec_dai. To be more readable code, this patch adds new for_each_rtd_codec_dai() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-03 02:12:56 +00:00
ASoC: rename for_each_rtd_codec_dai_reverse to rollback commit 0b7990e38971 ("ASoC: add for_each_rtd_codec_dai() macro") added for_each_rtd_codec_dai_reverse(). but _rollback() is better naming than _reverse(). This patch rename it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-18 01:28:30 +00:00
ASoC: fix oops w/ for_each_rtd_codec_dai_rollback() macro A kernel oops happens on an error case (usual missing BE mixer configuration required by Intel SST driver). Git bisect points to this macro and an operator precedence issue. for (; ((i--) >= 0) && ((dai) = rtd->codec_dais[i]);) The initial code replaced by this macro was while (--i >= 0) { codec_dai = rtd->codec_dais[i]; Fix the C operator precedence difference by reverting to pre-decrement Fixes: 0b7990e3897 ('ASoC: add for_each_rtd_codec_dai() macro') Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-10-26 16:54:43 -05:00
ASoC: add for_each_rtd_codec_dai() macro ALSA SoC snd_soc_pcm_runtime has snd_soc_dai array for codec_dai. To be more readable code, this patch adds new for_each_rtd_codec_dai() macro, and replace existing code to it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-03 02:12:56 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ALSA: ASoC: Convert bitfields in ASoC into full int width Convert bitfields in ASoC into full int width. This is a simple mechanical conversion. Two places in the DAPM code were fixed to properly use mask. Signed-off-by: Jon Smirl <jonsmirl@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-07-29 11:42:26 +01:00
ASoC: core: Fix for the volume limiting when invert is in use If the register for the volume needs invert, than the inversion need to be done from the chip maximum, and not from the platform dependent limit. Introduce soc_mixer_control.platform_max value, which initially equals to chip maximum. The snd_soc_limit_volume function only modify the platform_max, all volsw_info call returns this as well. The .max value holds the chip default (maximum), and it is used for the inversion, if it is needed. Additional check in the volsw_info call has been added to check the validity of the platform_max in case, when custom macros used by codec drivers are not initializing it correctly. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-05-10 14:39:24 +03:00
ASoC: dapm: Add support for virtual mixer controls This patch adds support for virtual DAPM mixer controls. They are similar to virtual DAPM enums. There is no hardware register backing the control, so changing the control's value wont have any direct effect on the hardware. But it still influences the DAPM graph by causing the path it sits on to be connected or disconnected. This in turn can cause power changes for some of the widgets on the DAPM graph, which will then modify the hardware state. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Tested-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-10-06 13:43:49 +02:00
ASoC: core: Add signed register volume control logic Some codecs use signed volume control representation with non standard register sizes, e.g. 6 or 7 bit signed integers. This patch adds generic signed register volume control logic to soc-core. Instead of a fixed width signed register control, this implementation uses a 'min' value and the signed bit location to translate it to an absolute volume. Using the 'sign_bit' we can calculate a correct mask for the register values and translate it back into signed integers of standard size. Signed-off-by: Markus Pargmann <mpa@pengutronix.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-01-16 16:02:10 +01:00
ASoC: dapm: Implement mixer input auto-disable Some devices have the problem that if a internal audio signal source is disabled the output of the source becomes undefined or goes to a undesired state (E.g. DAC output goes to ground instead of VMID). In this case it is necessary, in order to avoid unwanted clicks and pops, to disable any mixer input the signal feeds into or to active a mute control along the path to the output. Often it is still desirable to expose the same mixer input control to userspace, so cerain paths can sill be disabled manually. This means we can not use conventional DAPM to manage the mixer input control. This patch implements a method for letting DAPM overwrite the state of a userspace visible control. I.e. DAPM will disable the control if the path on which the control sits becomes inactive. Userspace will then only see a cached copy of the controls state. Once DAPM powers the path up again it will sync the userspace setting with the hardware and give control back to userspace. To implement this a new widget type is introduced. One widget of this type will be created for each DAPM kcontrol which has the auto-disable feature enabled. For each path that is controlled by the kcontrol the widget will be connected to the source of that path. The new widget type behaves like a supply widget, which means it will power up if one of its sinks are powered up and will only power down if all of its sinks are powered down. In order to only have the mixer input enabled when the source signal is valid the new widget type will be disabled before all other widget types and only be enabled after all other widget types. E.g. consider the following simplified example. A DAC is connected to a mixer and the mixer has a control to enable or disable the signal from the DAC. +-------+ +-----+ | | | DAC |-----[Ctrl]-| Mixer | +-----+ : | | | : +-------+ | : +-------------+ | Ctrl widget | +-------------+ If the control has the auto-disable feature enabled we'll create a widget for the control. This widget is connected to the DAC as it is the source for the mixer input. If the DAC powers up the control widget powers up and if the DAC powers down the control widget is powered down. As long as the control widget is powered down the hardware input control is kept disabled and if it is enabled userspace can freely change the control's state. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-08-05 11:27:31 +02:00
ASoC: topology: Add topology core The topology core parses the FW topology file for known block types and instanciates any common ALSA/ASoC objects that it discovers. The core also passes any block that is does not understand to client component drivers for enumeration. The core exports some APIs to client drivers in order to load and unload firmware topology data as use case require. Currently the core deals with the following object types :- o kcontrols. This includes TLV, enumerated and bytes controls. o DAPM widgets. All types with any associated kcontrol. o DAPM graph. o FE PCM. FE PCM capabilities and configuration can be defined. o BE DAI Link. BE DAI link capabilities and configuration can be defined. o Codec <-> codec style links capabilities and configuration. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-05-29 19:06:14 +01:00
ALSA: ASoC: Convert bitfields in ASoC into full int width Convert bitfields in ASoC into full int width. This is a simple mechanical conversion. Two places in the DAPM code were fixed to properly use mask. Signed-off-by: Jon Smirl <jonsmirl@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2008-07-29 11:42:26 +01:00
ASoC: core: Add SND_SOC_BYTES control for coefficient blocks Allow devices to export blocks of registers to the application layer, intended for use for reading and writing coefficient data which can't usefully be worked with by the kernel at runtime (for example, due to requiring complex and expensive calculations or being the results of callibration procedures). Currently drivers are using platform data to provide configurations for coefficient blocks which isn't at all convenient for runtime management or configuration development. Currently only devices using regmap are supported, an error will be generated for any attempt to work with a byte control on a non-regmap device. There's no fundamental block to other devices so support could be added if required. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-10-10 18:31:26 +01:00
ASoC: core: Add support for masking out parts of coefficient blocks Chip designers frequently include things like the enable and disable controls for algorithms in the register blocks which also hold the coefficients. Since it's desirable to split out the enable/disable control from userspace the plain SND_SOC_BYTES() isn't optimal for these devices. Add a SND_SOC_BYTES_MASK() which allows a bitmask from the first word of the block to be excluded from the control. This supports the needs of devices I've looked at and lets us have a reasonably simple API. Further controls can be added in future if that's needed. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2012-02-17 16:20:33 -08:00
ASoC: core: Add SND_SOC_BYTES control for coefficient blocks Allow devices to export blocks of registers to the application layer, intended for use for reading and writing coefficient data which can't usefully be worked with by the kernel at runtime (for example, due to requiring complex and expensive calculations or being the results of callibration procedures). Currently drivers are using platform data to provide configurations for coefficient blocks which isn't at all convenient for runtime management or configuration development. Currently only devices using regmap are supported, an error will be generated for any attempt to work with a byte control on a non-regmap device. There's no fundamental block to other devices so support could be added if required. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-10-10 18:31:26 +01:00
ASoC: add SND_SOC_BYTES_EXT we need _EXT version for SND_SOC_BYTES so that DSPs can use this to pass data for DSP modules Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02 10:58:11 +05:30
ASoC: topology: Add topology core The topology core parses the FW topology file for known block types and instanciates any common ALSA/ASoC objects that it discovers. The core also passes any block that is does not understand to client component drivers for enumeration. The core exports some APIs to client drivers in order to load and unload firmware topology data as use case require. Currently the core deals with the following object types :- o kcontrols. This includes TLV, enumerated and bytes controls. o DAPM widgets. All types with any associated kcontrol. o DAPM graph. o FE PCM. FE PCM capabilities and configuration can be defined. o BE DAI Link. BE DAI link capabilities and configuration can be defined. o Codec <-> codec style links capabilities and configuration. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-05-29 19:06:14 +01:00
ASoC: core: add a helper for extended byte controls using TLV ALSA supports arbitrary length TLVs for each kcontrol that can be used to pass metadata about the control (e.g. volumes, enum information). The same transport mechanism is now used for arbitrary length data by defining a new helper. Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah@intel.com> Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-07-15 21:34:48 +05:30
ASoC: core: Pass kcontrol to bytes tlv callbacks Add kcontrol to the tlv callbacks in soc_bytes_ext, as it is needed for referencing the corresponding control in the driver code Also fix the only upstream user in topology core Signed-off-by: Mythri P K <mythri.p.k@intel.com> Signed-off-by: Jeeja KP <jeeja.kp@intel.com> Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-11-09 23:20:00 +05:30
ASoC: add SND_SOC_BYTES_EXT we need _EXT version for SND_SOC_BYTES so that DSPs can use this to pass data for DSP modules Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-05-02 10:58:11 +05:30
ASoC: core: Add signed multi register control 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>
2012-04-20 11:32:13 +02:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00
ASoC: Add macros for defining virtual enums With the upcoming consolidation of normal MUXs and virtual MUXs we need to be able to distinguish between enums with and without a backing register at the enum level. Use the same approach as used for virtual mixer controls by setting the reg field of the enum to SND_SOC_NOPM for enums without a backing register. This patch adds a set of helper macros that can be used to define such enums. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-28 08:31:05 +01:00
ASoC: New enum type: value_enum This patch introduces a new enum type. In this enum type each enumerated items referred with a value. This new enum type can handle enums encoded in bitfield, or any other weird ways. twl4030 codec has several mux selection register, where the input/output mux is coded in a bitfield. With the normal enum type this type of mux can not be handled in a clean way. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-05 09:54:57 +02:00
ASoC: Rename soc_enum.max field with items The name "max" in struct soc_enum is rather confusing since it actually takes the number of items. With "max", one might try to assign (nitems - 1) value. Rename the field to a more appropriate one, "items", which is also used in struct snd_ctl_elem_info, too. This patch also rewrites some code like "if (x > e->nitems - 1)" with "if (x >= e->nitems)". Not only the latter improves the readability, it also fixes a potential bug when e->items is zero. Signed-off-by: Takashi Iwai <tiwai@suse.de> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-18 08:11:42 +01:00
ASoC: New enum type: value_enum This patch introduces a new enum type. In this enum type each enumerated items referred with a value. This new enum type can handle enums encoded in bitfield, or any other weird ways. twl4030 codec has several mux selection register, where the input/output mux is coded in a bitfield. With the normal enum type this type of mux can not be handled in a clean way. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-05 09:54:57 +02:00
ASoC: Declare const properly for enum texts The enum texts are supposed to be const char * const []. Without the second const, it gets compile warnings like sound/soc/codecs/max98095.c:607:2: warning: initialization discards qualifiers from pointer target type Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-03 12:50:29 +02:00
ASoC: New enum type: value_enum This patch introduces a new enum type. In this enum type each enumerated items referred with a value. This new enum type can handle enums encoded in bitfield, or any other weird ways. twl4030 codec has several mux selection register, where the input/output mux is coded in a bitfield. With the normal enum type this type of mux can not be handled in a clean way. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-05 09:54:57 +02:00
ASoC: dapm: Add support for autodisable mux controls Commit 57295073b6ac ("ASoC: dapm: Implement mixer input auto-disable") added support for autodisable controls, controls whose values are only written to the hardware when their respective widgets are powered up. But it only added support for controls based on the mixer abstraction. This patch add support for mux controls (DAPM controls based on the enum abstraction) to be auto-disabled as well. As each mux can only have a single control, there is no need to tie the autodisable widget to the inputs (as is done for the mixer controls) it can be tided directly to the mux widget itself. Note that it is assumed that the first entry in a autodisable mux control will always represent the off state for the mux and is what the mux will be set to whilst it is disabled. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Reviewed-by: Lars-Peter Clausen <lars@metafoo.de> Tested-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-05-01 12:37:26 +01:00
ASoC: topology: Add topology core The topology core parses the FW topology file for known block types and instanciates any common ALSA/ASoC objects that it discovers. The core also passes any block that is does not understand to client component drivers for enumeration. The core exports some APIs to client drivers in order to load and unload firmware topology data as use case require. Currently the core deals with the following object types :- o kcontrols. This includes TLV, enumerated and bytes controls. o DAPM widgets. All types with any associated kcontrol. o DAPM graph. o FE PCM. FE PCM capabilities and configuration can be defined. o BE DAI Link. BE DAI link capabilities and configuration can be defined. o Codec <-> codec style links capabilities and configuration. Signed-off-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-05-29 19:06:14 +01:00
ASoC: New enum type: value_enum This patch introduces a new enum type. In this enum type each enumerated items referred with a value. This new enum type can handle enums encoded in bitfield, or any other weird ways. twl4030 codec has several mux selection register, where the input/output mux is coded in a bitfield. With the normal enum type this type of mux can not be handled in a clean way. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-05 09:54:57 +02:00
ASoC: Add DAPM support at the component level This patch adds full DAPM support at the component level. Previously there was only full DAPM support for CODECs and partial DAPM support (e.g. no Mixers nor MUXs) for platforms. Having DAPM support at the component level will allow all types of components to use DAPM and also help in consolidating the DAPM support between CODECs and platforms. Since the DAPM context is directly embedded into the snd_soc_codec and snd_soc_platform struct and the 'dapm' field is directly referenced in a lot of drivers moving the field just right now is not possible without causing code churn. The approach this patch takes is to add two new fields to the component struct. One field which is the pointer to the actual DAPM context used by the component and one DAPM context that will be used as the default if no other context was specified. For CODECs and platforms the pointer is initialized to point to the CODEC or platform DAPM context. All generic code when referencing a component's DAPM struct will go via the pointer. This will make it possible to eventually seamlessly move the DAPM context from snd_soc_codec and snd_soc_platform struct over once all direct references have been eliminated. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-16 18:13:06 +02:00
ASoC: Remove snd_soc_codec dapm field There are no more direct users of the snd_soc_codec DAPM field left. So we can finally remove it and switch over to directly using the component DAPM context and remove the dapm_ptr indirection. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-07-06 15:38:11 +02:00
ASoC: Add DAPM support at the component level This patch adds full DAPM support at the component level. Previously there was only full DAPM support for CODECs and partial DAPM support (e.g. no Mixers nor MUXs) for platforms. Having DAPM support at the component level will allow all types of components to use DAPM and also help in consolidating the DAPM support between CODECs and platforms. Since the DAPM context is directly embedded into the snd_soc_codec and snd_soc_platform struct and the 'dapm' field is directly referenced in a lot of drivers moving the field just right now is not possible without causing code churn. The approach this patch takes is to add two new fields to the component struct. One field which is the pointer to the actual DAPM context used by the component and one DAPM context that will be used as the default if no other context was specified. For CODECs and platforms the pointer is initialized to point to the CODEC or platform DAPM context. All generic code when referencing a component's DAPM struct will go via the pointer. This will make it possible to eventually seamlessly move the DAPM context from snd_soc_codec and snd_soc_platform struct over once all direct references have been eliminated. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-06-16 18:13:06 +02:00
ASoC: add snd_soc_component_xxx_bias_level() snd_soc_codec_xxx_bias_level() (= for Codec) will be removed soon. This patch Component version of it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-11-06 01:48:37 +00:00
ASoC: add snd_soc_dapm_kcontrol_component() snd_soc_dapm_kcontrol_codec() (= for Codec) will be removed soon. This patch Component version of it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-11-06 01:49:06 +00:00
ASoC: add snd_soc_component_cache_sync() snd_soc_cache_sync() (= for Codec) will be removed soon. This patch Component version of it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-11-06 01:48:52 +00:00
ASoC: Move IO abstraction to the component level We currently have two very similar IO abstractions in ASoC, one for CODECs, the other for platforms. Moving this to the component level will allow us to unify those two. It will also enable us to move the standard kcontrol helpers as well as DAPM support to the component level. The new component level abstraction layer is primarily build around regmap. There is a per component pointer for the regmap instance for the underlying device. There are four new function snd_soc_component_read(), snd_soc_component_write(), snd_soc_component_update_bits() and snd_soc_component_update_bits_async(). They have the same signature as their regmap counter-part and will internally forward the call one-to-one to regmap. If the component it not using regmap it will fallback to using the custom IO callbacks. This is done to be able to support drivers that haven't been converted to regmap yet, but it is expected that this will eventually be removed in the future once all component drivers have been converted to regmap. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-22 13:23:13 +02:00
ASoC: add snd_soc_component_read32 Current codec drivers are using snd_soc_read(). It will be replaced into snd_soc_component_read(), but these 2 are using different style. For example, it will be - val = snd_soc_read(xxx, reg); + ret = snd_soc_component_read(xxx, reg, &val); + if (ret < 0) { + ... + } To more smooth replace, let's add snd_soc_component_read32 which is copied from snd_soc_read() - val = snd_soc_read(xxx, reg); + val = snd_soc_component_read32(xxx, reg); Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-11-06 01:48:19 +00:00
ASoC: Move IO abstraction to the component level We currently have two very similar IO abstractions in ASoC, one for CODECs, the other for platforms. Moving this to the component level will allow us to unify those two. It will also enable us to move the standard kcontrol helpers as well as DAPM support to the component level. The new component level abstraction layer is primarily build around regmap. There is a per component pointer for the regmap instance for the underlying device. There are four new function snd_soc_component_read(), snd_soc_component_write(), snd_soc_component_update_bits() and snd_soc_component_update_bits_async(). They have the same signature as their regmap counter-part and will internally forward the call one-to-one to regmap. If the component it not using regmap it will fallback to using the custom IO callbacks. This is done to be able to support drivers that haven't been converted to regmap yet, but it is expected that this will eventually be removed in the future once all component drivers have been converted to regmap. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-22 13:23:13 +02:00
ASoC: add Component level set_sysclk In current ALSA SoC, Codec only has set_sysclk feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-08-24 00:57:35 +00:00
ASoC: add Component level set_pll In current ALSA SoC, Codec only has set_pll feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-08-24 00:57:51 +00:00
ASoC: add missing snd_soc_component_set_jack commit 44c07365e9e2 ("ASoC: add Component level set_jack") added new snd_soc_component_set_jack(), but it didn't add definition on soc.h. This patch adds it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-09-20 06:08:42 +00:00
ASoC: add Component level set_sysclk In current ALSA SoC, Codec only has set_sysclk feature. Codec will be merged into Component in next generation ALSA SoC, thus current Codec specific feature need to be merged into it. This is glue patch for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-08-24 00:57:35 +00:00
ASoC: Disable regmap helpers if regmap is disabled If regmap is disabled there will be no users of the ASoC regmap helpers. Furthermore regmap_exit() will no be defined causing the following compile error: sound/soc/soc-core.c: In function 'snd_soc_component_exit_regmap': sound/soc/soc-core.c:2645:2: error: implicit declaration of function 'regmap_exit' [-Werror=implicit-function-declaration] So disable the helpers if regmap is disabled. Reported-by: kbuild test robot <fengguang.wu@intel.com> Fixes: 20feb881988c ASoC: Add helper functions for deferred regmap setup") Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2014-11-25 21:41:03 +01:00
ASoC: Add helper functions for deferred regmap setup Some drivers (most notably the AC'97 drivers) do not have access to their regmap struct when the component/codec is registered. For those drivers the automatic regmap setup will not work and needs to be done manually, typically from the component/CODEC drivers probe callback. This patch adds a set of helper function to handle deferred regmap initialization as well as early regmap tear-down. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2014-11-18 19:45:52 +01:00
ASoC: Disable regmap helpers if regmap is disabled If regmap is disabled there will be no users of the ASoC regmap helpers. Furthermore regmap_exit() will no be defined causing the following compile error: sound/soc/soc-core.c: In function 'snd_soc_component_exit_regmap': sound/soc/soc-core.c:2645:2: error: implicit declaration of function 'regmap_exit' [-Werror=implicit-function-declaration] So disable the helpers if regmap is disabled. Reported-by: kbuild test robot <fengguang.wu@intel.com> Fixes: 20feb881988c ASoC: Add helper functions for deferred regmap setup") Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@kernel.org>
2014-11-25 21:41:03 +01:00
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ASoC: Add card driver data Provide driver data for cards within the card structure. To simplify the implementation of the PM operations we don't use the struct device driver data as this is used by the core to retrieve the card in callbacks from the device model and PM core. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-01-28 13:11:47 +00:00
ASoC: Add snd_soc_component_{get,set}_drvdata() Add Add snd_soc_component_{get,set}_drvdata() similar to snd_soc_codec_{get,set}_drvdata() and snd_soc_platform_{get,set}_drvdata(). Also update them to use the new functions internally. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Reviewed-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-08-20 13:08:46 +02:00
ASoC: soc core add inline to handle card list initialzation Currently the soc_probe initializes the card hence it does the card list initialzation. But if machines directly register the card they would need to do these steps, so putting them as inline would save lot of code This patch adds an inline to do list initialzation Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Harsha Priya <harsha.priya@intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-01-13 22:48:52 +05:30
ASoC: Fix binding and probing of auxiliary components Currently binding of auxiliary devices doesn't work as in soc_bind_aux_dev() function a bound component is not being added to any list and in soc_probe_aux_devices() we are trying to walk the component_dev_list list to probe auxiliary components but at that time this list doesn't contain any auxiliary components since they are being added to the card only in soc_probe_component(). This patch adds a list to the card where are stored bound but not probed auxiliary devices, so that all aux devices can be probed. Fixes: 1a653aa44725 "ASoC: core: replace aux_comp_list to component_dev_list" Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-12-29 14:11:21 +01:00
ASoC: core: replace codec_dev_list to component_dev_list on Card Current Card has Codec list (= codec_dev_list), but Codec will be removed in the future. Because of this reason, this patch adds new Component list in Card, and replace Codec list. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-11-30 06:22:36 +00:00
ASoC: core: add support to card rebind Current behaviour of ASoC core w.r.t to component removal is that it unregisters dependent sound card totally. There is no support to rebind the card if the component comes back. Typical use case is DSP restart or kernel modules itself. With this patch, core now maintains list of cards that are unbind due to any of its depended components are removed and card not unregistered yet. This list is cleared when the card is rebind successfully or when the card is unregistered from machine driver. This list of unbind cards are tried to bind once again after every new component is successfully added, giving a fair chance for card bind to be successful. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-09-12 10:15:00 +01:00
ASoC: soc core add inline to handle card list initialzation Currently the soc_probe initializes the card hence it does the card list initialzation. But if machines directly register the card they would need to do these steps, so putting them as inline would save lot of code This patch adds an inline to do list initialzation Signed-off-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Harsha Priya <harsha.priya@intel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-01-13 22:48:52 +05:30
ASoC: core: Change SOC_SINGLE/DOUBLE_VALUE representation SOC_SINGLE/DOUBLE_VALUE is used for mixer controls, where the bits are within one register. Assign .rreg to be the same as .reg for these types. With this change we can tell if the mixer in question: is mono: mc->reg == mc->rreg && mc->shift == mc->rshift is stereo, within single register: mc->reg == mc->rreg && mc->shift != mc->rshift is stereo, in two registers: mc->reg != mc->rreg The patch provide a small inline function to query, if the mixer is stereo, or mono. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-10-05 10:29:22 +03:00
ASoC: Consolidate enum and value enum controls The implementations for enum and value enum controls are almost identical. The only difference is that the value enum uses an additional look-up table to map the control value to the register value, while the enum control uses a direct mapping. Enums and value enums can easily be distinguished at runtime, for value enums the values field of the snd_soc_enum struct contains the look-up table, while for enums it is NULL. This patch adds two new small helper functions called snd_soc_enum_item_to_val() and snd_soc_enum_val_to_item() which map between register value and control item. If the items field of the snd_soc_enum struct is NULL the function will do a direct mapping otherwise they'll use the look-up table to do the mapping. Using these small helper functions it is possible to use the same kcontrol handlers for both enums and value enums. The functions are added a inline functions in soc.h so they can also be used by the DAPM code to accomplish similar consolidation. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-28 08:31:03 +01:00
ASoC: Move active count from CODEC to component There is no reason why active count tracking should only be done for CODECs but not for other components. Moving the active count from the snd_soc_codec struct to the snd_soc_component struct reduces the differences between CODECs and other components and will eventually allow component to component DAI links (Which is a prerequisite for converting CODECs to components). Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-03-05 13:17:47 +01:00
ASoC: Move standard kcontrol helpers to the component level After moving the IO layer inside ASoC to the component level we can now easily move the standard control helpers also to the component level. This allows to reuse the same standard helper control implementations for other components. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-22 13:23:14 +02:00
ASoC: Remove platform code now everything is componentised As all drivers have been moved over to the new generic component code remove the now unused platform specific code. Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com> Acked-by: Vinod Koul <vkoul@kernel.org> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-04-24 16:39:02 +01:00
ASoC: Move standard kcontrol helpers to the component level After moving the IO layer inside ASoC to the component level we can now easily move the standard control helpers also to the component level. This allows to reuse the same standard helper control implementations for other components. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-04-22 13:23:14 +02:00
ASoC: Work around allmodconfig failure Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@ti.com>
2011-04-28 10:57:54 +01:00
ASoC: soc-core: remove OF adjusting for snd_soc_of_parse_card_name Because prototype of OF-graph sound card support didn't have Sound Card node, commit 8f5ebb1bee15b5720741a98414767bb86f6c2b23 ("ASoC: soc-core: adjust for graph on snd_soc_of_parse_card_name") adjusted to it on each functions. But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound card has node. Thus, this commit became no longer needed. This reverts commit 8f5ebb1bee15b5720741a98414767bb86f6c2b23. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-01-27 06:37:51 +00:00
ASoC: soc-core: remove OF adjusting for snd_soc_of_parse_audio_simple_widgets Because prototype of OF-graph sound card support didn't have Sound Card node, commit 1ef5bcd57be5c8b31286b7b47828064be25f266b ("ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_simple_widgets") adjusted to it on each functions. But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound card has node. Thus, this commit became no longer needed. This reverts commit 1ef5bcd57be5c8b31286b7b47828064be25f266b. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-01-27 06:37:34 +00:00
ASoC: export snd_soc_of_get_slot_mask Amlogic's axg card driver can't use snd_soc_of_parse_tdm_slot() directly because it needs to handle 4 mask for each direction. Yet the parsing of each mask is the same, so export snd_soc_of_get_slot_mask() to reuse the the existing code. Signed-off-by: Jerome Brunet <jbrunet@baylibre.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-07-17 17:43:02 +02:00
ASoC: core: add TDM slot parsing from DT supports For some CPU/CODEC DAI devices the TDM slot infomation maybe needed. This patch adds the slot parsing from DT supports. TDM slot properties: dai-tdm-slot-num : Number of slots in use. dai-tdm-slot-width : Width in bits for each slot. For instance: dai-tdm-slot-num = <2>; dai-tdm-slot-width = <8>; And for each spcified driver, there could be one .of_xlate_tdm_slot_mask() to specify a explicit mapping of the channels and the slots. If it's absent the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the tx and rx masks. For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit for an active slot as default, and the default active bits are at the LSB of the masks. Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-14 09:34:35 +08:00
ASoC: simple-card: Add tdm slot mask support to simple-card Adds DT binding for explicitly choosing a tdm mask for DAI and uses it in simple-card. The API for snd_soc_of_parse_tdm_slot() has also been changed. Signed-off-by: Jyri Sarha <jsarha@ti.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-09-09 21:27:43 +03:00
ASoC: core: add TDM slot parsing from DT supports For some CPU/CODEC DAI devices the TDM slot infomation maybe needed. This patch adds the slot parsing from DT supports. TDM slot properties: dai-tdm-slot-num : Number of slots in use. dai-tdm-slot-width : Width in bits for each slot. For instance: dai-tdm-slot-num = <2>; dai-tdm-slot-width = <8>; And for each spcified driver, there could be one .of_xlate_tdm_slot_mask() to specify a explicit mapping of the channels and the slots. If it's absent the default snd_soc_of_xlate_tdm_slot_mask() will be used to generating the tx and rx masks. For snd_soc_of_xlate_tdm_slot_mask(), the tx and rx masks will use a 1 bit for an active slot as default, and the default active bits are at the LSB of the masks. Signed-off-by: Xiubo Li <Li.Xiubo@freescale.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-14 09:34:35 +08:00
ASoC: soc-core: add snd_soc_of_parse_node_prefix() Current ASoC has snd_soc_of_parse_audio_prefix() to get codec_conf settings from DT which is used to avoid DAI naming conflict when CPU/Codec matching. Currently, it is parsing from "top node", but, we want to parse from "each sub node" if sound card had multi cpus/codecs. This patch adds new snd_soc_of_parse_node_prefix() to allow parsing settings from selected node. It is keeping existing snd_soc_of_parse_audio_prefix() by using macro. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-11-22 00:55:09 +00:00
ASoC: core: add snd_soc_of_parse_audio_prefix() Current ASoC can add name_prefix for DAPM, and it is necessary for route settings. This patch adds snd_soc_of_parse_audio_prefix() for this purpose. It will be used with snd_soc_of_parse_audio_routing(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Tested-by: Keita Kobayashi <keita.kobayashi.ym@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-07-15 07:07:42 +00:00
ASoC: soc.h: makes snd_soc_of_parse_audio_prefix() inline commit 3b7103562c03c ("ASoC: soc-core: add snd_soc_of_parse_node_prefix()") maked snd_soc_of_parse_audio_prefix() as #define. But it'd be better to make this a static inline rather than a #define. It helps with error messages and type safety. This patch makes it inline. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-11-26 01:21:22 +00:00
ASoC: soc-core: add snd_soc_of_parse_node_prefix() Current ASoC has snd_soc_of_parse_audio_prefix() to get codec_conf settings from DT which is used to avoid DAI naming conflict when CPU/Codec matching. Currently, it is parsing from "top node", but, we want to parse from "each sub node" if sound card had multi cpus/codecs. This patch adds new snd_soc_of_parse_node_prefix() to allow parsing settings from selected node. It is keeping existing snd_soc_of_parse_audio_prefix() by using macro. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-11-22 00:55:09 +00:00
ASoC: soc-core: remove OF adjusting for snd_soc_of_parse_audio_routing Because prototype of OF-graph sound card support didn't have Sound Card node, commit 7364c8dc255232db33bcd1c5b19eb8f34cf6108a ("ASoC: soc-core: adjust for graph on snd_soc_of_parse_audio_routing") adjusted to it on each functions. But final discussion result of ALSA SoC / OF-graph ML, OF-graph sound card has node. Thus, this commit became no longer needed. This reverts commit 7364c8dc255232db33bcd1c5b19eb8f34cf6108a. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-01-27 06:36:50 +00:00
ASoC: add snd_soc_of_parse_daifmt() for DeviceTree This patch adds snd_soc_of_parse_daifmt() and supports below style on DT. [prefix]format = "i2c"; [prefix]clock-gating = "continuous"; [prefix]bitclock-inversion; [prefix]bitclock-master; [prefix]frame-master; Each driver can use specific [prefix] (ex simple-card,cpu,dai,format = xxx;) This sample will be SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CONT | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2013-01-14 18:36:04 -08:00
ASoC: core: Update snd_soc_of_parse_daifmt() interface Adds struct device_node **bitclkmaster and struct device_node **framemaster function parameters. With the new syntax bitclock-master and frame-master properties can explicitly indicate the dai-link bit-clock and frame masters with a phandle. This patch also makes the minimal changes to simple-card for it to work with the updated snd_soc_of_parse_daifmt(). Simple-card appears to be the only user of snd_soc_of_parse_daifmt() for now. Signed-off-by: Jyri Sarha <jsarha@ti.com> Acked-by: Jean-Francois Moine <moinejf@free.fr> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-03-24 12:15:24 +02:00
ASoC: add snd_soc_get_dai_id() function ALSA SoC needs to know connected DAI ID for detecting. It is not a big problem if device/driver was only for sound, but getting DAI ID will be difficult if device includes both Video/Sound, like HDMI. To solve this issue, this patch adds new snd_soc_get_dai_id() and its related .of_xlate_dai_id callback on component driver. In below case, we can handle Sound port (= port@2) as ID = 0 if .of_xlate_dai_id has its support. hdmi { port@0 { /* VIDEO */ }; port@1 { /* VIDEO */ }; port@2 { /* SOUND */ }; }; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-05-18 01:39:25 +00:00
ASoC: soc-core: snd_soc_get_dai_name() become non static snd_soc_get_dai_name() is used from snd_soc_of_get_dai_name(), and it is assuming that DT is using "sound-dai" / "#sound-dai-cells". But graph base DT is using "remote-endpoint". This patch makes snd_soc_get_dai_name() non static for graph support. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-11-11 01:19:28 +00:00
ASoC: add .of_xlate_dai_name on snd_soc_component_driver ASoC sound driver requires CPU/CODEC drivers for probing, and each CPU/CODEC has some DAI on it. Then, "dai name matching" have been used to identify CPU-CODEC DAI pair on ASoC. But, the "dai port number matching" is now required from DeviceTree. The solution of this issue is to replace the dai port number into dai name. Now, CPU/CODEC are based on struct snd_soc_component, and it can care above as common issue. This patch adds .of_xlate_dai_name callback interface on struct snd_soc_component_driver, and snd_soc_of_get_dai_name() which is using .of_xlate_dai_name. Then, #sound-dai-cells which enables DAI specifier is required on CPU/CODEC device tree properties. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2013-09-10 17:39:56 -07:00
ASoC: core: add multi-codec support in DT This patch exports a core function which handles the DT description of multi-codec links (as: "sound-dai = <&hdmi 0>, <&spdif_codec>;") and creates a CODEC component array in the DAI link. Signed-off-by: Jean-Francois Moine <moinejf@free.fr> Signed-off-by: Mark Brown <broonie@kernel.org>
2014-11-25 13:16:12 +01:00
ASoC: Add snd_soc_of_put_dai_link_codecs() helper function The code for dereferencing device nodes in the 'codecs' array is moved to a separate function so we can avoid open coding that in drivers. Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2018-03-09 18:48:54 +01:00
ASoC: Add utility to set a card's name from device tree Implement snd_soc_of_parse_card_name(), a utility function that sets a card's name from device tree. The machine driver specifies the DT property to use, since this is binding-specific. Signed-off-by: Stephen Warren <swarren@nvidia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-12-12 15:55:34 -07:00
ASoC: Implement DAI links in a list & define API to add/remove a link Implement a dai link list for the soc card. Add APIs to add/remove a DAI links dynamically, e.g. by topology. And a dobj is embedded into the struct snd_soc_dai_link. Topology can use the dobj to find the links created by it and remove them when the topology component is unloaded. The predefined DAI links are reserved to keep backward compatibility. And they will also be added to the list. Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-12-02 14:11:22 +08:00
ASoC: Define API to find a dai link Define the API to find an existing DAI link of the soc card by matching the ID, name and stream name. Some cards may use unique ID for each DAI link, so matching ID is enough, and name or stream name are not necessary. But user need to specify name or stream name as well if not sure whether link ID is unique since most cards use 0 as the default link ID. Topology can use this API to find an existing BE link and configure it. Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-11-03 01:04:12 +08:00
ASoC: Implement DAI links in a list & define API to add/remove a link Implement a dai link list for the soc card. Add APIs to add/remove a DAI links dynamically, e.g. by topology. And a dobj is embedded into the struct snd_soc_dai_link. Topology can use the dobj to find the links created by it and remove them when the topology component is unloaded. The predefined DAI links are reserved to keep backward compatibility. And they will also be added to the list. Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-12-02 14:11:22 +08:00
ASoC: Support registering a DAI dynamically Define API snd_soc_register_dai() to add a DAI dynamically and create the DAI widgets. Topology can use this API to register DAIs when probing a component with topology info. These DAIs's playback & capture widgets will be freed when the sound card is unregistered and the DAIs will be freed when cleaning up the component. And a dobj is embedded into the struct snd_soc_dai_driver. Topology can use the dobj to find the DAI drivers created by it and free them when the topology component is removed. Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-12-31 16:40:43 +08:00
ASoC: Export snd_soc_find_dai() This API can be used by topology to find an existing BE dai by name and further configure it. Topology will also check DAI ID to avoid wrong match. Signed-off-by: Mengdong Lin <mengdong.lin@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-04-19 13:12:25 +08:00
ASoC: Move DAI structure definitions into new soc-dai.h ASoC v2 factors most of the contents of soc.h out into separate headers, including soc-dai.h for the DAI. Factor the existing DAI API out into this file in order to prepare for backporting of the ASoC v2 DAI API. Also backport some of Liam's improvements to the documentation. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-07-23 14:03:07 +01:00
ASoC: Add helper to find codec_dai from dai_name Create a helper function to remove duplicate code used in machine drivers Suggested-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Acked-by: Liam Girdwood <liam.r.girdwood@linux.intel.com> Reviewed-by: Andy Shevchenko <andriy.shevchenko@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2017-10-12 18:38:02 -05:00
ASoC: add helper to change platform name for all dailinks To reuse the same machine drivers with Atom/SST, Skylake and SOF, we need to change the default platform_name (or platforms->name in the "modern" representation). So far, this override was done with an automatic override, which was broken by a set of changes for DT platforms related to deferred probe handling. This automatic override is actually not really needed, the machine driver can already receive the platform name as a platform_data parameter. This is used e.g. for HDaudio support where we have different PCI aliases used for different platforms. We can reuse the same mechanism and modify the machine drivers to override the dailinks prior to registrating the card. This will require additional work for SOF, but with this helper it'll be just two lines of additional code per machine driver which is reused, not the end of the world. This helper can be simplified when all drivers have transitioned to the "modern" representation of dailinks. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-01-25 14:34:55 -06:00
ASoC: soc-core: remove legacy style dai_link All drivers switched to modern style dai_link (= struct snd_soc_dai_link_component). Let's remove legacy style dai_link. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-06-06 13:22:19 +09:00
ASoC: add helper to change platform name for all dailinks To reuse the same machine drivers with Atom/SST, Skylake and SOF, we need to change the default platform_name (or platforms->name in the "modern" representation). So far, this override was done with an automatic override, which was broken by a set of changes for DT platforms related to deferred probe handling. This automatic override is actually not really needed, the machine driver can already receive the platform name as a platform_data parameter. This is used e.g. for HDaudio support where we have different PCI aliases used for different platforms. We can reuse the same mechanism and modify the machine drivers to override the dailinks prior to registrating the card. This will require additional work for SOF, but with this helper it'll be just two lines of additional code per machine driver which is reused, not the end of the world. This helper can be simplified when all drivers have transitioned to the "modern" representation of dailinks. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2019-01-25 14:34:55 -06:00
ASoC: Export debugfs root dentry A couple Tegra ASoC drivers will create debugfs entries. Mark requested these by under debugfs/asoc/ not just debugfs/. To enable this, export the dentry representing debugfs/asoc/. Also, rename debugfs_root -> asoc_debugfs_root now it's exported to prevent potential symbol name clashes. 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>
2011-01-07 22:36:11 -07:00
ASoC: Update name of debugfs root symbol to snd_soc_ Everything else is using snd_soc_ so we should use it here too. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2011-01-10 22:25:21 +00:00
ASoC: Export debugfs root dentry A couple Tegra ASoC drivers will create debugfs entries. Mark requested these by under debugfs/asoc/ not just debugfs/. To enable this, export the dentry representing debugfs/asoc/. Also, rename debugfs_root -> asoc_debugfs_root now it's exported to prevent potential symbol name clashes. 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>
2011-01-07 22:36:11 -07:00
ASoC: Export card PM callbacks for use in direct registered cards 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>
2011-01-26 14:59:27 +00:00
ASoC: dapm: Add helpers to lock/unlock DAPM mutex Acquiring the DAPM mutex is necessary before using several DAPM functions and dereference is quite ugly. This patch provides a helper function to simplify this. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-18 15:22:13 +00:00
ASoC: Use nested lock for snd_soc_dapm_mutex_lock snd_soc_dapm_mutex_lock currently uses the un-nested call which can cause lockdep warnings when called from control handlers (a relatively common usage) and using modules. As creating the control causes a potential mutex inversion with the handler, creating the control will take the controls_rwsem under the dapm_mutex and accessing the control will take the dapm_mutex under controls_rwsem. All the users look like they want to be using the runtime class of the lock anyway, so this patch just changes snd_soc_dapm_mutex_lock to use the nested call, with the SND_SOC_DAPM_CLASS_RUNTIME class. Fixes: f6d5e586b416 ("ASoC: dapm: Add helpers to lock/unlock DAPM mutex") Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-12-29 16:25:13 +00:00
ASoC: dapm: Add helpers to lock/unlock DAPM mutex Acquiring the DAPM mutex is necessary before using several DAPM functions and dereference is quite ugly. This patch provides a helper function to simplify this. Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@linaro.org>
2014-02-18 15:22:13 +00:00
ASoC: core: Add component pin control functions It's often the case that a codec driver will need to control its own pins. However, if a name_prefix has been applied to this codec it must be included in the name passed to any of the snd_soc_dapm_x_pin() functions. The behaviour of the existing pin control functions is reasonable, since you may want to search for a fully-specified name within the scope of an entire card. This means that we can't apply the prefix in these functions because it will break card-scope searches. Constructing a prefixed string "manually" in codec drivers leads to a lot of repetition of the same code. To make this tidier in codec drivers this patch adds a new set of equivalent functions that take a struct snd_soc_component instead of a dapm context and automatically add the component's name_prefix to the given name. This makes it a simple change in codec drivers to be prefix-safe. The new functions are not quite trivial enough to be inlines and the compiler won't be able to compile-away any part of them. Although it looks somewhat inefficient to have to allocate a temporary buffer and combine strings, the current design of the widget list doesn't lend itself to a more optimized implementation - it's a single list of all widgets on a card and is searched linearly for a matching string. As pin state changes are generally low-frequency events it's unlikely to be a significant issue - at least not enough to rewrite the widget list handling just for this. Signed-off-by: Richard Fitzgerald <rf@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2016-11-29 15:44:38 +00:00
[ALSA] ASoC: core and dapm headers This patch adds the ASoC and DAPM headers. Features:- o Defines Digital Audio Interface (DAI) API o Defines Codec, Platform and Machine API o Defines Dynamic Audio Power Management API Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-06 18:20:14 +02:00