diff options
135 files changed, 4047 insertions, 2901 deletions
diff --git a/Documentation/devicetree/bindings/sound/cs42l73.txt b/Documentation/devicetree/bindings/sound/cs42l73.txt new file mode 100644 index 00000000000..80ae910dbf6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42l73.txt @@ -0,0 +1,22 @@ +CS42L73 audio CODEC + +Required properties: + + - compatible : "cirrus,cs42l73" + + - reg : the I2C address of the device for I2C + +Optional properties: + + - reset_gpio : a GPIO spec for the reset pin. + - chgfreq : Charge Pump Frequency values 0x00-0x0F + + +Example: + +codec: cs42l73@4a { + compatible = "cirrus,cs42l73"; + reg = <0x4a>; + reset_gpio = <&gpio 10 0>; + chgfreq = <0x05>; +};
\ No newline at end of file diff --git a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt new file mode 100644 index 00000000000..865178d5cdf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt @@ -0,0 +1,42 @@ +* Texas Instruments SoC audio setups with TLV320AIC3X Codec + +Required properties: +- compatible : "ti,da830-evm-audio" : forDM365/DA8xx/OMAPL1x/AM33xx +- ti,model : The user-visible name of this sound complex. +- ti,audio-codec : The phandle of the TLV320AIC3x audio codec +- ti,mcasp-controller : The phandle of the McASP controller +- ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec +- ti,audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names for sources and + sinks are the codec's pins, and the jacks on the board: + + Board connectors: + + * Headphone Jack + * Line Out + * Mic Jack + * Line In + + +Example: + +sound { + compatible = "ti,da830-evm-audio"; + ti,model = "DA830 EVM"; + ti,audio-codec = <&tlv320aic3x>; + ti,mcasp-controller = <&mcasp1>; + ti,codec-clock-rate = <12000000>; + ti,audio-routing = + "Headphone Jack", "HPLOUT", + "Headphone Jack", "HPROUT", + "Line Out", "LLOUT", + "Line Out", "RLOUT", + "MIC3L", "Mic Bias 2V", + "MIC3R", "Mic Bias 2V", + "Mic Bias 2V", "Mic Jack", + "LINE1L", "Line In", + "LINE2L", "Line In", + "LINE1R", "Line In", + "LINE2R", "Line In"; +}; diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 374e145c2ef..ed785b3f67b 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -4,17 +4,25 @@ Required properties: - compatible : "ti,dm646x-mcasp-audio" : for DM646x platforms "ti,da830-mcasp-audio" : for both DA830 & DA850 platforms - "ti,omap2-mcasp-audio" : for OMAP2 platforms (TI81xx, AM33xx) - -- reg : Should contain McASP registers offset and length -- interrupts : Interrupt number for McASP -- op-mode : I2S/DIT ops mode. -- tdm-slots : Slots for TDM operation. -- num-serializer : Serializers used by McASP. -- serial-dir : A list of serializer pin mode. The list number should be equal - to "num-serializer" parameter. Each entry is a number indication - serializer pin direction. (0 - INACTIVE, 1 - TX, 2 - RX) + "ti,am33xx-mcasp-audio" : for AM33xx platforms (AM33xx, TI81xx) +- reg : Should contain reg specifiers for the entries in the reg-names property. +- reg-names : Should contain: + * "mpu" for the main registers (required). For compatibility with + existing software, it is recommended this is the first entry. + * "dat" for separate data port register access (optional). +- op-mode : I2S/DIT ops mode. 0 for I2S mode. 1 for DIT mode used for S/PDIF, + IEC60958-1, and AES-3 formats. +- tdm-slots : Slots for TDM operation. Indicates number of channels transmitted + or received over one serializer. +- serial-dir : A list of serializer configuration. Each entry is a number + indication for serializer pin direction. + (0 - INACTIVE, 1 - TX, 2 - RX) +- dmas: two element list of DMA controller phandles and DMA request line + ordered pairs. +- dma-names: identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. The dma + identifiers must be "rx" and "tx". Optional properties: @@ -23,18 +31,23 @@ Optional properties: - rx-num-evt : FIFO levels. - sram-size-playback : size of sram to be allocated during playback - sram-size-capture : size of sram to be allocated during capture +- interrupts : Interrupt numbers for McASP, currently not used by the driver +- interrupt-names : Known interrupt names are "tx" and "rx" +- pinctrl-0: Should specify pin control group used for this controller. +- pinctrl-names: Should contain only one value - "default", for more details + please refer to pinctrl-bindings.txt + Example: mcasp0: mcasp0@1d00000 { compatible = "ti,da830-mcasp-audio"; - #address-cells = <1>; - #size-cells = <0>; reg = <0x100000 0x3000>; - interrupts = <82 83>; + reg-names "mpu"; + interrupts = <82>, <83>; + interrupts-names = "tx", "rx"; op-mode = <0>; /* MCASP_IIS_MODE */ tdm-slots = <2>; - num-serializer = <16>; serial-dir = < 0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */ 0 0 0 0 diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index 705a6b156c6..5e6040c2c2e 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -24,10 +24,36 @@ Optional properties: 3 - MICBIAS output is connected to AVDD, If this node is not mentioned or if the value is incorrect, then MicBias is powered down. +- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the + device as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +CODEC output pins: + * LLOUT + * RLOUT + * MONO_LOUT + * HPLOUT + * HPROUT + * HPLCOM + * HPRCOM + +CODEC input pins: + * MIC3L + * MIC3R + * LINE1L + * LINE2L + * LINE1R + * LINE2R + +The pins can be used in referring sound node's audio-routing property. Example: tlv320aic3x: tlv320aic3x@1b { compatible = "ti,tlv320aic3x"; reg = <0x1b>; + + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DRVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; }; diff --git a/Documentation/devicetree/bindings/sound/tpa6130a2.txt b/Documentation/devicetree/bindings/sound/tpa6130a2.txt new file mode 100644 index 00000000000..6dfa740e4b2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tpa6130a2.txt @@ -0,0 +1,27 @@ +Texas Instruments - tpa6130a2 Codec module + +The tpa6130a2 serial control bus communicates through I2C protocols + +Required properties: + +- compatible - "string" - One of: + "ti,tpa6130a2" - TPA6130A2 + "ti,tpa6140a2" - TPA6140A2 + + +- reg - <int> - I2C slave address + +- Vdd-supply - <phandle> - power supply regulator + +Optional properties: + +- power-gpio - gpio pin to power the device + +Example: + +tpa6130a2: tpa6130a2@60 { + compatible = "ti,tpa6130a2"; + reg = <0x60>; + Vdd-supply = <&vmmc2>; + power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; +}; diff --git a/Documentation/sound/alsa/soc/DPCM.txt b/Documentation/sound/alsa/soc/DPCM.txt new file mode 100644 index 00000000000..aa8546f2d14 --- /dev/null +++ b/Documentation/sound/alsa/soc/DPCM.txt @@ -0,0 +1,380 @@ +Dynamic PCM +=========== + +1. Description +============== + +Dynamic PCM allows an ALSA PCM device to digitally route its PCM audio to +various digital endpoints during the PCM stream runtime. e.g. PCM0 can route +digital audio to I2S DAI0, I2S DAI1 or PDM DAI2. This is useful for on SoC DSP +drivers that expose several ALSA PCMs and can route to multiple DAIs. + +The DPCM runtime routing is determined by the ALSA mixer settings in the same +way as the analog signal is routed in an ASoC codec driver. DPCM uses a DAPM +graph representing the DSP internal audio paths and uses the mixer settings to +determine the patch used by each ALSA PCM. + +DPCM re-uses all the existing component codec, platform and DAI drivers without +any modifications. + + +Phone Audio System with SoC based DSP +------------------------------------- + +Consider the following phone audio subsystem. This will be used in this +document for all examples :- + +| Front End PCMs | SoC DSP | Back End DAIs | Audio devices | + + ************* +PCM0 <------------> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <----DAI1-----> Codec Speakers + * DSP * +PCM2 <------------> * * <----DAI2-----> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +This diagram shows a simple smart phone audio subsystem. It supports Bluetooth, +FM digital radio, Speakers, Headset Jack, digital microphones and cellular +modem. This sound card exposes 4 DSP front end (FE) ALSA PCM devices and +supports 6 back end (BE) DAIs. Each FE PCM can digitally route audio data to any +of the BE DAIs. The FE PCM devices can also route audio to more than 1 BE DAI. + + + +Example - DPCM Switching playback from DAI0 to DAI1 +--------------------------------------------------- + +Audio is being played to the Headset. After a while the user removes the headset +and audio continues playing on the speakers. + +Playback on PCM0 to Headset would look like :- + + ************* +PCM0 <============> * * <====DAI0=====> Codec Headset + * * +PCM1 <------------> * * <----DAI1-----> Codec Speakers + * DSP * +PCM2 <------------> * * <----DAI2-----> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +The headset is removed from the jack by user so the speakers must now be used :- + + ************* +PCM0 <============> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <====DAI1=====> Codec Speakers + * DSP * +PCM2 <------------> * * <----DAI2-----> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +The audio driver processes this as follows :- + + 1) Machine driver receives Jack removal event. + + 2) Machine driver OR audio HAL disables the Headset path. + + 3) DPCM runs the PCM trigger(stop), hw_free(), shutdown() operations on DAI0 + for headset since the path is now disabled. + + 4) Machine driver or audio HAL enables the speaker path. + + 5) DPCM runs the PCM ops for startup(), hw_params(), prepapre() and + trigger(start) for DAI1 Speakers since the path is enabled. + +In this example, the machine driver or userspace audio HAL can alter the routing +and then DPCM will take care of managing the DAI PCM operations to either bring +the link up or down. Audio playback does not stop during this transition. + + + +DPCM machine driver +=================== + +The DPCM enabled ASoC machine driver is similar to normal machine drivers +except that we also have to :- + + 1) Define the FE and BE DAI links. + + 2) Define any FE/BE PCM operations. + + 3) Define widget graph connections. + + +1 FE and BE DAI links +--------------------- + +| Front End PCMs | SoC DSP | Back End DAIs | Audio devices | + + ************* +PCM0 <------------> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <----DAI1-----> Codec Speakers + * DSP * +PCM2 <------------> * * <----DAI2-----> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +For the example above we have to define 4 FE DAI links and 6 BE DAI links. The +FE DAI links are defined as follows :- + +static struct snd_soc_dai_link machine_dais[] = { + { + .name = "PCM0 System", + .stream_name = "System Playback", + .cpu_dai_name = "System Pin", + .platform_name = "dsp-audio", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + .....< other FE and BE DAI links here > +}; + +This FE DAI link is pretty similar to a regular DAI link except that we also +set the DAI link to a DPCM FE with the "dynamic = 1". The supported FE stream +directions should also be set with the "dpcm_playback" and "dpcm_capture" +flags. There is also an option to specify the ordering of the trigger call for +each FE. This allows the ASoC core to trigger the DSP before or after the other +components (as some DSPs have strong requirements for the ordering DAI/DSP +start and stop sequences). + +The FE DAI above sets the codec and code DAIs to dummy devices since the BE is +dynamic and will change depending on runtime config. + +The BE DAIs are configured as follows :- + +static struct snd_soc_dai_link machine_dais[] = { + .....< FE DAI links here > + { + .name = "Codec Headset", + .cpu_dai_name = "ssp-dai.0", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .codec_name = "rt5640.0-001c", + .codec_dai_name = "rt5640-aif1", + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = hswult_ssp0_fixup, + .ops = &haswell_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + .....< other BE DAI links here > +}; + +This BE DAI link connects DAI0 to the codec (in this case RT5460 AIF1). It sets +the "no_pcm" flag to mark it has a BE and sets flags for supported stream +directions using "dpcm_playback" and "dpcm_capture" above. + +The BE has also flags set for ignoreing suspend and PM down time. This allows +the BE to work in a hostless mode where the host CPU is not transferring data +like a BT phone call :- + + ************* +PCM0 <------------> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <----DAI1-----> Codec Speakers + * DSP * +PCM2 <------------> * * <====DAI2=====> MODEM + * * +PCM3 <------------> * * <====DAI3=====> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +This allows the host CPU to sleep whilst the DSP, MODEM DAI and the BT DAI are +still in operation. + +A BE DAI link can also set the codec to a dummy device if the code is a device +that is managed externally. + +Likewise a BE DAI can also set a dummy cpu DAI if the CPU DAI is managed by the +DSP firmware. + + +2 FE/BE PCM operations +---------------------- + +The BE above also exports some PCM operations and a "fixup" callback. The fixup +callback is used by the machine driver to (re)configure the DAI based upon the +FE hw params. i.e. the DSP may perform SRC or ASRC from the FE to BE. + +e.g. DSP converts all FE hw params to run at fixed rate of 48k, 16bit, stereo for +DAI0. This means all FE hw_params have to be fixed in the machine driver for +DAI0 so that the DAI is running at desired configuration regardless of the FE +configuration. + +static int dai0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The DSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set DAI0 to 16 bit */ + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +The other PCM operation are the same as for regular DAI links. Use as necessary. + + +3 Widget graph connections +-------------------------- + +The BE DAI links will normally be connected to the graph at initialisation time +by the ASoC DAPM core. However, if the BE codec or BE DAI is a dummy then this +has to be set explicitly in the driver :- + +/* BE for codec Headset - DAI0 is dummy and managed by DSP FW */ +{"DAI0 CODEC IN", NULL, "AIF1 Capture"}, +{"AIF1 Playback", NULL, "DAI0 CODEC OUT"}, + + +Writing a DPCM DSP driver +========================= + +The DPCM DSP driver looks much like a standard platform class ASoC driver +combined with elements from a codec class driver. A DSP platform driver must +implement :- + + 1) Front End PCM DAIs - i.e. struct snd_soc_dai_driver. + + 2) DAPM graph showing DSP audio routing from FE DAIs to BEs. + + 3) DAPM widgets from DSP graph. + + 4) Mixers for gains, routing, etc. + + 5) DMA configuration. + + 6) BE AIF widgets. + +Items 6 is important for routing the audio outside of the DSP. AIF need to be +defined for each BE and each stream direction. e.g for BE DAI0 above we would +have :- + +SND_SOC_DAPM_AIF_IN("DAI0 RX", NULL, 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("DAI0 TX", NULL, 0, SND_SOC_NOPM, 0, 0), + +The BE AIF are used to connect the DSP graph to the graphs for the other +component drivers (e.g. codec graph). + + +Hostless PCM streams +==================== + +A hostless PCM stream is a stream that is not routed through the host CPU. An +example of this would be a phone call from handset to modem. + + + ************* +PCM0 <------------> * * <----DAI0-----> Codec Headset + * * +PCM1 <------------> * * <====DAI1=====> Codec Speakers/Mic + * DSP * +PCM2 <------------> * * <====DAI2=====> MODEM + * * +PCM3 <------------> * * <----DAI3-----> BT + * * + * * <----DAI4-----> DMIC + * * + * * <----DAI5-----> FM + ************* + +In this case the PCM data is routed via the DSP. The host CPU in this use case +is only used for control and can sleep during the runtime of the stream. + +The host can control the hostless link either by :- + + 1) Configuring the link as a CODEC <-> CODEC style link. In this case the link + is enabled or disabled by the state of the DAPM graph. This usually means + there is a mixer control that can be used to connect or disconnect the path + between both DAIs. + + 2) Hostless FE. This FE has a virtual connection to the BE DAI links on the DAPM + graph. Control is then carried out by the FE as regualar PCM operations. + This method gives more control over the DAI links, but requires much more + userspace code to control the link. Its recommended to use CODEC<->CODEC + unless your HW needs more fine grained sequencing of the PCM ops. + + +CODEC <-> CODEC link +-------------------- + +This DAI link is enabled when DAPM detects a valid path within the DAPM graph. +The machine driver sets some additional parameters to the DAI link i.e. + +static const struct snd_soc_pcm_stream dai_params = { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 2, + .channels_max = 2, +}; + +static struct snd_soc_dai_link dais[] = { + < ... more DAI links above ... > + { + .name = "MODEM", + .stream_name = "MODEM", + .cpu_dai_name = "dai2", + .codec_dai_name = "modem-aif1", + .codec_name = "modem", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .params = &dai_params, + } + < ... more DAI links here ... > + +These parameters are used to configure the DAI hw_params() when DAPM detects a +valid path and then calls the PCM operations to start the link. DAPM will also +call the appropriate PCM operations to disable the DAI when the path is no +longer valid. + + +Hostless FE +----------- + +The DAI link(s) are enabled by a FE that does not read or write any PCM data. +This means creating a new FE that is connected with a virtual path to both +DAI links. The DAI links will be started when the FE PCM is started and stopped +when the FE PCM is stopped. Note that the FE PCM cannot read or write data in +this configuration. + + diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt index bce23a4a787..db5f9c9ae14 100644 --- a/Documentation/sound/alsa/soc/codec.txt +++ b/Documentation/sound/alsa/soc/codec.txt @@ -1,22 +1,23 @@ -ASoC Codec Driver -================= +ASoC Codec Class Driver +======================= -The codec driver is generic and hardware independent code that configures the -codec to provide audio capture and playback. It should contain no code that is -specific to the target platform or machine. All platform and machine specific -code should be added to the platform and machine drivers respectively. +The codec class driver is generic and hardware independent code that configures +the codec, FM, MODEM, BT or external DSP to provide audio capture and playback. +It should contain no code that is specific to the target platform or machine. +All platform and machine specific code should be added to the platform and +machine drivers respectively. -Each codec driver *must* provide the following features:- +Each codec class driver *must* provide the following features:- 1) Codec DAI and PCM configuration - 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs + 2) Codec control IO - using RegMap API 3) Mixers and audio controls 4) Codec audio operations + 5) DAPM description. + 6) DAPM event handler. Optionally, codec drivers can also provide:- - 5) DAPM description. - 6) DAPM event handler. 7) DAC Digital mute control. Its probably best to use this guide in conjunction with the existing codec @@ -64,26 +65,9 @@ struct snd_soc_dai_driver wm8731_dai = { 2 - Codec control IO -------------------- The codec can usually be controlled via an I2C or SPI style interface -(AC97 combines control with data in the DAI). The codec drivers provide -functions to read and write the codec registers along with supplying a -register cache:- - - /* IO control data and register cache */ - void *control_data; /* codec control (i2c/3wire) data */ - void *reg_cache; - -Codec read/write should do any data formatting and call the hardware -read write below to perform the IO. These functions are called by the -core and ALSA when performing DAPM or changing the mixer:- - - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); - -Codec hardware IO functions - usually points to either the I2C, SPI or AC97 -read/write:- - - hw_write_t hw_write; - hw_read_t hw_read; +(AC97 combines control with data in the DAI). The codec driver should use the +Regmap API for all codec IO. Please see include/linux/regmap.h and existing +codec drivers for example regmap usage. 3 - Mixers and audio controls @@ -127,7 +111,7 @@ Defines a stereo enumerated control 4 - Codec Audio Operations -------------------------- -The codec driver also supports the following ALSA operations:- +The codec driver also supports the following ALSA PCM operations:- /* SoC audio ops */ struct snd_soc_ops { diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 05bf5a0eee4..7dfd88ce31a 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -21,7 +21,7 @@ level power systems. There are 4 power domains within DAPM - 1. Codec domain - VREF, VMID (core codec and audio power) + 1. Codec bias domain - VREF, VMID (core codec and audio power) Usually controlled at codec probe/remove and suspend/resume, although can be set at stream time if power is not needed for sidetone, etc. @@ -63,14 +63,22 @@ Audio DAPM widgets fall into a number of types:- o Line - Line Input/Output (and optional Jack) o Speaker - Speaker o Supply - Power or clock supply widget used by other widgets. + o Regulator - External regulator that supplies power to audio components. + o Clock - External clock that supplies clock to audio componnents. + o AIF IN - Audio Interface Input (with TDM slot mask). + o AIF OUT - Audio Interface Output (with TDM slot mask). + o Siggen - Signal Generator. + o DAI IN - Digital Audio Interface Input. + o DAI OUT - Digital Audio Interface Output. + o DAI Link - DAI Link between two DAI structures */ o Pre - Special PRE widget (exec before all others) o Post - Special POST widget (exec after all others) (Widgets are defined in include/sound/soc-dapm.h) -Widgets are usually added in the codec driver and the machine driver. There are -convenience macros defined in soc-dapm.h that can be used to quickly build a -list of widgets of the codecs and machines DAPM widgets. +Widgets can be added to the sound card by any of the component driver types. +There are convenience macros defined in soc-dapm.h that can be used to quickly +build a list of widgets of the codecs and machines DAPM widgets. Most widgets have a name, register, shift and invert. Some widgets have extra parameters for stream name and kcontrols. @@ -80,11 +88,13 @@ parameters for stream name and kcontrols. ------------------------- Stream Widgets relate to the stream power domain and only consist of ADCs -(analog to digital converters) and DACs (digital to analog converters). +(analog to digital converters), DACs (digital to analog converters), +AIF IN and AIF OUT. Stream widgets have the following format:- SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert), +SND_SOC_DAPM_AIF_IN(name, stream, slot, reg, shift, invert) NOTE: the stream name must match the corresponding stream name in your codec snd_soc_codec_dai. @@ -94,6 +104,11 @@ e.g. stream widgets for HiFi playback and capture SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1), SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1), +e.g. stream widgets for AIF + +SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + 2.2 Path Domain Widgets ----------------------- @@ -121,12 +136,14 @@ If you dont want the mixer elements prefixed with the name of the mixer widget, you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same as for SND_SOC_DAPM_MIXER. -2.3 Platform/Machine domain Widgets ------------------------------------ + +2.3 Machine domain Widgets +-------------------------- Machine widgets are different from codec widgets in that they don't have a codec register bit associated with them. A machine widget is assigned to each -machine audio component (non codec) that can be independently powered. e.g. +machine audio component (non codec or DSP) that can be independently +powered. e.g. o Speaker Amp o Microphone Bias @@ -146,12 +163,12 @@ static int spitz_mic_bias(struct snd_soc_dapm_widget* w, int event) SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias), -2.4 Codec Domain ----------------- +2.4 Codec (BIAS) Domain +----------------------- -The codec power domain has no widgets and is handled by the codecs DAPM event -handler. This handler is called when the codec powerstate is changed wrt to any -stream event or by kernel PM events. +The codec bias power domain has no widgets and is handled by the codecs DAPM +event handler. This handler is called when the codec powerstate is changed wrt +to any stream event or by kernel PM events. 2.5 Virtual Widgets @@ -169,15 +186,16 @@ After all the widgets have been defined, they can then be added to the DAPM subsystem individually with a call to snd_soc_dapm_new_control(). -3. Codec Widget Interconnections -================================ +3. Codec/DSP Widget Interconnections +==================================== -Widgets are connected to each other within the codec and machine by audio paths -(called interconnections). Each interconnection must be defined in order to -create a map of all audio paths between widgets. +Widgets are connected to each other within the codec, platform and machine by +audio paths (called interconnections). Each interconnection must be defined in +order to create a map of all audio paths between widgets. -This is easiest with a diagram of the codec (and schematic of the machine audio -system), as it requires joining widgets together via their audio signal paths. +This is easiest with a diagram of the codec or DSP (and schematic of the machine +audio system), as it requires joining widgets together via their audio signal +paths. e.g., from the WM8731 output mixer (wm8731.c) @@ -247,16 +265,9 @@ machine and includes the codec. e.g. o Mic Jack o Codec Pins -When a codec pin is NC it can be marked as not used with a call to - -snd_soc_dapm_set_endpoint(codec, "Widget Name", 0); - -The last argument is 0 for inactive and 1 for active. This way the pin and its -input widget will never be powered up and consume power. - -This also applies to machine widgets. e.g. if a headphone is connected to a -jack then the jack can be marked active. If the headphone is removed, then -the headphone jack can be marked inactive. +Endpoints are added to the DAPM graph so that their usage can be determined in +order to save power. e.g. NC codecs pins will be switched OFF, unconnected +jacks can also be switched OFF. 5 DAPM Widget Events diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt index d50c14df341..74056dba52b 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/alsa/soc/machine.txt @@ -1,8 +1,10 @@ ASoC Machine Driver =================== -The ASoC machine (or board) driver is the code that glues together the platform -and codec drivers. +The ASoC machine (or board) driver is the code that glues together all the +component drivers (e.g. codecs, platforms and DAIs). It also describes the +relationships between each componnent which include audio paths, GPIOs, +interrupts, clocking, jacks and voltage regulators. The machine driver can contain codec and platform specific code. It registers the audio subsystem with the kernel as a platform device and is represented by diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt index d57efad37e0..3a08a2c9150 100644 --- a/Documentation/sound/alsa/soc/platform.txt +++ b/Documentation/sound/alsa/soc/platform.txt @@ -1,9 +1,9 @@ ASoC Platform Driver ==================== -An ASoC platform driver can be divided into audio DMA and SoC DAI configuration -and control. The platform drivers only target the SoC CPU and must have no board -specific code. +An ASoC platform driver class can be divided into audio DMA drivers, SoC DAI +drivers and DSP drivers. The platform drivers only target the SoC CPU and must +have no board specific code. Audio DMA ========= @@ -64,3 +64,16 @@ Each SoC DAI driver must provide the following features:- 5) Suspend and resume (optional) Please see codec.txt for a description of items 1 - 4. + + +SoC DSP Drivers +=============== + +Each SoC DSP driver usually supplies the following features :- + + 1) DAPM graph + 2) Mixer controls + 3) DMA IO to/from DSP buffers (if applicable) + 4) Definition of DSP front end (FE) PCM devices. + +Please see DPCM.txt for a description of item 4. diff --git a/drivers/base/regmap/internal.h b/drivers/base/regmap/internal.h index 57f777835d9..9010614f779 100644 --- a/drivers/base/regmap/internal.h +++ b/drivers/base/regmap/internal.h @@ -179,6 +179,9 @@ struct regmap_field { /* lsb */ unsigned int shift; unsigned int reg; + + unsigned int id_size; + unsigned int id_offset; }; #ifdef CONFIG_DEBUG_FS diff --git a/drivers/base/regmap/regmap.c b/drivers/base/regmap/regmap.c index 7d689a15c50..00152bf7ab1 100644 --- a/drivers/base/regmap/regmap.c +++ b/drivers/base/regmap/regmap.c @@ -821,6 +821,8 @@ static void regmap_field_init(struct regmap_field *rm_field, rm_field->reg = reg_field.reg; rm_field->shift = reg_field.lsb; rm_field->mask = ((BIT(field_bits) - 1) << reg_field.lsb); + rm_field->id_size = reg_field.id_size; + rm_field->id_offset = reg_field.id_offset; } /** @@ -1369,6 +1371,74 @@ int regmap_field_write(struct regmap_field *field, unsigned int val) } EXPORT_SYMBOL_GPL(regmap_field_write); +/** + * regmap_field_update_bits(): Perform a read/modify/write cycle + * on the register field + * + * @field: Register field to write to + * @mask: Bitmask to change + * @val: Value to be written + * + * A value of zero will be returned on success, a negative errno will + * be returned in error cases. + */ +int regmap_field_update_bits(struct regmap_field *field, unsigned int mask, unsigned int val) +{ + mask = (mask << field->shift) & field->mask; + + return regmap_update_bits(field->regmap, field->reg, + mask, val << field->shift); +} +EXPORT_SYMBOL_GPL(regmap_field_update_bits); + +/** + * regmap_fields_write(): Write a value to a single register field with port ID + * + * @field: Register field to write to + * @id: port ID + * @val: Value to be written + * + * A value of zero will be returned on success, a negative errno will + * be returned in error cases. + */ +int regmap_fields_write(struct regmap_field *field, unsigned int id, + unsigned int val) +{ + if (id >= field->id_size) + return -EINVAL; + + return regmap_update_bits(field->regmap, + field->reg + (field->id_offset * id), + field->mask, val << field->shift); +} +EXPORT_SYMBOL_GPL(regmap_fields_write); + +/** + * regmap_fields_update_bits(): Perform a read/modify/write cycle + * on the register field + * + * @field: Register field to write to + * @id: port ID + * @mask: Bitmask to change + * @val: Value to be written + * + * A value of zero will be returned on success, a negative errno will + * be returned in error cases. + */ +int regmap_fields_update_bits(struct regmap_field *field, unsigned int id, + unsigned int mask, unsigned int val) +{ + if (id >= field->id_size) + return -EINVAL; + + mask = (mask << field->shift) & field->mask; + + return regmap_update_bits(field->regmap, + field->reg + (field->id_offset * id), + mask, val << field->shift); +} +EXPORT_SYMBOL_GPL(regmap_fields_update_bits); + /* * regmap_bulk_write(): Write multiple registers to the device * @@ -1677,6 +1747,39 @@ int regmap_field_read(struct regmap_field *field, unsigned int *val) EXPORT_SYMBOL_GPL(regmap_field_read); /** + * regmap_fields_read(): Read a value to a single register field with port ID + * + * @field: Register field to read from + * @id: port ID + * @val: Pointer to store read value + * + * A value of zero will be returned on success, a negative errno will + * be returned in error cases. + */ +int regmap_fields_read(struct regmap_field *field, unsigned int id, + unsigned int *val) +{ + int ret; + unsigned int reg_val; + + if (id >= field->id_size) + return -EINVAL; + + ret = regmap_read(field->regmap, + field->reg + (field->id_offset * id), + ®_val); + if (ret != 0) + return ret; + + reg_val &= field->mask; + reg_val >>= field->shift; + *val = reg_val; + + return ret; +} +EXPORT_SYMBOL_GPL(regmap_fields_read); + +/** * regmap_bulk_read(): Read multiple registers from the device * * @map: Register map to write to diff --git a/drivers/mfd/mc13xxx-core.c b/drivers/mfd/mc13xxx-core.c index 2a9b100c482..dbbf8ee3f59 100644 --- a/drivers/mfd/mc13xxx-core.c +++ b/drivers/mfd/mc13xxx-core.c @@ -158,8 +158,6 @@ int mc13xxx_reg_read(struct mc13xxx *mc13xxx, unsigned int offset, u32 *val) { int ret; - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); - if (offset > MC13XXX_NUMREGS) return -EINVAL; @@ -172,8 +170,6 @@ EXPORT_SYMBOL(mc13xxx_reg_read); int mc13xxx_reg_write(struct mc13xxx *mc13xxx, unsigned int offset, u32 val) { - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); - dev_vdbg(mc13xxx->dev, "[0x%02x] <- 0x%06x\n", offset, val); if (offset > MC13XXX_NUMREGS || val > 0xffffff) @@ -186,7 +182,6 @@ EXPORT_SYMBOL(mc13xxx_reg_write); int mc13xxx_reg_rmw(struct mc13xxx *mc13xxx, unsigned int offset, u32 mask, u32 val) { - BUG_ON(!mutex_is_locked(&mc13xxx->lock)); BUG_ON(val & ~mask); dev_vdbg(mc13xxx->dev, "[0x%02x] <- 0x%06x (mask: 0x%06x)\n", offset, val, mask); diff --git a/drivers/mfd/mc13xxx-spi.c b/drivers/mfd/mc13xxx-spi.c index 77189daadf1..5f14ef6693c 100644 --- a/drivers/mfd/mc13xxx-spi.c +++ b/drivers/mfd/mc13xxx-spi.c @@ -94,10 +94,15 @@ static int mc13xxx_spi_write(void *context, const void *data, size_t count) { struct device *dev = context; struct spi_device *spi = to_spi_device(dev); + const char *reg = data; if (count != 4) return -ENOTSUPP; + /* include errata fix for spi audio problems */ + if (*reg == MC13783_AUDIO_CODEC || *reg == MC13783_AUDIO_DAC) + spi_write(spi, data, count); + return spi_write(spi, data, count); } diff --git a/include/linux/mfd/mc13xxx.h b/include/linux/mfd/mc13xxx.h index 41ed59276c0..67c17b5a6f4 100644 --- a/include/linux/mfd/mc13xxx.h +++ b/include/linux/mfd/mc13xxx.h @@ -41,6 +41,13 @@ int mc13xxx_adc_do_conversion(struct mc13xxx *mc13xxx, unsigned int mode, unsigned int channel, u8 ato, bool atox, unsigned int *sample); +#define MC13783_AUDIO_RX0 36 +#define MC13783_AUDIO_RX1 37 +#define MC13783_AUDIO_TX 38 +#define MC13783_SSI_NETWORK 39 +#define MC13783_AUDIO_CODEC 40 +#define MC13783_AUDIO_DAC 41 + #define MC13XXX_IRQ_ADCDONE 0 #define MC13XXX_IRQ_ADCBISDONE 1 #define MC13XXX_IRQ_TS 2 diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index 8db5ae03b6e..689a856b86f 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -84,6 +84,8 @@ struct snd_platform_data { u8 version; u8 txnumevt; u8 rxnumevt; + int tx_dma_channel; + int rx_dma_channel; }; enum { diff --git a/include/linux/regmap.h b/include/linux/regmap.h index a10380bfbea..a12bea07f79 100644 --- a/include/linux/regmap.h +++ b/include/linux/regmap.h @@ -425,11 +425,15 @@ bool regmap_reg_in_ranges(unsigned int reg, * @reg: Offset of the register within the regmap bank * @lsb: lsb of the register field. * @reg: msb of the register field. + * @id_size: port size if it has some ports + * @id_offset: address offset for each ports */ struct reg_field { unsigned int reg; unsigned int lsb; unsigned int msb; + unsigned int id_size; + unsigned int id_offset; }; #define REG_FIELD(_reg, _lsb, _msb) { \ @@ -448,6 +452,15 @@ void devm_regmap_field_free(struct device *dev, struct regmap_field *field); int regmap_field_read(struct regmap_field *field, unsigned int *val); int regmap_field_write(struct regmap_field *field, unsigned int val); +int regmap_field_update_bits(struct regmap_field *field, + unsigned int mask, unsigned int val); + +int regmap_fields_write(struct regmap_field *field, unsigned int id, + unsigned int val); +int regmap_fields_read(struct regmap_field *field, unsigned int id, + unsigned int *val); +int regmap_fields_update_bits(struct regmap_field *field, unsigned int id, + unsigned int mask, unsigned int val); /** * Description of an IRQ for the generic regmap irq_chip. diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h index 4c68955f733..7c2be4a5189 100644 --- a/include/sound/cs42l52.h +++ b/include/sound/cs42l52.h @@ -31,6 +31,8 @@ struct cs42l52_platform_data { /* Charge Pump Freq. Check datasheet Pg73 */ unsigned int chgfreq; + /* Reset GPIO */ + unsigned int reset_gpio; }; #endif /* __CS42L52_H */ diff --git a/include/sound/cs42l73.h b/include/sound/cs42l73.h new file mode 100644 index 00000000000..f354be4cdc9 --- /dev/null +++ b/include/sound/cs42l73.h @@ -0,0 +1,22 @@ +/* + * linux/sound/cs42l73.h -- Platform data for CS42L73 + * + * Copyright (c) 2012 Cirrus Logic Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __CS42L73_H +#define __CS42L73_H + +struct cs42l73_platform_data { + /* RST GPIO */ + unsigned int reset_gpio; + unsigned int chgfreq; + int jack_detection; + unsigned int mclk_freq; +}; + +#endif /* __CS42L73_H */ diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index f11c35cd553..15017311f2e 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -61,6 +61,8 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * @slave_id: Slave requester id for the DMA channel. * @filter_data: Custom DMA channel filter data, this will usually be used when * requesting the DMA channel. + * @chan_name: Custom channel name to use when requesting DMA channel. + * @fifo_size: FIFO size of the DAI controller in bytes */ struct snd_dmaengine_dai_dma_data { dma_addr_t addr; @@ -68,6 +70,8 @@ struct snd_dmaengine_dai_dma_data { u32 maxburst; unsigned int slave_id; void *filter_data; + const char *chan_name; + unsigned int fifo_size; }; void snd_dmaengine_pcm_set_config_from_dai_data( @@ -96,6 +100,10 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * playback. */ #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) +/* + * The PCM streams have custom channel names specified. + */ +#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM diff --git a/include/sound/rcar_snd.h b/include/sound/rcar_snd.h index fb0a312bcb8..12afab18945 100644 --- a/include/sound/rcar_snd.h +++ b/include/sound/rcar_snd.h @@ -36,7 +36,6 @@ #define RSND_SSI_CLK_PIN_SHARE (1 << 31) #define RSND_SSI_CLK_FROM_ADG (1 << 30) /* clock parent is master */ #define RSND_SSI_SYNC (1 << 29) /* SSI34_sync etc */ -#define RSND_SSI_DEPENDENT (1 << 28) /* SSI needs SRU/SCU */ #define RSND_SSI_PLAY (1 << 24) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ae9a227d35d..800c101bb09 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -105,6 +105,8 @@ int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); +int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); + /* Digital Audio interface formatting */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); @@ -131,6 +133,7 @@ struct snd_soc_dai_ops { int (*set_pll)(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div); + int (*set_bclk_ratio)(struct snd_soc_dai *dai, unsigned int ratio); /* * DAI format configuration @@ -166,6 +169,13 @@ struct snd_soc_dai_ops { struct snd_soc_dai *); int (*prepare)(struct snd_pcm_substream *, struct snd_soc_dai *); + /* + * NOTE: Commands passed to the trigger function are not necessarily + * compatible with the current state of the dai. For example this + * sequence of commands is possible: START STOP STOP. + * So do not unconditionally use refcounting functions in the trigger + * function, e.g. clk_enable/disable. + */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); int (*bespoke_trigger)(struct snd_pcm_substream *, int, @@ -276,6 +286,13 @@ static inline void snd_soc_dai_set_dma_data(struct snd_soc_dai *dai, dai->capture_dma_data = data; } +static inline void snd_soc_dai_init_dma_data(struct snd_soc_dai *dai, + void *playback, void *capture) +{ + dai->playback_dma_data = playback; + dai->capture_dma_data = capture; +} + static inline void snd_soc_dai_set_drvdata(struct snd_soc_dai *dai, void *data) { diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 27a72d5d4b0..2037c45adfe 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -286,6 +286,8 @@ struct device; .info = snd_soc_info_volsw, \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 1) } +#define SOC_DAPM_SINGLE_VIRT(xname, max) \ + SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0) #define SOC_DAPM_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, \ @@ -300,6 +302,8 @@ struct device; .tlv.p = (tlv_array), \ .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert, 0) } +#define SOC_DAPM_SINGLE_TLV_VIRT(xname, max, tlv_array) \ + SOC_DAPM_SINGLE(xname, SND_SOC_NOPM, 0, max, 0, tlv_array) #define SOC_DAPM_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ diff --git a/include/sound/soc.h b/include/sound/soc.h index d22cb0a06fe..1f741cb24f3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -13,6 +13,7 @@ #ifndef __LINUX_SND_SOC_H #define __LINUX_SND_SOC_H +#include <linux/of.h> #include <linux/platform_device.h> #include <linux/types.h> #include <linux/notifier.h> @@ -330,7 +331,6 @@ struct soc_enum; struct snd_soc_jack; struct snd_soc_jack_zone; struct snd_soc_jack_pin; -struct snd_soc_cache_ops; #include <sound/soc-dapm.h> #include <sound/soc-dpcm.h> @@ -348,10 +348,6 @@ enum snd_soc_control_type { SND_SOC_REGMAP, }; -enum snd_soc_compress_type { - SND_SOC_FLAT_COMPRESSION = 1, -}; - enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_PCM = 0, SND_SOC_PCM_CLASS_BE = 1, @@ -369,6 +365,7 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, int snd_soc_register_card(struct snd_soc_card *card); int snd_soc_unregister_card(struct snd_soc_card *card); +int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card); int snd_soc_suspend(struct device *dev); int snd_soc_resume(struct device *dev); int snd_soc_poweroff(struct device *dev); @@ -386,6 +383,9 @@ void snd_soc_unregister_codec(struct device *dev); int snd_soc_register_component(struct device *dev, const struct snd_soc_component_driver *cmpnt_drv, struct snd_soc_dai_driver *dai_drv, int num_dai); +int devm_snd_soc_register_component(struct device *dev, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai); void snd_soc_unregister_component(struct device *dev); int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, unsigned int reg); @@ -403,12 +403,6 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value); int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value); -int snd_soc_default_volatile_register(struct snd_soc_codec *codec, - unsigned int reg); -int snd_soc_default_readable_register(struct snd_soc_codec *codec, - unsigned int reg); -int snd_soc_default_writable_register(struct snd_soc_codec *codec, - unsigned int reg); int snd_soc_platform_read(struct snd_soc_platform *platform, unsigned int reg); int snd_soc_platform_write(struct snd_soc_platform *platform, @@ -542,22 +536,6 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); /** - * struct snd_soc_reg_access - Describes whether a given register is - * readable, writable or volatile. - * - * @reg: the register number - * @read: whether this register is readable - * @write: whether this register is writable - * @vol: whether this register is volatile - */ -struct snd_soc_reg_access { - u16 reg; - u16 read; - u16 write; - u16 vol; -}; - -/** * struct snd_soc_jack_pin - Describes a pin to update based on jack detection * * @pin: name of the pin to update @@ -657,17 +635,26 @@ struct snd_soc_compr_ops { int (*trigger)(struct snd_compr_stream *); }; -/* SoC cache ops */ -struct snd_soc_cache_ops { +/* component interface */ +struct snd_soc_component_driver { + const char *name; + + /* DT */ + int (*of_xlate_dai_name)(struct snd_soc_component *component, + struct of_phandle_args *args, + const char **dai_name); +}; + +struct snd_soc_component { const char *name; - enum snd_soc_compress_type id; - int (*init)(struct snd_soc_codec *codec); - int (*exit)(struct snd_soc_codec *codec); - int (*read)(struct snd_soc_codec *codec, unsigned int reg, - unsigned int *value); - int (*write)(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value); - int (*sync)(struct snd_soc_codec *codec); + int id; + struct device *dev; + struct list_head list; + + struct snd_soc_dai_driver *dai_drv; + int num_dai; + + const struct snd_soc_component_driver *driver; }; /* SoC Audio Codec device */ @@ -683,8 +670,6 @@ struct snd_soc_codec { struct list_head list; struct list_head card_list; int num_dai; - enum snd_soc_compress_type compress_type; - size_t reg_size; /* reg_cache_size * reg_word_size */ int (*volatile_register)(struct snd_soc_codec *, unsigned int); int (*readable_register)(struct snd_soc_codec *, unsigned int); int (*writable_register)(struct snd_soc_codec *, unsigned int); @@ -708,13 +693,13 @@ struct snd_soc_codec { unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); - int (*bulk_write_raw)(struct snd_soc_codec *, unsigned int, const void *, size_t); void *reg_cache; - const void *reg_def_copy; - const struct snd_soc_cache_ops *cache_ops; struct mutex cache_rw_mutex; int val_bytes; + /* component */ + struct snd_soc_component component; + /* dapm */ struct snd_soc_dapm_context dapm; unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */ @@ -733,6 +718,7 @@ struct snd_soc_codec_driver { int (*remove)(struct snd_soc_codec *); int (*suspend)(struct snd_soc_codec *); int (*resume)(struct snd_soc_codec *); + struct snd_soc_component_driver component_driver; /* Default control and setup, added after probe() is run */ const struct snd_kcontrol_new *controls; @@ -760,9 +746,6 @@ struct snd_soc_codec_driver { short reg_cache_step; short reg_word_size; const void *reg_cache_default; - short reg_access_size; - const struct snd_soc_reg_access *reg_access_default; - enum snd_soc_compress_type compress_type; /* codec bias level */ int (*set_bias_level)(struct snd_soc_codec *, @@ -849,20 +832,6 @@ struct snd_soc_platform { #endif }; -struct snd_soc_component_driver { - const char *name; -}; - -struct snd_soc_component { - const char *name; - int id; - int num_dai; - struct device *dev; - struct list_head list; - - const struct snd_soc_component_driver *driver; -}; - struct snd_soc_dai_link { /* config - must be set by machine driver */ const char *name; /* Codec name */ @@ -944,12 +913,6 @@ struct snd_soc_codec_conf { * associated per device */ const char *name_prefix; - - /* - * set this to the desired compression type if you want to - * override the one supplied in codec->driver->compress_type - */ - enum snd_soc_compress_type compress_type; }; struct snd_soc_aux_dev { @@ -1088,7 +1051,8 @@ struct snd_soc_pcm_runtime { /* mixer control */ struct soc_mixer_control { int min, max, platform_max; - unsigned int reg, rreg, shift, rshift; + int reg, rreg; + unsigned int shift, rshift; unsigned int invert:1; unsigned int autodisable:1; }; @@ -1121,8 +1085,6 @@ struct soc_enum { unsigned int snd_soc_read(struct snd_soc_codec *codec, unsigned int reg); unsigned int snd_soc_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val); -unsigned int snd_soc_bulk_write_raw(struct snd_soc_codec *codec, - unsigned int reg, const void *data, size_t len); /* device driver data */ @@ -1201,6 +1163,8 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname); unsigned int snd_soc_of_parse_daifmt(struct device_node *np, const char *prefix); +int snd_soc_of_get_dai_name(struct device_node *of_node, + const char **dai_name); #include <sound/soc-dai.h> diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 5fc2dcdd21c..03996b2bb04 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -14,6 +14,7 @@ struct snd_soc_codec; struct snd_soc_platform; struct snd_soc_card; struct snd_soc_dapm_widget; +struct snd_soc_dapm_path; /* * Log register events diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index e6f4633b8dd..99a466822a7 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -117,8 +117,7 @@ static inline void pxa_ac97_warm_pxa25x(void) { gsr_bits = 0; - GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN; - wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); + GCR |= GCR_WARM_RST; } static inline void pxa_ac97_cold_pxa25x(void) @@ -129,8 +128,6 @@ static inline void pxa_ac97_cold_pxa25x(void) gsr_bits = 0; GCR = GCR_COLD_RST; - GCR |= GCR_CDONE_IE|GCR_SDONE_IE; - wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); } #endif @@ -149,8 +146,6 @@ static inline void pxa_ac97_warm_pxa27x(void) static inline void pxa_ac97_cold_pxa27x(void) { - unsigned int timeout; - GCR &= GCR_COLD_RST; /* clear everything but nCRST */ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ @@ -161,29 +156,20 @@ static inline void pxa_ac97_cold_pxa27x(void) udelay(5); clk_disable(ac97conf_clk); GCR = GCR_COLD_RST | GCR_WARM_RST; - timeout = 100; /* wait for the codec-ready bit to be set */ - while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) - mdelay(1); } #endif #ifdef CONFIG_PXA3xx static inline void pxa_ac97_warm_pxa3xx(void) { - int timeout = 100; - gsr_bits = 0; /* Can't use interrupts */ GCR |= GCR_WARM_RST; - while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) - mdelay(1); } static inline void pxa_ac97_cold_pxa3xx(void) { - int timeout = 1000; - /* Hold CLKBPB for 100us */ GCR = 0; GCR = GCR_CLKBPB; @@ -199,14 +185,13 @@ static inline void pxa_ac97_cold_pxa3xx(void) GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); GCR = GCR_WARM_RST | GCR_COLD_RST; - while (!(GSR & (GSR_PCR | GSR_SCR)) && timeout--) - mdelay(10); } #endif bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) { unsigned long gsr; + unsigned int timeout = 100; #ifdef CONFIG_PXA25x if (cpu_is_pxa25x()) @@ -224,6 +209,10 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) else #endif BUG(); + + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); + gsr = GSR | gsr_bits; if (!(gsr & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", @@ -239,6 +228,7 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset); bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) { unsigned long gsr; + unsigned int timeout = 1000; #ifdef CONFIG_PXA25x if (cpu_is_pxa25x()) @@ -257,6 +247,9 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) #endif BUG(); + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); + gsr = GSR | gsr_bits; if (!(gsr & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 61a64d28190..8b9e70105dd 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,5 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o +snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 3109db7b901..612e5801003 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -50,7 +50,7 @@ static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, buf->area = dma_alloc_coherent(pcm->card->dev, size, &buf->addr, GFP_KERNEL); pr_debug("atmel-pcm: alloc dma buffer: area=%p, addr=%p, size=%zu\n", - (void *)buf->area, (void *)buf->addr, size); + (void *)buf->area, (void *)(long)buf->addr, size); if (!buf->area) return -ENOMEM; diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index 7222380131e..b4e36901a40 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -12,7 +12,6 @@ #include <linux/module.h> #include <linux/of.h> #include <linux/of_device.h> -#include <linux/pinctrl/consumer.h> #include <sound/soc.h> @@ -155,15 +154,8 @@ static int atmel_asoc_wm8904_probe(struct platform_device *pdev) struct snd_soc_card *card = &atmel_asoc_wm8904_card; struct snd_soc_dai_link *dailink = &atmel_asoc_wm8904_dailink; struct clk *clk_src; - struct pinctrl *pinctrl; int id, ret; - pinctrl = devm_pinctrl_get_select_default(&pdev->dev); - if (IS_ERR(pinctrl)) { - dev_err(&pdev->dev, "failed to request pinctrl\n"); - return PTR_ERR(pinctrl); - } - card->dev = &pdev->dev; ret = atmel_asoc_wm8904_dt_init(pdev); if (ret) { diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 802717eccbd..f15bff1548f 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -37,6 +37,7 @@ #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/i2c.h> +#include <linux/of.h> #include <linux/atmel-ssc.h> diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 2c20f01e1f7..06f938deda1 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -1,6 +1,6 @@ config SND_EP93XX_SOC tristate "SoC Audio support for the Cirrus Logic EP93xx series" - depends on ARCH_EP93XX && SND_SOC + depends on (ARCH_EP93XX || COMPILE_TEST) && SND_SOC select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index 0e9f56e0d4b..cfe517e6800 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -57,9 +57,22 @@ static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param) return false; } +static struct dma_chan *ep93xx_compat_request_channel( + struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) +{ + struct snd_dmaengine_dai_dma_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + return snd_dmaengine_pcm_request_channel(ep93xx_pcm_dma_filter, + dma_data); +} + static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = { .pcm_hardware = &ep93xx_pcm_hardware, .compat_filter_fn = ep93xx_pcm_dma_filter, + .compat_request_channel = ep93xx_compat_request_channel, .prealloc_buffer_size = 131072, }; diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 259d1ac4492..75d0ad5d2dc 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -16,6 +16,7 @@ #include <linux/mfd/88pm860x.h> #include <linux/slab.h> #include <linux/delay.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -140,6 +141,7 @@ struct pm860x_priv { unsigned int filter; struct snd_soc_codec *codec; struct i2c_client *i2c; + struct regmap *regmap; struct pm860x_chip *chip; struct pm860x_det det; @@ -269,48 +271,6 @@ static struct st_gain st_table[] = { { -86, 29, 0}, { -56, 30, 0}, { -28, 31, 0}, { 0, 0, 0}, }; -static int pm860x_volatile(unsigned int reg) -{ - BUG_ON(reg >= REG_CACHE_SIZE); - - switch (reg) { - case PM860X_AUDIO_SUPPLIES_2: - return 1; - } - - return 0; -} - -static unsigned int pm860x_read_reg_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - unsigned char *cache = codec->reg_cache; - - BUG_ON(reg >= REG_CACHE_SIZE); - - if (pm860x_volatile(reg)) - return cache[reg]; - - reg += REG_CACHE_BASE; - - return pm860x_reg_read(codec->control_data, reg); -} - -static int pm860x_write_reg_cache(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - unsigned char *cache = codec->reg_cache; - - BUG_ON(reg >= REG_CACHE_SIZE); - - if (!pm860x_volatile(reg)) - cache[reg] = (unsigned char)value; - - reg += REG_CACHE_BASE; - - return pm860x_reg_write(codec->control_data, reg, value); -} - static int snd_soc_get_volsw_2r_st(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1169,6 +1129,7 @@ static int pm860x_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, static int pm860x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); int data; switch (level) { @@ -1182,17 +1143,17 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ data = AUDIO_PLL | AUDIO_SECTION_ON; - pm860x_reg_write(codec->control_data, REG_MISC2, data); + pm860x_reg_write(pm860x->i2c, REG_MISC2, data); udelay(300); data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; - pm860x_reg_write(codec->control_data, REG_MISC2, data); + pm860x_reg_write(pm860x->i2c, REG_MISC2, data); } break; case SND_SOC_BIAS_OFF: data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; - pm860x_set_bits(codec->control_data, REG_MISC2, data, 0); + pm860x_set_bits(pm860x->i2c, REG_MISC2, data, 0); break; } codec->dapm.bias_level = level; @@ -1322,17 +1283,17 @@ int pm860x_hs_jack_detect(struct snd_soc_codec *codec, pm860x->det.lo_shrt = lo_shrt; if (det & SND_JACK_HEADPHONE) - pm860x_set_bits(codec->control_data, REG_HS_DET, + pm860x_set_bits(pm860x->i2c, REG_HS_DET, EN_HS_DET, EN_HS_DET); /* headset short detect */ if (hs_shrt) { data = CLR_SHORT_HS2 | CLR_SHORT_HS1; - pm860x_set_bits(codec->control_data, REG_SHORTS, data, data); + pm860x_set_bits(pm860x->i2c, REG_SHORTS, data, data); } /* Lineout short detect */ if (lo_shrt) { data = CLR_SHORT_LO2 | CLR_SHORT_LO1; - pm860x_set_bits(codec->control_data, REG_SHORTS, data, data); + pm860x_set_bits(pm860x->i2c, REG_SHORTS, data, data); } /* sync status */ @@ -1350,7 +1311,7 @@ int pm860x_mic_jack_detect(struct snd_soc_codec *codec, pm860x->det.mic_det = det; if (det & SND_JACK_MICROPHONE) - pm860x_set_bits(codec->control_data, REG_MIC_DET, + pm860x_set_bits(pm860x->i2c, REG_MIC_DET, MICDET_MASK, MICDET_MASK); /* sync status */ @@ -1366,7 +1327,7 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x->codec = codec; - codec->control_data = pm860x->i2c; + codec->control_data = pm860x->regmap; for (i = 0; i < 4; i++) { ret = request_threaded_irq(pm860x->irq[i], NULL, @@ -1380,14 +1341,6 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - ret = pm860x_bulk_read(codec->control_data, REG_CACHE_BASE, - REG_CACHE_SIZE, codec->reg_cache); - if (ret < 0) { - dev_err(codec->dev, "Failed to fill register cache: %d\n", - ret); - goto out; - } - return 0; out: @@ -1410,10 +1363,6 @@ static int pm860x_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_pm860x = { .probe = pm860x_probe, .remove = pm860x_remove, - .read = pm860x_read_reg_cache, - .write = pm860x_write_reg_cache, - .reg_cache_size = REG_CACHE_SIZE, - .reg_word_size = sizeof(u8), .set_bias_level = pm860x_set_bias_level, .controls = pm860x_snd_controls, @@ -1439,6 +1388,8 @@ static int pm860x_codec_probe(struct platform_device *pdev) pm860x->chip = chip; pm860x->i2c = (chip->id == CHIP_PM8607) ? chip->client : chip->companion; + pm860x->regmap = (chip->id == CHIP_PM8607) ? chip->regmap + : chip->regmap_companion; platform_set_drvdata(pdev, pm860x); for (i = 0; i < 4; i++) { diff --git a/sound/soc/codecs/88pm860x-codec.h b/sound/soc/codecs/88pm860x-codec.h index 3364ba4a360..f7282f4f4a7 100644 --- a/sound/soc/codecs/88pm860x-codec.h +++ b/sound/soc/codecs/88pm860x-codec.h @@ -12,67 +12,66 @@ #ifndef __88PM860X_H #define __88PM860X_H -/* The offset of these registers are 0xb0 */ -#define PM860X_PCM_IFACE_1 0x00 -#define PM860X_PCM_IFACE_2 0x01 -#define PM860X_PCM_IFACE_3 0x02 -#define PM860X_PCM_RATE 0x03 -#define PM860X_EC_PATH 0x04 -#define PM860X_SIDETONE_L_GAIN 0x05 -#define PM860X_SIDETONE_R_GAIN 0x06 -#define PM860X_SIDETONE_SHIFT 0x07 -#define PM860X_ADC_OFFSET_1 0x08 -#define PM860X_ADC_OFFSET_2 0x09 -#define PM860X_DMIC_DELAY 0x0a +#define PM860X_PCM_IFACE_1 0xb0 +#define PM860X_PCM_IFACE_2 0xb1 +#define PM860X_PCM_IFACE_3 0xb2 +#define PM860X_PCM_RATE 0xb3 +#define PM860X_EC_PATH 0xb4 +#define PM860X_SIDETONE_L_GAIN 0xb5 +#define PM860X_SIDETONE_R_GAIN 0xb6 +#define PM860X_SIDETONE_SHIFT 0xb7 +#define PM860X_ADC_OFFSET_1 0xb8 +#define PM860X_ADC_OFFSET_2 0xb9 +#define PM860X_DMIC_DELAY 0xba -#define PM860X_I2S_IFACE_1 0x0b -#define PM860X_I2S_IFACE_2 0x0c -#define PM860X_I2S_IFACE_3 0x0d -#define PM860X_I2S_IFACE_4 0x0e -#define PM860X_EQUALIZER_N0_1 0x0f -#define PM860X_EQUALIZER_N0_2 0x10 -#define PM860X_EQUALIZER_N1_1 0x11 -#define PM860X_EQUALIZER_N1_2 0x12 -#define PM860X_EQUALIZER_D1_1 0x13 -#define PM860X_EQUALIZER_D1_2 0x14 -#define PM860X_LOFI_GAIN_LEFT 0x15 -#define PM860X_LOFI_GAIN_RIGHT 0x16 -#define PM860X_HIFIL_GAIN_LEFT 0x17 -#define PM860X_HIFIL_GAIN_RIGHT 0x18 -#define PM860X_HIFIR_GAIN_LEFT 0x19 -#define PM860X_HIFIR_GAIN_RIGHT 0x1a -#define PM860X_DAC_OFFSET 0x1b -#define PM860X_OFFSET_LEFT_1 0x1c -#define PM860X_OFFSET_LEFT_2 0x1d -#define PM860X_OFFSET_RIGHT_1 0x1e -#define PM860X_OFFSET_RIGHT_2 0x1f -#define PM860X_ADC_ANA_1 0x20 -#define PM860X_ADC_ANA_2 0x21 -#define PM860X_ADC_ANA_3 0x22 -#define PM860X_ADC_ANA_4 0x23 -#define PM860X_ANA_TO_ANA 0x24 -#define PM860X_HS1_CTRL 0x25 -#define PM860X_HS2_CTRL 0x26 -#define PM860X_LO1_CTRL 0x27 -#define PM860X_LO2_CTRL 0x28 -#define PM860X_EAR_CTRL_1 0x29 -#define PM860X_EAR_CTRL_2 0x2a -#define PM860X_AUDIO_SUPPLIES_1 0x2b -#define PM860X_AUDIO_SUPPLIES_2 0x2c -#define PM860X_ADC_EN_1 0x2d -#define PM860X_ADC_EN_2 0x2e -#define PM860X_DAC_EN_1 0x2f -#define PM860X_DAC_EN_2 0x31 -#define PM860X_AUDIO_CAL_1 0x32 -#define PM860X_AUDIO_CAL_2 0x33 -#define PM860X_AUDIO_CAL_3 0x34 -#define PM860X_AUDIO_CAL_4 0x35 -#define PM860X_AUDIO_CAL_5 0x36 -#define PM860X_ANA_INPUT_SEL_1 0x37 -#define PM860X_ANA_INPUT_SEL_2 0x38 +#define PM860X_I2S_IFACE_1 0xbb +#define PM860X_I2S_IFACE_2 0xbc +#define PM860X_I2S_IFACE_3 0xbd +#define PM860X_I2S_IFACE_4 0xbe +#define PM860X_EQUALIZER_N0_1 0xbf +#define PM860X_EQUALIZER_N0_2 0xc0 +#define PM860X_EQUALIZER_N1_1 0xc1 +#define PM860X_EQUALIZER_N1_2 0xc2 +#define PM860X_EQUALIZER_D1_1 0xc3 +#define PM860X_EQUALIZER_D1_2 0xc4 +#define PM860X_LOFI_GAIN_LEFT 0xc5 +#define PM860X_LOFI_GAIN_RIGHT 0xc6 +#define PM860X_HIFIL_GAIN_LEFT 0xc7 +#define PM860X_HIFIL_GAIN_RIGHT 0xc8 +#define PM860X_HIFIR_GAIN_LEFT 0xc9 +#define PM860X_HIFIR_GAIN_RIGHT 0xca +#define PM860X_DAC_OFFSET 0xcb +#define PM860X_OFFSET_LEFT_1 0xcc +#define PM860X_OFFSET_LEFT_2 0xcd +#define PM860X_OFFSET_RIGHT_1 0xce +#define PM860X_OFFSET_RIGHT_2 0xcf +#define PM860X_ADC_ANA_1 0xd0 +#define PM860X_ADC_ANA_2 0xd1 +#define PM860X_ADC_ANA_3 0xd2 +#define PM860X_ADC_ANA_4 0xd3 +#define PM860X_ANA_TO_ANA 0xd4 +#define PM860X_HS1_CTRL 0xd5 +#define PM860X_HS2_CTRL 0xd6 +#define PM860X_LO1_CTRL 0xd7 +#define PM860X_LO2_CTRL 0xd8 +#define PM860X_EAR_CTRL_1 0xd9 +#define PM860X_EAR_CTRL_2 0xda +#define PM860X_AUDIO_SUPPLIES_1 0xdb +#define PM860X_AUDIO_SUPPLIES_2 0xdc +#define PM860X_ADC_EN_1 0xdd +#define PM860X_ADC_EN_2 0xde +#define PM860X_DAC_EN_1 0xdf +#define PM860X_DAC_EN_2 0xe1 +#define PM860X_AUDIO_CAL_1 0xe2 +#define PM860X_AUDIO_CAL_2 0xe3 +#define PM860X_AUDIO_CAL_3 0xe4 +#define PM860X_AUDIO_CAL_4 0xe5 +#define PM860X_AUDIO_CAL_5 0xe6 +#define PM860X_ANA_INPUT_SEL_1 0xe7 +#define PM860X_ANA_INPUT_SEL_2 0xe8 -#define PM860X_PCM_IFACE_4 0x39 -#define PM860X_I2S_IFACE_5 0x3a +#define PM860X_PCM_IFACE_4 0xe9 +#define PM860X_I2S_IFACE_5 0xea #define PM860X_SHORTS 0x3b #define PM860X_PLL_ADJ_1 0x3c diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 80555d7551e..21ae8d4fdbf 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -126,6 +126,8 @@ struct ab8500_codec_drvdata_dbg { /* Private data for AB8500 device-driver */ struct ab8500_codec_drvdata { + struct regmap *regmap; + /* Sidetone */ long *sid_fir_values; enum sid_state sid_status; @@ -166,49 +168,35 @@ static inline const char *amic_type_str(enum amic_type type) */ /* Read a register from the audio-bank of AB8500 */ -static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec, - unsigned int reg) +static int ab8500_codec_read_reg(void *context, unsigned int reg, + unsigned int *value) { + struct device *dev = context; int status; - unsigned int value = 0; u8 value8; - status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO, - reg, &value8); - if (status < 0) { - dev_err(codec->dev, - "%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - } else { - dev_dbg(codec->dev, - "%s: Read 0x%02x from register 0x%02x:0x%02x\n", - __func__, value8, (u8)AB8500_AUDIO, (u8)reg); - value = (unsigned int)value8; - } + status = abx500_get_register_interruptible(dev, AB8500_AUDIO, + reg, &value8); + *value = (unsigned int)value8; - return value; + return status; } /* Write to a register in the audio-bank of AB8500 */ -static int ab8500_codec_write_reg(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static int ab8500_codec_write_reg(void *context, unsigned int reg, + unsigned int value) { - int status; - - status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO, - reg, value); - if (status < 0) - dev_err(codec->dev, - "%s: ERROR: Register (%02x:%02x) write failed (%d).\n", - __func__, (u8)AB8500_AUDIO, (u8)reg, status); - else - dev_dbg(codec->dev, - "%s: Wrote 0x%02x into register %02x:%02x\n", - __func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg); + struct device *dev = context; - return status; + return abx500_set_register_interruptible(dev, AB8500_AUDIO, + reg, value); } +static const struct regmap_config ab8500_codec_regmap = { + .reg_read = ab8500_codec_read_reg, + .reg_write = ab8500_codec_write_reg, +}; + /* * Controls - DAPM */ @@ -2312,17 +2300,17 @@ static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, case 0: break; case 1: - slot = find_first_bit((unsigned long *)&tx_mask, 32); + slot = ffs(tx_mask); snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, slot); snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, slot); snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, slot); snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, slot); break; case 2: - slot = find_first_bit((unsigned long *)&tx_mask, 32); + slot = ffs(tx_mask); snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, slot); snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, slot); - slot = find_next_bit((unsigned long *)&tx_mask, 32, slot + 1); + slot = fls(tx_mask); snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, slot); snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, slot); break; @@ -2353,18 +2341,18 @@ static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, case 0: break; case 1: - slot = find_first_bit((unsigned long *)&rx_mask, 32); + slot = ffs(rx_mask); snd_soc_update_bits(codec, AB8500_ADSLOTSEL(slot), AB8500_MASK_SLOT(slot), AB8500_ADSLOTSELX_AD_OUT_TO_SLOT(AB8500_AD_OUT3, slot)); break; case 2: - slot = find_first_bit((unsigned long *)&rx_mask, 32); + slot = ffs(rx_mask); snd_soc_update_bits(codec, AB8500_ADSLOTSEL(slot), AB8500_MASK_SLOT(slot), AB8500_ADSLOTSELX_AD_OUT_TO_SLOT(AB8500_AD_OUT3, slot)); - slot = find_next_bit((unsigned long *)&rx_mask, 32, slot + 1); + slot = fls(rx_mask); snd_soc_update_bits(codec, AB8500_ADSLOTSEL(slot), AB8500_MASK_SLOT(slot), @@ -2485,9 +2473,13 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) dev_dbg(dev, "%s: Enter.\n", __func__); + snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + /* Setup AB8500 according to board-settings */ pdata = dev_get_platdata(dev->parent); + codec->control_data = drvdata->regmap; + if (np) { if (!pdata) pdata = devm_kzalloc(dev, @@ -2532,12 +2524,10 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) } /* Override HW-defaults */ - ab8500_codec_write_reg(codec, - AB8500_ANACONF5, - BIT(AB8500_ANACONF5_HSAUTOEN)); - ab8500_codec_write_reg(codec, - AB8500_SHORTCIRCONF, - BIT(AB8500_SHORTCIRCONF_HSZCDDIS)); + snd_soc_write(codec, AB8500_ANACONF5, + BIT(AB8500_ANACONF5_HSAUTOEN)); + snd_soc_write(codec, AB8500_SHORTCIRCONF, + BIT(AB8500_SHORTCIRCONF_HSZCDDIS)); /* Add filter controls */ status = snd_soc_add_codec_controls(codec, ab8500_filter_controls, @@ -2567,9 +2557,6 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver ab8500_codec_driver = { .probe = ab8500_codec_probe, - .read = ab8500_codec_read_reg, - .write = ab8500_codec_write_reg, - .reg_word_size = sizeof(u8), .controls = ab8500_ctrls, .num_controls = ARRAY_SIZE(ab8500_ctrls), .dapm_widgets = ab8500_dapm_widgets, @@ -2588,10 +2575,21 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) /* Create driver private-data struct */ drvdata = devm_kzalloc(&pdev->dev, sizeof(struct ab8500_codec_drvdata), GFP_KERNEL); + if (!drvdata) + return -ENOMEM; drvdata->sid_status = SID_UNCONFIGURED; drvdata->anc_status = ANC_UNCONFIGURED; dev_set_drvdata(&pdev->dev, drvdata); + drvdata->regmap = devm_regmap_init(&pdev->dev, NULL, &pdev->dev, + &ab8500_codec_regmap); + if (IS_ERR(drvdata->regmap)) { + status = PTR_ERR(drvdata->regmap); + dev_err(&pdev->dev, "%s: Failed to allocate regmap: %d\n", + __func__, status); + return status; + } + dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__); status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver, ab8500_codec_dai, @@ -2606,7 +2604,7 @@ static int ab8500_codec_driver_probe(struct platform_device *pdev) static int ab8500_codec_driver_remove(struct platform_device *pdev) { - dev_info(&pdev->dev, "%s Enter.\n", __func__); + dev_dbg(&pdev->dev, "%s Enter.\n", __func__); snd_soc_unregister_codec(&pdev->dev); diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 1aa10ddf3a6..59654b1e7f3 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -32,6 +32,7 @@ struct adau1373_dai { }; struct adau1373 { + struct regmap *regmap; struct adau1373_dai dais[3]; }; @@ -73,7 +74,6 @@ struct adau1373 { #define ADAU1373_PLL_CTRL4(x) (0x2c + (x) * 7) #define ADAU1373_PLL_CTRL5(x) (0x2d + (x) * 7) #define ADAU1373_PLL_CTRL6(x) (0x2e + (x) * 7) -#define ADAU1373_PLL_CTRL7(x) (0x2f + (x) * 7) #define ADAU1373_HEADDECT 0x36 #define ADAU1373_ADC_DAC_STATUS 0x37 #define ADAU1373_ADC_CTRL 0x3c @@ -152,37 +152,172 @@ struct adau1373 { #define ADAU1373_EP_CTRL_MICBIAS1_OFFSET 4 #define ADAU1373_EP_CTRL_MICBIAS2_OFFSET 2 -static const uint8_t adau1373_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x00 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x10 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x20 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, /* 0x30 */ - 0x00, 0x00, 0x00, 0x80, 0x00, 0x01, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x0a, 0x0a, 0x0a, 0x00, /* 0x40 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x08, 0x08, 0x08, 0x00, 0x00, 0x00, 0x00, /* 0x50 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x60 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x70 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x80 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x90 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0xa0 */ - 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, - 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff, 0xff, /* 0xb0 */ - 0xff, 0xff, 0xff, 0xff, 0xff, 0x1f, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xc0 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xd0 */ - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 0xe0 */ - 0x00, 0x1f, 0x0f, 0x00, 0x00, +static const struct reg_default adau1373_reg_defaults[] = { + { ADAU1373_INPUT_MODE, 0x00 }, + { ADAU1373_AINL_CTRL(0), 0x00 }, + { ADAU1373_AINR_CTRL(0), 0x00 }, + { ADAU1373_AINL_CTRL(1), 0x00 }, + { ADAU1373_AINR_CTRL(1), 0x00 }, + { ADAU1373_AINL_CTRL(2), 0x00 }, + { ADAU1373_AINR_CTRL(2), 0x00 }, + { ADAU1373_AINL_CTRL(3), 0x00 }, + { ADAU1373_AINR_CTRL(3), 0x00 }, + { ADAU1373_LLINE_OUT(0), 0x00 }, + { ADAU1373_RLINE_OUT(0), 0x00 }, + { ADAU1373_LLINE_OUT(1), 0x00 }, + { ADAU1373_RLINE_OUT(1), 0x00 }, + { ADAU1373_LSPK_OUT, 0x00 }, + { ADAU1373_RSPK_OUT, 0x00 }, + { ADAU1373_LHP_OUT, 0x00 }, + { ADAU1373_RHP_OUT, 0x00 }, + { ADAU1373_ADC_GAIN, 0x00 }, + { ADAU1373_LADC_MIXER, 0x00 }, + { ADAU1373_RADC_MIXER, 0x00 }, + { ADAU1373_LLINE1_MIX, 0x00 }, + { ADAU1373_RLINE1_MIX, 0x00 }, + { ADAU1373_LLINE2_MIX, 0x00 }, + { ADAU1373_RLINE2_MIX, 0x00 }, + { ADAU1373_LSPK_MIX, 0x00 }, + { ADAU1373_RSPK_MIX, 0x00 }, + { ADAU1373_LHP_MIX, 0x00 }, + { ADAU1373_RHP_MIX, 0x00 }, + { ADAU1373_EP_MIX, 0x00 }, + { ADAU1373_HP_CTRL, 0x00 }, + { ADAU1373_HP_CTRL2, 0x00 }, + { ADAU1373_LS_CTRL, 0x00 }, + { ADAU1373_EP_CTRL, 0x00 }, + { ADAU1373_MICBIAS_CTRL1, 0x00 }, + { ADAU1373_MICBIAS_CTRL2, 0x00 }, + { ADAU1373_OUTPUT_CTRL, 0x00 }, + { ADAU1373_PWDN_CTRL1, 0x00 }, + { ADAU1373_PWDN_CTRL2, 0x00 }, + { ADAU1373_PWDN_CTRL3, 0x00 }, + { ADAU1373_DPLL_CTRL(0), 0x00 }, + { ADAU1373_PLL_CTRL1(0), 0x00 }, + { ADAU1373_PLL_CTRL2(0), 0x00 }, + { ADAU1373_PLL_CTRL3(0), 0x00 }, + { ADAU1373_PLL_CTRL4(0), 0x00 }, + { ADAU1373_PLL_CTRL5(0), 0x00 }, + { ADAU1373_PLL_CTRL6(0), 0x02 }, + { ADAU1373_DPLL_CTRL(1), 0x00 }, + { ADAU1373_PLL_CTRL1(1), 0x00 }, + { ADAU1373_PLL_CTRL2(1), 0x00 }, + { ADAU1373_PLL_CTRL3(1), 0x00 }, + { ADAU1373_PLL_CTRL4(1), 0x00 }, + { ADAU1373_PLL_CTRL5(1), 0x00 }, + { ADAU1373_PLL_CTRL6(1), 0x02 }, + { ADAU1373_HEADDECT, 0x00 }, + { ADAU1373_ADC_CTRL, 0x00 }, + { ADAU1373_CLK_SRC_DIV(0), 0x00 }, + { ADAU1373_CLK_SRC_DIV(1), 0x00 }, + { ADAU1373_DAI(0), 0x0a }, + { ADAU1373_DAI(1), 0x0a }, + { ADAU1373_DAI(2), 0x0a }, + { ADAU1373_BCLKDIV(0), 0x00 }, + { ADAU1373_BCLKDIV(1), 0x00 }, + { ADAU1373_BCLKDIV(2), 0x00 }, + { ADAU1373_SRC_RATIOA(0), 0x00 }, + { ADAU1373_SRC_RATIOB(0), 0x00 }, + { ADAU1373_SRC_RATIOA(1), 0x00 }, + { ADAU1373_SRC_RATIOB(1), 0x00 }, + { ADAU1373_SRC_RATIOA(2), 0x00 }, + { ADAU1373_SRC_RATIOB(2), 0x00 }, + { ADAU1373_DEEMP_CTRL, 0x00 }, + { ADAU1373_SRC_DAI_CTRL(0), 0x08 }, + { ADAU1373_SRC_DAI_CTRL(1), 0x08 }, + { ADAU1373_SRC_DAI_CTRL(2), 0x08 }, + { ADAU1373_DIN_MIX_CTRL(0), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(1), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(2), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(3), 0x00 }, + { ADAU1373_DIN_MIX_CTRL(4), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(0), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(1), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(2), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(3), 0x00 }, + { ADAU1373_DOUT_MIX_CTRL(4), 0x00 }, + { ADAU1373_DAI_PBL_VOL(0), 0x00 }, + { ADAU1373_DAI_PBR_VOL(0), 0x00 }, + { ADAU1373_DAI_PBL_VOL(1), 0x00 }, + { ADAU1373_DAI_PBR_VOL(1), 0x00 }, + { ADAU1373_DAI_PBL_VOL(2), 0x00 }, + { ADAU1373_DAI_PBR_VOL(2), 0x00 }, + { ADAU1373_DAI_RECL_VOL(0), 0x00 }, + { ADAU1373_DAI_RECR_VOL(0), 0x00 }, + { ADAU1373_DAI_RECL_VOL(1), 0x00 }, + { ADAU1373_DAI_RECR_VOL(1), 0x00 }, + { ADAU1373_DAI_RECL_VOL(2), 0x00 }, + { ADAU1373_DAI_RECR_VOL(2), 0x00 }, + { ADAU1373_DAC1_PBL_VOL, 0x00 }, + { ADAU1373_DAC1_PBR_VOL, 0x00 }, + { ADAU1373_DAC2_PBL_VOL, 0x00 }, + { ADAU1373_DAC2_PBR_VOL, 0x00 }, + { ADAU1373_ADC_RECL_VOL, 0x00 }, + { ADAU1373_ADC_RECR_VOL, 0x00 }, + { ADAU1373_DMIC_RECL_VOL, 0x00 }, + { ADAU1373_DMIC_RECR_VOL, 0x00 }, + { ADAU1373_VOL_GAIN1, 0x00 }, + { ADAU1373_VOL_GAIN2, 0x00 }, + { ADAU1373_VOL_GAIN3, 0x00 }, + { ADAU1373_HPF_CTRL, 0x00 }, + { ADAU1373_BASS1, 0x00 }, + { ADAU1373_BASS2, 0x00 }, + { ADAU1373_DRC(0) + 0x0, 0x78 }, + { ADAU1373_DRC(0) + 0x1, 0x18 }, + { ADAU1373_DRC(0) + 0x2, 0x00 }, + { ADAU1373_DRC(0) + 0x3, 0x00 }, + { ADAU1373_DRC(0) + 0x4, 0x00 }, + { ADAU1373_DRC(0) + 0x5, 0xc0 }, + { ADAU1373_DRC(0) + 0x6, 0x00 }, + { ADAU1373_DRC(0) + 0x7, 0x00 }, + { ADAU1373_DRC(0) + 0x8, 0x00 }, + { ADAU1373_DRC(0) + 0x9, 0xc0 }, + { ADAU1373_DRC(0) + 0xa, 0x88 }, + { ADAU1373_DRC(0) + 0xb, 0x7a }, + { ADAU1373_DRC(0) + 0xc, 0xdf }, + { ADAU1373_DRC(0) + 0xd, 0x20 }, + { ADAU1373_DRC(0) + 0xe, 0x00 }, + { ADAU1373_DRC(0) + 0xf, 0x00 }, + { ADAU1373_DRC(1) + 0x0, 0x78 }, + { ADAU1373_DRC(1) + 0x1, 0x18 }, + { ADAU1373_DRC(1) + 0x2, 0x00 }, + { ADAU1373_DRC(1) + 0x3, 0x00 }, + { ADAU1373_DRC(1) + 0x4, 0x00 }, + { ADAU1373_DRC(1) + 0x5, 0xc0 }, + { ADAU1373_DRC(1) + 0x6, 0x00 }, + { ADAU1373_DRC(1) + 0x7, 0x00 }, + { ADAU1373_DRC(1) + 0x8, 0x00 }, + { ADAU1373_DRC(1) + 0x9, 0xc0 }, + { ADAU1373_DRC(1) + 0xa, 0x88 }, + { ADAU1373_DRC(1) + 0xb, 0x7a }, + { ADAU1373_DRC(1) + 0xc, 0xdf }, + { ADAU1373_DRC(1) + 0xd, 0x20 }, + { ADAU1373_DRC(1) + 0xe, 0x00 }, + { ADAU1373_DRC(1) + 0xf, 0x00 }, + { ADAU1373_DRC(2) + 0x0, 0x78 }, + { ADAU1373_DRC(2) + 0x1, 0x18 }, + { ADAU1373_DRC(2) + 0x2, 0x00 }, + { ADAU1373_DRC(2) + 0x3, 0x00 }, + { ADAU1373_DRC(2) + 0x4, 0x00 }, + { ADAU1373_DRC(2) + 0x5, 0xc0 }, + { ADAU1373_DRC(2) + 0x6, 0x00 }, + { ADAU1373_DRC(2) + 0x7, 0x00 }, + { ADAU1373_DRC(2) + 0x8, 0x00 }, + { ADAU1373_DRC(2) + 0x9, 0xc0 }, + { ADAU1373_DRC(2) + 0xa, 0x88 }, + { ADAU1373_DRC(2) + 0xb, 0x7a }, + { ADAU1373_DRC(2) + 0xc, 0xdf }, + { ADAU1373_DRC(2) + 0xd, 0x20 }, + { ADAU1373_DRC(2) + 0xe, 0x00 }, + { ADAU1373_DRC(2) + 0xf, 0x00 }, + { ADAU1373_3D_CTRL1, 0x00 }, + { ADAU1373_3D_CTRL2, 0x00 }, + { ADAU1373_FDSP_SEL1, 0x00 }, + { ADAU1373_FDSP_SEL2, 0x00 }, + { ADAU1373_FDSP_SEL2, 0x00 }, + { ADAU1373_FDSP_SEL4, 0x00 }, + { ADAU1373_DIGMICCTRL, 0x00 }, + { ADAU1373_DIGEN, 0x00 }, }; static const unsigned int adau1373_out_tlv[] = { @@ -418,6 +553,7 @@ static int adau1373_pll_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); unsigned int pll_id = w->name[3] - '1'; unsigned int val; @@ -426,7 +562,7 @@ static int adau1373_pll_event(struct snd_soc_dapm_widget *w, else val = 0; - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_PLL_EN, val); if (SND_SOC_DAPM_EVENT_ON(event)) @@ -938,7 +1074,7 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, adau1373_dai->enable_src = (div != 0); - snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_BCLKDIV(dai->id), ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK, (div << 2) | ADAU1373_BCLKDIV_64); @@ -959,7 +1095,7 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + return regmap_update_bits(adau1373->regmap, ADAU1373_DAI(dai->id), ADAU1373_DAI_WLEN_MASK, ctrl); } @@ -1016,7 +1152,7 @@ static int adau1373_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_DAI(dai->id), ~ADAU1373_DAI_WLEN_MASK, ctrl); return 0; @@ -1039,7 +1175,7 @@ static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai, adau1373_dai->sysclk = freq; adau1373_dai->clk_src = clk_id; - snd_soc_update_bits(dai->codec, ADAU1373_BCLKDIV(dai->id), + regmap_update_bits(adau1373->regmap, ADAU1373_BCLKDIV(dai->id), ADAU1373_BCLKDIV_SOURCE, clk_id << 5); return 0; @@ -1120,6 +1256,7 @@ static struct snd_soc_dai_driver adau1373_dai_driver[] = { static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); unsigned int dpll_div = 0; unsigned int x, r, n, m, i, j, mode; @@ -1187,36 +1324,36 @@ static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, if (dpll_div) { dpll_div = 11 - dpll_div; - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_DPLL_BYPASS, 0); } else { - snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + regmap_update_bits(adau1373->regmap, ADAU1373_PLL_CTRL6(pll_id), ADAU1373_PLL_CTRL6_DPLL_BYPASS, ADAU1373_PLL_CTRL6_DPLL_BYPASS); } - snd_soc_write(codec, ADAU1373_DPLL_CTRL(pll_id), + regmap_write(adau1373->regmap, ADAU1373_DPLL_CTRL(pll_id), (source << 4) | dpll_div); - snd_soc_write(codec, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); - snd_soc_write(codec, ADAU1373_PLL_CTRL5(pll_id), + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL5(pll_id), (r << 3) | (x << 1) | mode); /* Set sysclk to pll_rate / 4 */ - snd_soc_update_bits(codec, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); + regmap_update_bits(adau1373->regmap, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); return 0; } -static void adau1373_load_drc_settings(struct snd_soc_codec *codec, +static void adau1373_load_drc_settings(struct adau1373 *adau1373, unsigned int nr, uint8_t *drc) { unsigned int i; for (i = 0; i < ADAU1373_DRC_SIZE; ++i) - snd_soc_write(codec, ADAU1373_DRC(nr) + i, drc[i]); + regmap_write(adau1373->regmap, ADAU1373_DRC(nr) + i, drc[i]); } static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias) @@ -1235,13 +1372,14 @@ static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias) static int adau1373_probe(struct snd_soc_codec *codec) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); struct adau1373_platform_data *pdata = codec->dev->platform_data; bool lineout_differential = false; unsigned int val; int ret; int i; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; @@ -1256,7 +1394,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) return -EINVAL; for (i = 0; i < pdata->num_drc; ++i) { - adau1373_load_drc_settings(codec, i, + adau1373_load_drc_settings(adau1373, i, pdata->drc_setting[i]); } @@ -1268,18 +1406,18 @@ static int adau1373_probe(struct snd_soc_codec *codec) if (pdata->input_differential[i]) val |= BIT(i); } - snd_soc_write(codec, ADAU1373_INPUT_MODE, val); + regmap_write(adau1373->regmap, ADAU1373_INPUT_MODE, val); val = 0; if (pdata->lineout_differential) val |= ADAU1373_OUTPUT_CTRL_LDIFF; if (pdata->lineout_ground_sense) val |= ADAU1373_OUTPUT_CTRL_LNFBEN; - snd_soc_write(codec, ADAU1373_OUTPUT_CTRL, val); + regmap_write(adau1373->regmap, ADAU1373_OUTPUT_CTRL, val); lineout_differential = pdata->lineout_differential; - snd_soc_write(codec, ADAU1373_EP_CTRL, + regmap_write(adau1373->regmap, ADAU1373_EP_CTRL, (pdata->micbias1 << ADAU1373_EP_CTRL_MICBIAS1_OFFSET) | (pdata->micbias2 << ADAU1373_EP_CTRL_MICBIAS2_OFFSET)); } @@ -1289,7 +1427,7 @@ static int adau1373_probe(struct snd_soc_codec *codec) ARRAY_SIZE(adau1373_lineout2_controls)); } - snd_soc_write(codec, ADAU1373_ADC_CTRL, + regmap_write(adau1373->regmap, ADAU1373_ADC_CTRL, ADAU1373_ADC_CTRL_RESET_FORCE | ADAU1373_ADC_CTRL_PEAK_DETECT); return 0; @@ -1298,17 +1436,19 @@ static int adau1373_probe(struct snd_soc_codec *codec) static int adau1373_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + regmap_update_bits(adau1373->regmap, ADAU1373_PWDN_CTRL3, ADAU1373_PWDN_CTRL3_PWR_EN, ADAU1373_PWDN_CTRL3_PWR_EN); break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + regmap_update_bits(adau1373->regmap, ADAU1373_PWDN_CTRL3, ADAU1373_PWDN_CTRL3_PWR_EN, 0); break; } @@ -1324,17 +1464,49 @@ static int adau1373_remove(struct snd_soc_codec *codec) static int adau1373_suspend(struct snd_soc_codec *codec) { - return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(adau1373->regmap, true); + + return ret; } static int adau1373_resume(struct snd_soc_codec *codec) { + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(adau1373->regmap, false); adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_cache_sync(codec); + regcache_sync(adau1373->regmap); return 0; } +static bool adau1373_register_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADAU1373_SOFT_RESET: + case ADAU1373_ADC_DAC_STATUS: + return true; + default: + return false; + } +} + +static const struct regmap_config adau1373_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + + .volatile_reg = adau1373_register_volatile, + .max_register = ADAU1373_SOFT_RESET, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adau1373_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau1373_reg_defaults), +}; + static struct snd_soc_codec_driver adau1373_codec_driver = { .probe = adau1373_probe, .remove = adau1373_remove, @@ -1342,9 +1514,6 @@ static struct snd_soc_codec_driver adau1373_codec_driver = { .resume = adau1373_resume, .set_bias_level = adau1373_set_bias_level, .idle_bias_off = true, - .reg_cache_size = ARRAY_SIZE(adau1373_default_regs), - .reg_cache_default = adau1373_default_regs, - .reg_word_size = sizeof(uint8_t), .set_pll = adau1373_set_pll, @@ -1366,6 +1535,13 @@ static int adau1373_i2c_probe(struct i2c_client *client, if (!adau1373) return -ENOMEM; + adau1373->regmap = devm_regmap_init_i2c(client, + &adau1373_regmap_config); + if (IS_ERR(adau1373->regmap)) + return PTR_ERR(adau1373->regmap); + + regmap_write(adau1373->regmap, ADAU1373_SOFT_RESET, 0x00); + dev_set_drvdata(&client->dev, adau1373); ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver, diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 15b012d0f22..14a7c169d00 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -115,22 +115,34 @@ #define ADAV80X_PLL_OUTE_SYSCLKPD(x) BIT(2 - (x)) -static u8 adav80x_default_regs[] = { - 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00, - 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37, - 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b, - 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00, - 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee, - 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f, - 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00, - 0x00, 0x00, 0x00, 0x00, 0x52, 0x00, +static struct reg_default adav80x_reg_defaults[] = { + { ADAV80X_PLAYBACK_CTRL, 0x01 }, + { ADAV80X_AUX_IN_CTRL, 0x01 }, + { ADAV80X_REC_CTRL, 0x02 }, + { ADAV80X_AUX_OUT_CTRL, 0x01 }, + { ADAV80X_DPATH_CTRL1, 0xc0 }, + { ADAV80X_DPATH_CTRL2, 0x11 }, + { ADAV80X_DAC_CTRL1, 0x00 }, + { ADAV80X_DAC_CTRL2, 0x00 }, + { ADAV80X_DAC_CTRL3, 0x00 }, + { ADAV80X_DAC_L_VOL, 0xff }, + { ADAV80X_DAC_R_VOL, 0xff }, + { ADAV80X_PGA_L_VOL, 0x00 }, + { ADAV80X_PGA_R_VOL, 0x00 }, + { ADAV80X_ADC_CTRL1, 0x00 }, + { ADAV80X_ADC_CTRL2, 0x00 }, + { ADAV80X_ADC_L_VOL, 0xff }, + { ADAV80X_ADC_R_VOL, 0xff }, + { ADAV80X_PLL_CTRL1, 0x00 }, + { ADAV80X_PLL_CTRL2, 0x00 }, + { ADAV80X_ICLK_CTRL1, 0x00 }, + { ADAV80X_ICLK_CTRL2, 0x00 }, + { ADAV80X_PLL_CLK_SRC, 0x00 }, + { ADAV80X_PLL_OUTE, 0x00 }, }; struct adav80x { - enum snd_soc_control_type control_type; + struct regmap *regmap; enum adav80x_clk_src clk_src; unsigned int sysclk; @@ -298,7 +310,7 @@ static int adav80x_set_deemph(struct snd_soc_codec *codec) val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; } - return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + return regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL2, ADAV80X_DAC_CTRL2_DEEMPH_MASK, val); } @@ -394,10 +406,11 @@ static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][0], ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER, capture); - snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback); + regmap_write(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][1], + playback); adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK; @@ -407,6 +420,7 @@ static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) static int adav80x_set_adc_clock(struct snd_soc_codec *codec, unsigned int sample_rate) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; if (sample_rate <= 48000) @@ -414,7 +428,7 @@ static int adav80x_set_adc_clock(struct snd_soc_codec *codec, else val = ADAV80X_ADC_CTRL1_MODULATOR_64FS; - snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1, + regmap_update_bits(adav80x->regmap, ADAV80X_ADC_CTRL1, ADAV80X_ADC_CTRL1_MODULATOR_MASK, val); return 0; @@ -423,6 +437,7 @@ static int adav80x_set_adc_clock(struct snd_soc_codec *codec, static int adav80x_set_dac_clock(struct snd_soc_codec *codec, unsigned int sample_rate) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; if (sample_rate <= 48000) @@ -430,7 +445,7 @@ static int adav80x_set_dac_clock(struct snd_soc_codec *codec, else val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS; - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL2, ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK, val); @@ -440,6 +455,7 @@ static int adav80x_set_dac_clock(struct snd_soc_codec *codec, static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, struct snd_soc_dai *dai, snd_pcm_format_t format) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int val; switch (format) { @@ -459,7 +475,7 @@ static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][0], ADAV80X_CAPTURE_WORD_LEN_MASK, val); return 0; @@ -491,7 +507,7 @@ static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, return -EINVAL; } - snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1], + regmap_update_bits(adav80x->regmap, adav80x_port_ctrl_regs[dai->id][1], ADAV80X_PLAYBACK_MODE_MASK, val); return 0; @@ -554,8 +570,10 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, ADAV80X_ICLK_CTRL1_ICLK2_SRC(clk_id); iclk_ctrl2 = ADAV80X_ICLK_CTRL2_ICLK1_SRC(clk_id); - snd_soc_write(codec, ADAV80X_ICLK_CTRL1, iclk_ctrl1); - snd_soc_write(codec, ADAV80X_ICLK_CTRL2, iclk_ctrl2); + regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL1, + iclk_ctrl1); + regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL2, + iclk_ctrl2); snd_soc_dapm_sync(&codec->dapm); } @@ -575,10 +593,12 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, mask = ADAV80X_PLL_OUTE_SYSCLKPD(clk_id); if (freq == 0) { - snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, mask); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_OUTE, + mask, mask); adav80x->sysclk_pd[clk_id] = true; } else { - snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, 0); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_OUTE, + mask, 0); adav80x->sysclk_pd[clk_id] = false; } @@ -650,9 +670,9 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, return -EINVAL; } - snd_soc_update_bits(codec, ADAV80X_PLL_CTRL1, ADAV80X_PLL_CTRL1_PLLDIV, - pll_ctrl1); - snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2, + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CTRL1, + ADAV80X_PLL_CTRL1_PLLDIV, pll_ctrl1); + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CTRL2, ADAV80X_PLL_CTRL2_PLL_MASK(pll_id), pll_ctrl2); if (source != adav80x->pll_src) { @@ -661,7 +681,7 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, else pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll_id); - snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC, + regmap_update_bits(adav80x->regmap, ADAV80X_PLL_CLK_SRC, ADAV80X_PLL_CLK_SRC_PLL_MASK(pll_id), pll_src); adav80x->pll_src = source; @@ -675,6 +695,7 @@ static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, static int adav80x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); unsigned int mask = ADAV80X_DAC_CTRL1_PD; switch (level) { @@ -683,10 +704,12 @@ static int adav80x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00); + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL1, mask, + 0x00); break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask); + regmap_update_bits(adav80x->regmap, ADAV80X_DAC_CTRL1, mask, + mask); break; } @@ -780,7 +803,7 @@ static int adav80x_probe(struct snd_soc_codec *codec) int ret; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type); + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (ret) { dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); return ret; @@ -791,23 +814,31 @@ static int adav80x_probe(struct snd_soc_codec *codec) snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); /* Power down S/PDIF receiver, since it is currently not supported */ - snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x20); + regmap_write(adav80x->regmap, ADAV80X_PLL_OUTE, 0x20); /* Disable DAC zero flag */ - snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6); + regmap_write(adav80x->regmap, ADAV80X_DAC_CTRL3, 0x6); return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); } static int adav80x_suspend(struct snd_soc_codec *codec) { - return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(adav80x->regmap, true); + + return ret; } static int adav80x_resume(struct snd_soc_codec *codec) { + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(adav80x->regmap, false); adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - codec->cache_sync = 1; - snd_soc_cache_sync(codec); + regcache_sync(adav80x->regmap); return 0; } @@ -827,10 +858,6 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .set_pll = adav80x_set_pll, .set_sysclk = adav80x_set_sysclk, - .reg_word_size = sizeof(u8), - .reg_cache_size = ARRAY_SIZE(adav80x_default_regs), - .reg_cache_default = adav80x_default_regs, - .controls = adav80x_controls, .num_controls = ARRAY_SIZE(adav80x_controls), .dapm_widgets = adav80x_dapm_widgets, @@ -839,18 +866,21 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes), }; -static int adav80x_bus_probe(struct device *dev, - enum snd_soc_control_type control_type) +static int adav80x_bus_probe(struct device *dev, struct regmap *regmap) { struct adav80x *adav80x; int ret; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL); if (!adav80x) return -ENOMEM; + dev_set_drvdata(dev, adav80x); - adav80x->control_type = control_type; + adav80x->regmap = regmap; ret = snd_soc_register_codec(dev, &adav80x_codec_driver, adav80x_dais, ARRAY_SIZE(adav80x_dais)); @@ -868,6 +898,19 @@ static int adav80x_bus_remove(struct device *dev) } #if defined(CONFIG_SPI_MASTER) +static const struct regmap_config adav80x_spi_regmap_config = { + .val_bits = 8, + .pad_bits = 1, + .reg_bits = 7, + .read_flag_mask = 0x01, + + .max_register = ADAV80X_PLL_OUTE, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adav80x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults), +}; + static const struct spi_device_id adav80x_spi_id[] = { { "adav801", 0 }, { } @@ -876,7 +919,8 @@ MODULE_DEVICE_TABLE(spi, adav80x_spi_id); static int adav80x_spi_probe(struct spi_device *spi) { - return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); + return adav80x_bus_probe(&spi->dev, + devm_regmap_init_spi(spi, &adav80x_spi_regmap_config)); } static int adav80x_spi_remove(struct spi_device *spi) @@ -896,6 +940,18 @@ static struct spi_driver adav80x_spi_driver = { #endif #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct regmap_config adav80x_i2c_regmap_config = { + .val_bits = 8, + .pad_bits = 1, + .reg_bits = 7, + + .max_register = ADAV80X_PLL_OUTE, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adav80x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults), +}; + static const struct i2c_device_id adav80x_i2c_id[] = { { "adav803", 0 }, { } @@ -905,7 +961,8 @@ MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id); static int adav80x_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - return adav80x_bus_probe(&client->dev, SND_SOC_I2C); + return adav80x_bus_probe(&client->dev, + devm_regmap_init_i2c(client, &adav80x_i2c_regmap_config)); } static int adav80x_i2c_remove(struct i2c_client *client) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 71059c07ae7..b4819dcd4f4 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -45,8 +45,6 @@ #define AK4104_TX_TXE (1 << 0) #define AK4104_TX_V (1 << 1) -#define DRV_NAME "ak4104-codec" - struct ak4104_private { struct regmap *regmap; }; @@ -291,12 +289,19 @@ static const struct of_device_id ak4104_of_match[] = { }; MODULE_DEVICE_TABLE(of, ak4104_of_match); +static const struct spi_device_id ak4104_id_table[] = { + { "ak4104", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, ak4104_id_table); + static struct spi_driver ak4104_spi_driver = { .driver = { - .name = DRV_NAME, + .name = "ak4104", .owner = THIS_MODULE, .of_match_table = ak4104_of_match, }, + .id_table = ak4104_id_table, .probe = ak4104_spi_probe, .remove = ak4104_spi_remove, }; diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2d037870970..090d499bb7e 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -257,7 +257,7 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ - snd_soc_write(codec, SG_SL1, PMMP | MGAIN0); + snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0); snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL); @@ -352,7 +352,6 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) */ default: return -EINVAL; - break; } snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data); @@ -405,7 +404,6 @@ static int ak4642_dai_hw_params(struct snd_pcm_substream *substream, break; default: return -EINVAL; - break; } snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 657808ba141..6f05b17d196 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1477,21 +1477,25 @@ static void arizona_enable_fll(struct arizona_fll *fll, { struct arizona *arizona = fll->arizona; int ret; + bool use_sync = false; /* * If we have both REFCLK and SYNCCLK then enable both, * otherwise apply the SYNCCLK settings to REFCLK. */ - if (fll->ref_src >= 0 && fll->ref_src != fll->sync_src) { + if (fll->ref_src >= 0 && fll->ref_freq && + fll->ref_src != fll->sync_src) { regmap_update_bits(arizona->regmap, fll->base + 5, ARIZONA_FLL1_OUTDIV_MASK, ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, false); - if (fll->sync_src >= 0) + if (fll->sync_src >= 0) { arizona_apply_fll(arizona, fll->base + 0x10, sync, fll->sync_src, true); + use_sync = true; + } } else if (fll->sync_src >= 0) { regmap_update_bits(arizona->regmap, fll->base + 5, ARIZONA_FLL1_OUTDIV_MASK, @@ -1511,7 +1515,7 @@ static void arizona_enable_fll(struct arizona_fll *fll, * Increase the bandwidth if we're not using a low frequency * sync source. */ - if (fll->sync_src >= 0 && fll->sync_freq > 100000) + if (use_sync && fll->sync_freq > 100000) regmap_update_bits(arizona->regmap, fll->base + 0x17, ARIZONA_FLL1_SYNC_BW, 0); else @@ -1526,8 +1530,7 @@ static void arizona_enable_fll(struct arizona_fll *fll, regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); - if (fll->ref_src >= 0 && fll->sync_src >= 0 && - fll->ref_src != fll->sync_src) + if (use_sync) regmap_update_bits(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, ARIZONA_FLL1_SYNC_ENA); @@ -1561,10 +1564,12 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (fll->fout && Fref > 0) { - ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); - if (ret != 0) - return ret; + if (fll->fout) { + if (Fref > 0) { + ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); + if (ret != 0) + return ret; + } if (fll->sync_src >= 0) { ret = arizona_calc_fll(fll, &sync, fll->sync_freq, diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 23316c887b1..43737a27d79 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -38,24 +38,6 @@ #include <sound/soc.h> #include <sound/initval.h> -static inline unsigned int cq93vc_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct davinci_vc *davinci_vc = codec->control_data; - - return readl(davinci_vc->base + reg); -} - -static inline int cq93vc_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - struct davinci_vc *davinci_vc = codec->control_data; - - writel(value, davinci_vc->base + reg); - - return 0; -} - static const struct snd_kcontrol_new cq93vc_snd_controls[] = { SOC_SINGLE("PGA Capture Volume", DAVINCI_VC_REG05, 0, 0x03, 0), SOC_SINGLE("Mono DAC Playback Volume", DAVINCI_VC_REG09, 0, 0x3f, 0), @@ -64,13 +46,15 @@ static const struct snd_kcontrol_new cq93vc_snd_controls[] = { static int cq93vc_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; - u8 reg = cq93vc_read(codec, DAVINCI_VC_REG09) & ~DAVINCI_VC_REG09_MUTE; + u8 reg; if (mute) - cq93vc_write(codec, DAVINCI_VC_REG09, - reg | DAVINCI_VC_REG09_MUTE); + reg = DAVINCI_VC_REG09_MUTE; else - cq93vc_write(codec, DAVINCI_VC_REG09, reg); + reg = 0; + + snd_soc_update_bits(codec, DAVINCI_VC_REG09, DAVINCI_VC_REG09_MUTE, + reg); return 0; } @@ -79,7 +63,7 @@ static int cq93vc_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = codec_dai->codec; - struct davinci_vc *davinci_vc = codec->control_data; + struct davinci_vc *davinci_vc = codec->dev->platform_data; switch (freq) { case 22579200: @@ -97,18 +81,18 @@ static int cq93vc_set_bias_level(struct snd_soc_codec *codec, { switch (level) { case SND_SOC_BIAS_ON: - cq93vc_write(codec, DAVINCI_VC_REG12, + snd_soc_write(codec, DAVINCI_VC_REG12, DAVINCI_VC_REG12_POWER_ALL_ON); break; case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - cq93vc_write(codec, DAVINCI_VC_REG12, + snd_soc_write(codec, DAVINCI_VC_REG12, DAVINCI_VC_REG12_POWER_ALL_OFF); break; case SND_SOC_BIAS_OFF: /* force all power off */ - cq93vc_write(codec, DAVINCI_VC_REG12, + snd_soc_write(codec, DAVINCI_VC_REG12, DAVINCI_VC_REG12_POWER_ALL_OFF); break; } @@ -154,11 +138,9 @@ static int cq93vc_probe(struct snd_soc_codec *codec) struct davinci_vc *davinci_vc = codec->dev->platform_data; davinci_vc->cq93vc.codec = codec; - codec->control_data = davinci_vc; + codec->control_data = davinci_vc->regmap; - /* Set controls */ - snd_soc_add_codec_controls(codec, cq93vc_snd_controls, - ARRAY_SIZE(cq93vc_snd_controls)); + snd_soc_codec_set_cache_io(codec, 32, 32, SND_SOC_REGMAP); /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -174,12 +156,12 @@ static int cq93vc_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_cq93vc = { - .read = cq93vc_read, - .write = cq93vc_write, .set_bias_level = cq93vc_set_bias_level, .probe = cq93vc_probe, .remove = cq93vc_remove, .resume = cq93vc_resume, + .controls = cq93vc_snd_controls, + .num_controls = ARRAY_SIZE(cq93vc_snd_controls), }; static int cq93vc_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index a20f1bb8f07..f6e953454bc 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -25,6 +25,7 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/spi/spi.h> +#include <linux/of.h> #include <linux/of_device.h> #include <linux/of_gpio.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index be2ba1b6fe4..8b427c97708 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -17,6 +17,7 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/gpio.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/input.h> @@ -1116,40 +1117,6 @@ static int cs42l52_probe(struct snd_soc_codec *codec) cs42l52->sysclk = CS42L52_DEFAULT_CLK; cs42l52->config.format = CS42L52_DEFAULT_FORMAT; - /* Set Platform MICx CFG */ - snd_soc_update_bits(codec, CS42L52_MICA_CTL, - CS42L52_MIC_CTL_TYPE_MASK, - cs42l52->pdata.mica_cfg << - CS42L52_MIC_CTL_TYPE_SHIFT); - - snd_soc_update_bits(codec, CS42L52_MICB_CTL, - CS42L52_MIC_CTL_TYPE_MASK, - cs42l52->pdata.micb_cfg << - CS42L52_MIC_CTL_TYPE_SHIFT); - - /* if Single Ended, Get Mic_Select */ - if (cs42l52->pdata.mica_cfg) - snd_soc_update_bits(codec, CS42L52_MICA_CTL, - CS42L52_MIC_CTL_MIC_SEL_MASK, - cs42l52->pdata.mica_sel << - CS42L52_MIC_CTL_MIC_SEL_SHIFT); - if (cs42l52->pdata.micb_cfg) - snd_soc_update_bits(codec, CS42L52_MICB_CTL, - CS42L52_MIC_CTL_MIC_SEL_MASK, - cs42l52->pdata.micb_sel << - CS42L52_MIC_CTL_MIC_SEL_SHIFT); - - /* Set Platform Charge Pump Freq */ - snd_soc_update_bits(codec, CS42L52_CHARGE_PUMP, - CS42L52_CHARGE_PUMP_MASK, - cs42l52->pdata.chgfreq << - CS42L52_CHARGE_PUMP_SHIFT); - - /* Set Platform Bias Level */ - snd_soc_update_bits(codec, CS42L52_IFACE_CTL2, - CS42L52_IFACE_CTL2_BIAS_LVL, - cs42l52->pdata.micbias_lvl); - return ret; } @@ -1205,6 +1172,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs42l52_private *cs42l52; + struct cs42l52_platform_data *pdata = dev_get_platdata(&i2c_client->dev); int ret; unsigned int devid = 0; unsigned int reg; @@ -1222,11 +1190,22 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, return ret; } - i2c_set_clientdata(i2c_client, cs42l52); + if (pdata) + cs42l52->pdata = *pdata; + + if (cs42l52->pdata.reset_gpio) { + ret = gpio_request_one(cs42l52->pdata.reset_gpio, + GPIOF_OUT_INIT_HIGH, "CS42L52 /RST"); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n", + cs42l52->pdata.reset_gpio, ret); + return ret; + } + gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 0); + gpio_set_value_cansleep(cs42l52->pdata.reset_gpio, 1); + } - if (dev_get_platdata(&i2c_client->dev)) - memcpy(&cs42l52->pdata, dev_get_platdata(&i2c_client->dev), - sizeof(cs42l52->pdata)); + i2c_set_clientdata(i2c_client, cs42l52); ret = regmap_register_patch(cs42l52->regmap, cs42l52_threshold_patch, ARRAY_SIZE(cs42l52_threshold_patch)); @@ -1244,7 +1223,43 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, return ret; } - regcache_cache_only(cs42l52->regmap, true); + dev_info(&i2c_client->dev, "Cirrus Logic CS42L52, Revision: %02X\n", + reg & 0xFF); + + /* Set Platform Data */ + if (cs42l52->pdata.mica_cfg) + regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL, + CS42L52_MIC_CTL_TYPE_MASK, + cs42l52->pdata.mica_cfg << + CS42L52_MIC_CTL_TYPE_SHIFT); + + if (cs42l52->pdata.micb_cfg) + regmap_update_bits(cs42l52->regmap, CS42L52_MICB_CTL, + CS42L52_MIC_CTL_TYPE_MASK, + cs42l52->pdata.micb_cfg << + CS42L52_MIC_CTL_TYPE_SHIFT); + + if (cs42l52->pdata.mica_sel) + regmap_update_bits(cs42l52->regmap, CS42L52_MICA_CTL, + CS42L52_MIC_CTL_MIC_SEL_MASK, + cs42l52->pdata.mica_sel << + CS42L52_MIC_CTL_MIC_SEL_SHIFT); + if (cs42l52->pdata.micb_sel) + regmap_update_bits(cs42l52->regmap, CS42L52_MICB_CTL, + CS42L52_MIC_CTL_MIC_SEL_MASK, + cs42l52->pdata.micb_sel << + CS42L52_MIC_CTL_MIC_SEL_SHIFT); + + if (cs42l52->pdata.chgfreq) + regmap_update_bits(cs42l52->regmap, CS42L52_CHARGE_PUMP, + CS42L52_CHARGE_PUMP_MASK, + cs42l52->pdata.chgfreq << + CS42L52_CHARGE_PUMP_SHIFT); + + if (cs42l52->pdata.micbias_lvl) + regmap_update_bits(cs42l52->regmap, CS42L52_IFACE_CTL2, + CS42L52_IFACE_CTL2_BIAS_LVL, + cs42l52->pdata.micbias_lvl); ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_cs42l52, &cs42l52_dai, 1); diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 4277012c471..1a9412d86d1 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -269,6 +269,6 @@ #define CS42L52_FIX_BITS1 0x3E #define CS42L52_FIX_BITS2 0x47 -#define CS42L52_MAX_REGISTER 0x34 +#define CS42L52_MAX_REGISTER 0x47 #endif diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 3b20c86cdb0..549d5d6a3fe 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -17,6 +17,7 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/of_gpio.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/regmap.h> @@ -28,6 +29,7 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> #include <sound/tlv.h> +#include <sound/cs42l73.h> #include "cs42l73.h" struct sp_config { @@ -35,6 +37,7 @@ struct sp_config { u32 srate; }; struct cs42l73_private { + struct cs42l73_platform_data pdata; struct sp_config config[3]; struct regmap *regmap; u32 sysclk; @@ -310,15 +313,6 @@ static const struct soc_enum ng_delay_enum = SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); -static const char * const charge_pump_freq_text[] = { - "0", "1", "2", "3", "4", - "5", "6", "7", "8", "9", - "10", "11", "12", "13", "14", "15" }; - -static const struct soc_enum charge_pump_enum = - SOC_ENUM_SINGLE(CS42L73_CPFCHC, 4, - ARRAY_SIZE(charge_pump_freq_text), charge_pump_freq_text); - static const char * const cs42l73_mono_mix_texts[] = { "Left", "Right", "Mono Mix"}; @@ -511,8 +505,6 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0), SOC_ENUM("NG Delay", ng_delay_enum), - SOC_ENUM("Charge Pump Frequency", charge_pump_enum), - SOC_DOUBLE_R_TLV("XSP-IP Volume", CS42L73_XSPAIPAA, CS42L73_XSPBIPBA, 0, 0x3F, 1, attn_tlv), @@ -1055,11 +1047,11 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - mmcc |= MS_MASTER; + mmcc |= CS42L73_MS_MASTER; break; case SND_SOC_DAIFMT_CBS_CFS: - mmcc &= ~MS_MASTER; + mmcc &= ~CS42L73_MS_MASTER; break; default: @@ -1071,11 +1063,11 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) switch (format) { case SND_SOC_DAIFMT_I2S: - spc &= ~SPDIF_PCM; + spc &= ~CS42L73_SPDIF_PCM; break; case SND_SOC_DAIFMT_DSP_A: case SND_SOC_DAIFMT_DSP_B: - if (mmcc & MS_MASTER) { + if (mmcc & CS42L73_MS_MASTER) { dev_err(codec->dev, "PCM format in slave mode only\n"); return -EINVAL; @@ -1085,25 +1077,25 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) "PCM format is not supported on ASP port\n"); return -EINVAL; } - spc |= SPDIF_PCM; + spc |= CS42L73_SPDIF_PCM; break; default: return -EINVAL; } - if (spc & SPDIF_PCM) { + if (spc & CS42L73_SPDIF_PCM) { /* Clear PCM mode, clear PCM_BIT_ORDER bit for MSB->LSB */ - spc &= ~(PCM_MODE_MASK | PCM_BIT_ORDER); + spc &= ~(CS42L73_PCM_MODE_MASK | CS42L73_PCM_BIT_ORDER); switch (format) { case SND_SOC_DAIFMT_DSP_B: if (inv == SND_SOC_DAIFMT_IB_IF) - spc |= PCM_MODE0; + spc |= CS42L73_PCM_MODE0; if (inv == SND_SOC_DAIFMT_IB_NF) - spc |= PCM_MODE1; + spc |= CS42L73_PCM_MODE1; break; case SND_SOC_DAIFMT_DSP_A: if (inv == SND_SOC_DAIFMT_IB_IF) - spc |= PCM_MODE1; + spc |= CS42L73_PCM_MODE1; break; default: return -EINVAL; @@ -1163,7 +1155,7 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, int mclk_coeff; int srate = params_rate(params); - if (priv->config[id].mmcc & MS_MASTER) { + if (priv->config[id].mmcc & CS42L73_MS_MASTER) { /* CS42L73 Master */ /* MCLK -> srate */ mclk_coeff = @@ -1182,13 +1174,13 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, priv->config[id].spc &= 0xFC; /* Use SCLK=64*Fs if internal MCLK >= 6.4MHz */ if (priv->mclk >= 6400000) - priv->config[id].spc |= MCK_SCLK_64FS; + priv->config[id].spc |= CS42L73_MCK_SCLK_64FS; else - priv->config[id].spc |= MCK_SCLK_MCLK; + priv->config[id].spc |= CS42L73_MCK_SCLK_MCLK; } else { /* CS42L73 Slave */ priv->config[id].spc &= 0xFC; - priv->config[id].spc |= MCK_SCLK_64FS; + priv->config[id].spc |= CS42L73_MCK_SCLK_64FS; } /* Update ASRCs */ priv->config[id].srate = srate; @@ -1208,8 +1200,8 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 0); - snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 0); + snd_soc_update_bits(codec, CS42L73_DMMCC, CS42L73_MCLKDIS, 0); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, CS42L73_PDN, 0); break; case SND_SOC_BIAS_PREPARE: @@ -1220,11 +1212,11 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec, regcache_cache_only(cs42l73->regmap, false); regcache_sync(cs42l73->regmap); } - snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, CS42L73_PDN, 1); break; case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + snd_soc_update_bits(codec, CS42L73_PWRCTL1, CS42L73_PDN, 1); if (cs42l73->shutdwn_delay > 0) { mdelay(cs42l73->shutdwn_delay); cs42l73->shutdwn_delay = 0; @@ -1233,7 +1225,7 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec, * down. */ } - snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1); + snd_soc_update_bits(codec, CS42L73_DMMCC, CS42L73_MCLKDIS, 1); break; } codec->dapm.bias_level = level; @@ -1367,11 +1359,16 @@ static int cs42l73_probe(struct snd_soc_codec *codec) return ret; } - regcache_cache_only(cs42l73->regmap, true); - cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - cs42l73->mclksel = CS42L73_CLKID_MCLK1; /* MCLK1 as master clk */ + /* Set Charge Pump Frequency */ + if (cs42l73->pdata.chgfreq) + snd_soc_update_bits(codec, CS42L73_CPFCHC, + CS42L73_CHARGEPUMP_MASK, + cs42l73->pdata.chgfreq << 4); + + /* MCLK1 as master clk */ + cs42l73->mclksel = CS42L73_CLKID_MCLK1; cs42l73->mclk = 0; return ret; @@ -1415,9 +1412,11 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs42l73_private *cs42l73; + struct cs42l73_platform_data *pdata = dev_get_platdata(&i2c_client->dev); int ret; unsigned int devid = 0; unsigned int reg; + u32 val32; cs42l73 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l73_private), GFP_KERNEL); @@ -1426,14 +1425,49 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, return -ENOMEM; } - i2c_set_clientdata(i2c_client, cs42l73); - cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap); if (IS_ERR(cs42l73->regmap)) { ret = PTR_ERR(cs42l73->regmap); dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); return ret; } + + if (pdata) { + cs42l73->pdata = *pdata; + } else { + pdata = devm_kzalloc(&i2c_client->dev, + sizeof(struct cs42l73_platform_data), + GFP_KERNEL); + if (!pdata) { + dev_err(&i2c_client->dev, "could not allocate pdata\n"); + return -ENOMEM; + } + if (i2c_client->dev.of_node) { + if (of_property_read_u32(i2c_client->dev.of_node, + "chgfreq", &val32) >= 0) + pdata->chgfreq = val32; + } + pdata->reset_gpio = of_get_named_gpio(i2c_client->dev.of_node, + "reset-gpio", 0); + cs42l73->pdata = *pdata; + } + + i2c_set_clientdata(i2c_client, cs42l73); + + if (cs42l73->pdata.reset_gpio) { + ret = gpio_request_one(cs42l73->pdata.reset_gpio, + GPIOF_OUT_INIT_HIGH, "CS42L73 /RST"); + if (ret < 0) { + dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n", + cs42l73->pdata.reset_gpio, ret); + return ret; + } + gpio_set_value_cansleep(cs42l73->pdata.reset_gpio, 0); + gpio_set_value_cansleep(cs42l73->pdata.reset_gpio, 1); + } + + regcache_cache_bypass(cs42l73->regmap, true); + /* initialize codec */ ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, ®); devid = (reg & 0xFF) << 12; @@ -1444,7 +1478,6 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_E, ®); devid |= (reg & 0xF0) >> 4; - if (devid != CS42L73_DEVID) { ret = -ENODEV; dev_err(&i2c_client->dev, @@ -1462,7 +1495,7 @@ static int cs42l73_i2c_probe(struct i2c_client *i2c_client, dev_info(&i2c_client->dev, "Cirrus Logic CS42L73, Revision: %02X\n", reg & 0xFF); - regcache_cache_only(cs42l73->regmap, true); + regcache_cache_bypass(cs42l73->regmap, false); ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_cs42l73, cs42l73_dai, @@ -1478,6 +1511,12 @@ static int cs42l73_i2c_remove(struct i2c_client *client) return 0; } +static const struct of_device_id cs42l73_of_match[] = { + { .compatible = "cirrus,cs42l73", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs42l73_of_match); + static const struct i2c_device_id cs42l73_id[] = { {"cs42l73", 0}, {} @@ -1489,6 +1528,7 @@ static struct i2c_driver cs42l73_i2c_driver = { .driver = { .name = "cs42l73", .owner = THIS_MODULE, + .of_match_table = cs42l73_of_match, }, .id_table = cs42l73_id, .probe = cs42l73_i2c_probe, diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h index f30a4c4d62e..45746186a67 100644 --- a/sound/soc/codecs/cs42l73.h +++ b/sound/soc/codecs/cs42l73.h @@ -128,59 +128,60 @@ /* Bitfield Definitions */ /* CS42L73_PWRCTL1 */ -#define PDN_ADCB (1 << 7) -#define PDN_DMICB (1 << 6) -#define PDN_ADCA (1 << 5) -#define PDN_DMICA (1 << 4) -#define PDN_LDO (1 << 2) -#define DISCHG_FILT (1 << 1) -#define PDN (1 << 0) +#define CS42L73_PDN_ADCB (1 << 7) +#define CS42L73_PDN_DMICB (1 << 6) +#define CS42L73_PDN_ADCA (1 << 5) +#define CS42L73_PDN_DMICA (1 << 4) +#define CS42L73_PDN_LDO (1 << 2) +#define CS42L73_DISCHG_FILT (1 << 1) +#define CS42L73_PDN (1 << 0) /* CS42L73_PWRCTL2 */ -#define PDN_MIC2_BIAS (1 << 7) -#define PDN_MIC1_BIAS (1 << 6) -#define PDN_VSP (1 << 4) -#define PDN_ASP_SDOUT (1 << 3) -#define PDN_ASP_SDIN (1 << 2) -#define PDN_XSP_SDOUT (1 << 1) -#define PDN_XSP_SDIN (1 << 0) +#define CS42L73_PDN_MIC2_BIAS (1 << 7) +#define CS42L73_PDN_MIC1_BIAS (1 << 6) +#define CS42L73_PDN_VSP (1 << 4) +#define CS42L73_PDN_ASP_SDOUT (1 << 3) +#define CS42L73_PDN_ASP_SDIN (1 << 2) +#define CS42L73_PDN_XSP_SDOUT (1 << 1) +#define CS42L73_PDN_XSP_SDIN (1 << 0) /* CS42L73_PWRCTL3 */ -#define PDN_THMS (1 << 5) -#define PDN_SPKLO (1 << 4) -#define PDN_EAR (1 << 3) -#define PDN_SPK (1 << 2) -#define PDN_LO (1 << 1) -#define PDN_HP (1 << 0) +#define CS42L73_PDN_THMS (1 << 5) +#define CS42L73_PDN_SPKLO (1 << 4) +#define CS42L73_PDN_EAR (1 << 3) +#define CS42L73_PDN_SPK (1 << 2) +#define CS42L73_PDN_LO (1 << 1) +#define CS42L73_PDN_HP (1 << 0) /* Thermal Overload Detect. Requires interrupt ... */ -#define THMOVLD_150C 0 -#define THMOVLD_132C 1 -#define THMOVLD_115C 2 -#define THMOVLD_098C 3 +#define CS42L73_THMOVLD_150C 0 +#define CS42L73_THMOVLD_132C 1 +#define CS42L73_THMOVLD_115C 2 +#define CS42L73_THMOVLD_098C 3 +#define CS42L73_CHARGEPUMP_MASK (0xF0) /* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ -#define SP_3ST (1 << 7) -#define SPDIF_I2S (0 << 6) -#define SPDIF_PCM (1 << 6) -#define PCM_MODE0 (0 << 4) -#define PCM_MODE1 (1 << 4) -#define PCM_MODE2 (2 << 4) -#define PCM_MODE_MASK (3 << 4) -#define PCM_BIT_ORDER (1 << 3) -#define MCK_SCLK_64FS (0 << 0) -#define MCK_SCLK_MCLK (2 << 0) -#define MCK_SCLK_PREMCLK (3 << 0) +#define CS42L73_SP_3ST (1 << 7) +#define CS42L73_SPDIF_I2S (0 << 6) +#define CS42L73_SPDIF_PCM (1 << 6) +#define CS42L73_PCM_MODE0 (0 << 4) +#define CS42L73_PCM_MODE1 (1 << 4) +#define CS42L73_PCM_MODE2 (2 << 4) +#define CS42L73_PCM_MODE_MASK (3 << 4) +#define CS42L73_PCM_BIT_ORDER (1 << 3) +#define CS42L73_MCK_SCLK_64FS (0 << 0) +#define CS42L73_MCK_SCLK_MCLK (2 << 0) +#define CS42L73_MCK_SCLK_PREMCLK (3 << 0) /* CS42L73_xSPMMCC */ -#define MS_MASTER (1 << 7) +#define CS42L73_MS_MASTER (1 << 7) /* CS42L73_DMMCC */ -#define MCLKDIS (1 << 0) -#define MCLKSEL_MCLK2 (1 << 4) -#define MCLKSEL_MCLK1 (0 << 4) +#define CS42L73_MCLKDIS (1 << 0) +#define CS42L73_MCLKSEL_MCLK2 (1 << 4) +#define CS42L73_MCLKSEL_MCLK1 (0 << 4) /* CS42L73 MCLK derived from MCLK1 or MCLK2 */ #define CS42L73_CLKID_MCLK1 0 @@ -194,28 +195,26 @@ #define CS42L73_VSP 2 /* IS1, IM1 */ -#define MIC2_SDET (1 << 6) -#define THMOVLD (1 << 4) -#define DIGMIXOVFL (1 << 3) -#define IPBOVFL (1 << 1) -#define IPAOVFL (1 << 0) +#define CS42L73_MIC2_SDET (1 << 6) +#define CS42L73_THMOVLD (1 << 4) +#define CS42L73_DIGMIXOVFL (1 << 3) +#define CS42L73_IPBOVFL (1 << 1) +#define CS42L73_IPAOVFL (1 << 0) /* Analog Softramp */ -#define ANLGOSFT (1 << 0) +#define CS42L73_ANLGOSFT (1 << 0) /* HP A/B Analog Mute */ -#define HPA_MUTE (1 << 7) +#define CS42L73_HPA_MUTE (1 << 7) /* LO A/B Analog Mute */ -#define LOA_MUTE (1 << 7) +#define CS42L73_LOA_MUTE (1 << 7) /* Digital Mute */ -#define HLAD_MUTE (1 << 0) -#define HLBD_MUTE (1 << 1) -#define SPKD_MUTE (1 << 2) -#define ESLD_MUTE (1 << 3) +#define CS42L73_HLAD_MUTE (1 << 0) +#define CS42L73_HLBD_MUTE (1 << 1) +#define CS42L73_SPKD_MUTE (1 << 2) +#define CS42L73_ESLD_MUTE (1 << 3) /* Misc defines for codec */ -#define CS42L73_RESET_GPIO 143 - #define CS42L73_DEVID 0x00042A73 #define CS42L73_MCLKX_MIN 5644800 #define CS42L73_MCLKX_MAX 38400000 diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 8bd2d8a6a2f..53d7dab4e05 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -15,6 +15,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -38,294 +39,223 @@ struct max98088_cdata { }; struct max98088_priv { - enum max98088_type devtype; - struct max98088_pdata *pdata; - unsigned int sysclk; - struct max98088_cdata dai[2]; - int eq_textcnt; - const char **eq_texts; - struct soc_enum eq_enum; - u8 ina_state; - u8 inb_state; - unsigned int ex_mode; - unsigned int digmic; - unsigned int mic1pre; - unsigned int mic2pre; - unsigned int extmic_mode; + struct regmap *regmap; + enum max98088_type devtype; + struct max98088_pdata *pdata; + unsigned int sysclk; + struct max98088_cdata dai[2]; + int eq_textcnt; + const char **eq_texts; + struct soc_enum eq_enum; + u8 ina_state; + u8 inb_state; + unsigned int ex_mode; + unsigned int digmic; + unsigned int mic1pre; + unsigned int mic2pre; + unsigned int extmic_mode; }; -static const u8 max98088_reg[M98088_REG_CNT] = { - 0x00, /* 00 IRQ status */ - 0x00, /* 01 MIC status */ - 0x00, /* 02 jack status */ - 0x00, /* 03 battery voltage */ - 0x00, /* 04 */ - 0x00, /* 05 */ - 0x00, /* 06 */ - 0x00, /* 07 */ - 0x00, /* 08 */ - 0x00, /* 09 */ - 0x00, /* 0A */ - 0x00, /* 0B */ - 0x00, /* 0C */ - 0x00, /* 0D */ - 0x00, /* 0E */ - 0x00, /* 0F interrupt enable */ - - 0x00, /* 10 master clock */ - 0x00, /* 11 DAI1 clock mode */ - 0x00, /* 12 DAI1 clock control */ - 0x00, /* 13 DAI1 clock control */ - 0x00, /* 14 DAI1 format */ - 0x00, /* 15 DAI1 clock */ - 0x00, /* 16 DAI1 config */ - 0x00, /* 17 DAI1 TDM */ - 0x00, /* 18 DAI1 filters */ - 0x00, /* 19 DAI2 clock mode */ - 0x00, /* 1A DAI2 clock control */ - 0x00, /* 1B DAI2 clock control */ - 0x00, /* 1C DAI2 format */ - 0x00, /* 1D DAI2 clock */ - 0x00, /* 1E DAI2 config */ - 0x00, /* 1F DAI2 TDM */ - - 0x00, /* 20 DAI2 filters */ - 0x00, /* 21 data config */ - 0x00, /* 22 DAC mixer */ - 0x00, /* 23 left ADC mixer */ - 0x00, /* 24 right ADC mixer */ - 0x00, /* 25 left HP mixer */ - 0x00, /* 26 right HP mixer */ - 0x00, /* 27 HP control */ - 0x00, /* 28 left REC mixer */ - 0x00, /* 29 right REC mixer */ - 0x00, /* 2A REC control */ - 0x00, /* 2B left SPK mixer */ - 0x00, /* 2C right SPK mixer */ - 0x00, /* 2D SPK control */ - 0x00, /* 2E sidetone */ - 0x00, /* 2F DAI1 playback level */ - - 0x00, /* 30 DAI1 playback level */ - 0x00, /* 31 DAI2 playback level */ - 0x00, /* 32 DAI2 playbakc level */ - 0x00, /* 33 left ADC level */ - 0x00, /* 34 right ADC level */ - 0x00, /* 35 MIC1 level */ - 0x00, /* 36 MIC2 level */ - 0x00, /* 37 INA level */ - 0x00, /* 38 INB level */ - 0x00, /* 39 left HP volume */ - 0x00, /* 3A right HP volume */ - 0x00, /* 3B left REC volume */ - 0x00, /* 3C right REC volume */ - 0x00, /* 3D left SPK volume */ - 0x00, /* 3E right SPK volume */ - 0x00, /* 3F MIC config */ - - 0x00, /* 40 MIC threshold */ - 0x00, /* 41 excursion limiter filter */ - 0x00, /* 42 excursion limiter threshold */ - 0x00, /* 43 ALC */ - 0x00, /* 44 power limiter threshold */ - 0x00, /* 45 power limiter config */ - 0x00, /* 46 distortion limiter config */ - 0x00, /* 47 audio input */ - 0x00, /* 48 microphone */ - 0x00, /* 49 level control */ - 0x00, /* 4A bypass switches */ - 0x00, /* 4B jack detect */ - 0x00, /* 4C input enable */ - 0x00, /* 4D output enable */ - 0xF0, /* 4E bias control */ - 0x00, /* 4F DAC power */ - - 0x0F, /* 50 DAC power */ - 0x00, /* 51 system */ - 0x00, /* 52 DAI1 EQ1 */ - 0x00, /* 53 DAI1 EQ1 */ - 0x00, /* 54 DAI1 EQ1 */ - 0x00, /* 55 DAI1 EQ1 */ - 0x00, /* 56 DAI1 EQ1 */ - 0x00, /* 57 DAI1 EQ1 */ - 0x00, /* 58 DAI1 EQ1 */ - 0x00, /* 59 DAI1 EQ1 */ - 0x00, /* 5A DAI1 EQ1 */ - 0x00, /* 5B DAI1 EQ1 */ - 0x00, /* 5C DAI1 EQ2 */ - 0x00, /* 5D DAI1 EQ2 */ - 0x00, /* 5E DAI1 EQ2 */ - 0x00, /* 5F DAI1 EQ2 */ - - 0x00, /* 60 DAI1 EQ2 */ - 0x00, /* 61 DAI1 EQ2 */ - 0x00, /* 62 DAI1 EQ2 */ - 0x00, /* 63 DAI1 EQ2 */ - 0x00, /* 64 DAI1 EQ2 */ - 0x00, /* 65 DAI1 EQ2 */ - 0x00, /* 66 DAI1 EQ3 */ - 0x00, /* 67 DAI1 EQ3 */ - 0x00, /* 68 DAI1 EQ3 */ - 0x00, /* 69 DAI1 EQ3 */ - 0x00, /* 6A DAI1 EQ3 */ - 0x00, /* 6B DAI1 EQ3 */ - 0x00, /* 6C DAI1 EQ3 */ - 0x00, /* 6D DAI1 EQ3 */ - 0x00, /* 6E DAI1 EQ3 */ - 0x00, /* 6F DAI1 EQ3 */ - - 0x00, /* 70 DAI1 EQ4 */ - 0x00, /* 71 DAI1 EQ4 */ - 0x00, /* 72 DAI1 EQ4 */ - 0x00, /* 73 DAI1 EQ4 */ - 0x00, /* 74 DAI1 EQ4 */ - 0x00, /* 75 DAI1 EQ4 */ - 0x00, /* 76 DAI1 EQ4 */ - 0x00, /* 77 DAI1 EQ4 */ - 0x00, /* 78 DAI1 EQ4 */ - 0x00, /* 79 DAI1 EQ4 */ - 0x00, /* 7A DAI1 EQ5 */ - 0x00, /* 7B DAI1 EQ5 */ - 0x00, /* 7C DAI1 EQ5 */ - 0x00, /* 7D DAI1 EQ5 */ - 0x00, /* 7E DAI1 EQ5 */ - 0x00, /* 7F DAI1 EQ5 */ - - 0x00, /* 80 DAI1 EQ5 */ - 0x00, /* 81 DAI1 EQ5 */ - 0x00, /* 82 DAI1 EQ5 */ - 0x00, /* 83 DAI1 EQ5 */ - 0x00, /* 84 DAI2 EQ1 */ - 0x00, /* 85 DAI2 EQ1 */ - 0x00, /* 86 DAI2 EQ1 */ - 0x00, /* 87 DAI2 EQ1 */ - 0x00, /* 88 DAI2 EQ1 */ - 0x00, /* 89 DAI2 EQ1 */ - 0x00, /* 8A DAI2 EQ1 */ - 0x00, /* 8B DAI2 EQ1 */ - 0x00, /* 8C DAI2 EQ1 */ - 0x00, /* 8D DAI2 EQ1 */ - 0x00, /* 8E DAI2 EQ2 */ - 0x00, /* 8F DAI2 EQ2 */ - - 0x00, /* 90 DAI2 EQ2 */ - 0x00, /* 91 DAI2 EQ2 */ - 0x00, /* 92 DAI2 EQ2 */ - 0x00, /* 93 DAI2 EQ2 */ - 0x00, /* 94 DAI2 EQ2 */ - 0x00, /* 95 DAI2 EQ2 */ - 0x00, /* 96 DAI2 EQ2 */ - 0x00, /* 97 DAI2 EQ2 */ - 0x00, /* 98 DAI2 EQ3 */ - 0x00, /* 99 DAI2 EQ3 */ - 0x00, /* 9A DAI2 EQ3 */ - 0x00, /* 9B DAI2 EQ3 */ - 0x00, /* 9C DAI2 EQ3 */ - 0x00, /* 9D DAI2 EQ3 */ - 0x00, /* 9E DAI2 EQ3 */ - 0x00, /* 9F DAI2 EQ3 */ - - 0x00, /* A0 DAI2 EQ3 */ - 0x00, /* A1 DAI2 EQ3 */ - 0x00, /* A2 DAI2 EQ4 */ - 0x00, /* A3 DAI2 EQ4 */ - 0x00, /* A4 DAI2 EQ4 */ - 0x00, /* A5 DAI2 EQ4 */ - 0x00, /* A6 DAI2 EQ4 */ - 0x00, /* A7 DAI2 EQ4 */ - 0x00, /* A8 DAI2 EQ4 */ - 0x00, /* A9 DAI2 EQ4 */ - 0x00, /* AA DAI2 EQ4 */ - 0x00, /* AB DAI2 EQ4 */ - 0x00, /* AC DAI2 EQ5 */ - 0x00, /* AD DAI2 EQ5 */ - 0x00, /* AE DAI2 EQ5 */ - 0x00, /* AF DAI2 EQ5 */ - - 0x00, /* B0 DAI2 EQ5 */ - 0x00, /* B1 DAI2 EQ5 */ - 0x00, /* B2 DAI2 EQ5 */ - 0x00, /* B3 DAI2 EQ5 */ - 0x00, /* B4 DAI2 EQ5 */ - 0x00, /* B5 DAI2 EQ5 */ - 0x00, /* B6 DAI1 biquad */ - 0x00, /* B7 DAI1 biquad */ - 0x00, /* B8 DAI1 biquad */ - 0x00, /* B9 DAI1 biquad */ - 0x00, /* BA DAI1 biquad */ - 0x00, /* BB DAI1 biquad */ - 0x00, /* BC DAI1 biquad */ - 0x00, /* BD DAI1 biquad */ - 0x00, /* BE DAI1 biquad */ - 0x00, /* BF DAI1 biquad */ - - 0x00, /* C0 DAI2 biquad */ - 0x00, /* C1 DAI2 biquad */ - 0x00, /* C2 DAI2 biquad */ - 0x00, /* C3 DAI2 biquad */ - 0x00, /* C4 DAI2 biquad */ - 0x00, /* C5 DAI2 biquad */ - 0x00, /* C6 DAI2 biquad */ - 0x00, /* C7 DAI2 biquad */ - 0x00, /* C8 DAI2 biquad */ - 0x00, /* C9 DAI2 biquad */ - 0x00, /* CA */ - 0x00, /* CB */ - 0x00, /* CC */ - 0x00, /* CD */ - 0x00, /* CE */ - 0x00, /* CF */ - - 0x00, /* D0 */ - 0x00, /* D1 */ - 0x00, /* D2 */ - 0x00, /* D3 */ - 0x00, /* D4 */ - 0x00, /* D5 */ - 0x00, /* D6 */ - 0x00, /* D7 */ - 0x00, /* D8 */ - 0x00, /* D9 */ - 0x00, /* DA */ - 0x70, /* DB */ - 0x00, /* DC */ - 0x00, /* DD */ - 0x00, /* DE */ - 0x00, /* DF */ - - 0x00, /* E0 */ - 0x00, /* E1 */ - 0x00, /* E2 */ - 0x00, /* E3 */ - 0x00, /* E4 */ - 0x00, /* E5 */ - 0x00, /* E6 */ - 0x00, /* E7 */ - 0x00, /* E8 */ - 0x00, /* E9 */ - 0x00, /* EA */ - 0x00, /* EB */ - 0x00, /* EC */ - 0x00, /* ED */ - 0x00, /* EE */ - 0x00, /* EF */ - - 0x00, /* F0 */ - 0x00, /* F1 */ - 0x00, /* F2 */ - 0x00, /* F3 */ - 0x00, /* F4 */ - 0x00, /* F5 */ - 0x00, /* F6 */ - 0x00, /* F7 */ - 0x00, /* F8 */ - 0x00, /* F9 */ - 0x00, /* FA */ - 0x00, /* FB */ - 0x00, /* FC */ - 0x00, /* FD */ - 0x00, /* FE */ - 0x00, /* FF */ +static const struct reg_default max98088_reg[] = { + { 0xf, 0x00 }, /* 0F interrupt enable */ + + { 0x10, 0x00 }, /* 10 master clock */ + { 0x11, 0x00 }, /* 11 DAI1 clock mode */ + { 0x12, 0x00 }, /* 12 DAI1 clock control */ + { 0x13, 0x00 }, /* 13 DAI1 clock control */ + { 0x14, 0x00 }, /* 14 DAI1 format */ + { 0x15, 0x00 }, /* 15 DAI1 clock */ + { 0x16, 0x00 }, /* 16 DAI1 config */ + { 0x17, 0x00 }, /* 17 DAI1 TDM */ + { 0x18, 0x00 }, /* 18 DAI1 filters */ + { 0x19, 0x00 }, /* 19 DAI2 clock mode */ + { 0x1a, 0x00 }, /* 1A DAI2 clock control */ + { 0x1b, 0x00 }, /* 1B DAI2 clock control */ + { 0x1c, 0x00 }, /* 1C DAI2 format */ + { 0x1d, 0x00 }, /* 1D DAI2 clock */ + { 0x1e, 0x00 }, /* 1E DAI2 config */ + { 0x1f, 0x00 }, /* 1F DAI2 TDM */ + + { 0x20, 0x00 }, /* 20 DAI2 filters */ + { 0x21, 0x00 }, /* 21 data config */ + { 0x22, 0x00 }, /* 22 DAC mixer */ + { 0x23, 0x00 }, /* 23 left ADC mixer */ + { 0x24, 0x00 }, /* 24 right ADC mixer */ + { 0x25, 0x00 }, /* 25 left HP mixer */ + { 0x26, 0x00 }, /* 26 right HP mixer */ + { 0x27, 0x00 }, /* 27 HP control */ + { 0x28, 0x00 }, /* 28 left REC mixer */ + { 0x29, 0x00 }, /* 29 right REC mixer */ + { 0x2a, 0x00 }, /* 2A REC control */ + { 0x2b, 0x00 }, /* 2B left SPK mixer */ + { 0x2c, 0x00 }, /* 2C right SPK mixer */ + { 0x2d, 0x00 }, /* 2D SPK control */ + { 0x2e, 0x00 }, /* 2E sidetone */ + { 0x2f, 0x00 }, /* 2F DAI1 playback level */ + + { 0x30, 0x00 }, /* 30 DAI1 playback level */ + { 0x31, 0x00 }, /* 31 DAI2 playback level */ + { 0x32, 0x00 }, /* 32 DAI2 playbakc level */ + { 0x33, 0x00 }, /* 33 left ADC level */ + { 0x34, 0x00 }, /* 34 right ADC level */ + { 0x35, 0x00 }, /* 35 MIC1 level */ + { 0x36, 0x00 }, /* 36 MIC2 level */ + { 0x37, 0x00 }, /* 37 INA level */ + { 0x38, 0x00 }, /* 38 INB level */ + { 0x39, 0x00 }, /* 39 left HP volume */ + { 0x3a, 0x00 }, /* 3A right HP volume */ + { 0x3b, 0x00 }, /* 3B left REC volume */ + { 0x3c, 0x00 }, /* 3C right REC volume */ + { 0x3d, 0x00 }, /* 3D left SPK volume */ + { 0x3e, 0x00 }, /* 3E right SPK volume */ + { 0x3f, 0x00 }, /* 3F MIC config */ + + { 0x40, 0x00 }, /* 40 MIC threshold */ + { 0x41, 0x00 }, /* 41 excursion limiter filter */ + { 0x42, 0x00 }, /* 42 excursion limiter threshold */ + { 0x43, 0x00 }, /* 43 ALC */ + { 0x44, 0x00 }, /* 44 power limiter threshold */ + { 0x45, 0x00 }, /* 45 power limiter config */ + { 0x46, 0x00 }, /* 46 distortion limiter config */ + { 0x47, 0x00 }, /* 47 audio input */ + { 0x48, 0x00 }, /* 48 microphone */ + { 0x49, 0x00 }, /* 49 level control */ + { 0x4a, 0x00 }, /* 4A bypass switches */ + { 0x4b, 0x00 }, /* 4B jack detect */ + { 0x4c, 0x00 }, /* 4C input enable */ + { 0x4d, 0x00 }, /* 4D output enable */ + { 0x4e, 0xF0 }, /* 4E bias control */ + { 0x4f, 0x00 }, /* 4F DAC power */ + + { 0x50, 0x0F }, /* 50 DAC power */ + { 0x51, 0x00 }, /* 51 system */ + { 0x52, 0x00 }, /* 52 DAI1 EQ1 */ + { 0x53, 0x00 }, /* 53 DAI1 EQ1 */ + { 0x54, 0x00 }, /* 54 DAI1 EQ1 */ + { 0x55, 0x00 }, /* 55 DAI1 EQ1 */ + { 0x56, 0x00 }, /* 56 DAI1 EQ1 */ + { 0x57, 0x00 }, /* 57 DAI1 EQ1 */ + { 0x58, 0x00 }, /* 58 DAI1 EQ1 */ + { 0x59, 0x00 }, /* 59 DAI1 EQ1 */ + { 0x5a, 0x00 }, /* 5A DAI1 EQ1 */ + { 0x5b, 0x00 }, /* 5B DAI1 EQ1 */ + { 0x5c, 0x00 }, /* 5C DAI1 EQ2 */ + { 0x5d, 0x00 }, /* 5D DAI1 EQ2 */ + { 0x5e, 0x00 }, /* 5E DAI1 EQ2 */ + { 0x5f, 0x00 }, /* 5F DAI1 EQ2 */ + + { 0x60, 0x00 }, /* 60 DAI1 EQ2 */ + { 0x61, 0x00 }, /* 61 DAI1 EQ2 */ + { 0x62, 0x00 }, /* 62 DAI1 EQ2 */ + { 0x63, 0x00 }, /* 63 DAI1 EQ2 */ + { 0x64, 0x00 }, /* 64 DAI1 EQ2 */ + { 0x65, 0x00 }, /* 65 DAI1 EQ2 */ + { 0x66, 0x00 }, /* 66 DAI1 EQ3 */ + { 0x67, 0x00 }, /* 67 DAI1 EQ3 */ + { 0x68, 0x00 }, /* 68 DAI1 EQ3 */ + { 0x69, 0x00 }, /* 69 DAI1 EQ3 */ + { 0x6a, 0x00 }, /* 6A DAI1 EQ3 */ + { 0x6b, 0x00 }, /* 6B DAI1 EQ3 */ + { 0x6c, 0x00 }, /* 6C DAI1 EQ3 */ + { 0x6d, 0x00 }, /* 6D DAI1 EQ3 */ + { 0x6e, 0x00 }, /* 6E DAI1 EQ3 */ + { 0x6f, 0x00 }, /* 6F DAI1 EQ3 */ + + { 0x70, 0x00 }, /* 70 DAI1 EQ4 */ + { 0x71, 0x00 }, /* 71 DAI1 EQ4 */ + { 0x72, 0x00 }, /* 72 DAI1 EQ4 */ + { 0x73, 0x00 }, /* 73 DAI1 EQ4 */ + { 0x74, 0x00 }, /* 74 DAI1 EQ4 */ + { 0x75, 0x00 }, /* 75 DAI1 EQ4 */ + { 0x76, 0x00 }, /* 76 DAI1 EQ4 */ + { 0x77, 0x00 }, /* 77 DAI1 EQ4 */ + { 0x78, 0x00 }, /* 78 DAI1 EQ4 */ + { 0x79, 0x00 }, /* 79 DAI1 EQ4 */ + { 0x7a, 0x00 }, /* 7A DAI1 EQ5 */ + { 0x7b, 0x00 }, /* 7B DAI1 EQ5 */ + { 0x7c, 0x00 }, /* 7C DAI1 EQ5 */ + { 0x7d, 0x00 }, /* 7D DAI1 EQ5 */ + { 0x7e, 0x00 }, /* 7E DAI1 EQ5 */ + { 0x7f, 0x00 }, /* 7F DAI1 EQ5 */ + + { 0x80, 0x00 }, /* 80 DAI1 EQ5 */ + { 0x81, 0x00 }, /* 81 DAI1 EQ5 */ + { 0x82, 0x00 }, /* 82 DAI1 EQ5 */ + { 0x83, 0x00 }, /* 83 DAI1 EQ5 */ + { 0x84, 0x00 }, /* 84 DAI2 EQ1 */ + { 0x85, 0x00 }, /* 85 DAI2 EQ1 */ + { 0x86, 0x00 }, /* 86 DAI2 EQ1 */ + { 0x87, 0x00 }, /* 87 DAI2 EQ1 */ + { 0x88, 0x00 }, /* 88 DAI2 EQ1 */ + { 0x89, 0x00 }, /* 89 DAI2 EQ1 */ + { 0x8a, 0x00 }, /* 8A DAI2 EQ1 */ + { 0x8b, 0x00 }, /* 8B DAI2 EQ1 */ + { 0x8c, 0x00 }, /* 8C DAI2 EQ1 */ + { 0x8d, 0x00 }, /* 8D DAI2 EQ1 */ + { 0x8e, 0x00 }, /* 8E DAI2 EQ2 */ + { 0x8f, 0x00 }, /* 8F DAI2 EQ2 */ + + { 0x90, 0x00 }, /* 90 DAI2 EQ2 */ + { 0x91, 0x00 }, /* 91 DAI2 EQ2 */ + { 0x92, 0x00 }, /* 92 DAI2 EQ2 */ + { 0x93, 0x00 }, /* 93 DAI2 EQ2 */ + { 0x94, 0x00 }, /* 94 DAI2 EQ2 */ + { 0x95, 0x00 }, /* 95 DAI2 EQ2 */ + { 0x96, 0x00 }, /* 96 DAI2 EQ2 */ + { 0x97, 0x00 }, /* 97 DAI2 EQ2 */ + { 0x98, 0x00 }, /* 98 DAI2 EQ3 */ + { 0x99, 0x00 }, /* 99 DAI2 EQ3 */ + { 0x9a, 0x00 }, /* 9A DAI2 EQ3 */ + { 0x9b, 0x00 }, /* 9B DAI2 EQ3 */ + { 0x9c, 0x00 }, /* 9C DAI2 EQ3 */ + { 0x9d, 0x00 }, /* 9D DAI2 EQ3 */ + { 0x9e, 0x00 }, /* 9E DAI2 EQ3 */ + { 0x9f, 0x00 }, /* 9F DAI2 EQ3 */ + + { 0xa0, 0x00 }, /* A0 DAI2 EQ3 */ + { 0xa1, 0x00 }, /* A1 DAI2 EQ3 */ + { 0xa2, 0x00 }, /* A2 DAI2 EQ4 */ + { 0xa3, 0x00 }, /* A3 DAI2 EQ4 */ + { 0xa4, 0x00 }, /* A4 DAI2 EQ4 */ + { 0xa5, 0x00 }, /* A5 DAI2 EQ4 */ + { 0xa6, 0x00 }, /* A6 DAI2 EQ4 */ + { 0xa7, 0x00 }, /* A7 DAI2 EQ4 */ + { 0xa8, 0x00 }, /* A8 DAI2 EQ4 */ + { 0xa9, 0x00 }, /* A9 DAI2 EQ4 */ + { 0xaa, 0x00 }, /* AA DAI2 EQ4 */ + { 0xab, 0x00 }, /* AB DAI2 EQ4 */ + { 0xac, 0x00 }, /* AC DAI2 EQ5 */ + { 0xad, 0x00 }, /* AD DAI2 EQ5 */ + { 0xae, 0x00 }, /* AE DAI2 EQ5 */ + { 0xaf, 0x00 }, /* AF DAI2 EQ5 */ + + { 0xb0, 0x00 }, /* B0 DAI2 EQ5 */ + { 0xb1, 0x00 }, /* B1 DAI2 EQ5 */ + { 0xb2, 0x00 }, /* B2 DAI2 EQ5 */ + { 0xb3, 0x00 }, /* B3 DAI2 EQ5 */ + { 0xb4, 0x00 }, /* B4 DAI2 EQ5 */ + { 0xb5, 0x00 }, /* B5 DAI2 EQ5 */ + { 0xb6, 0x00 }, /* B6 DAI1 biquad */ + { 0xb7, 0x00 }, /* B7 DAI1 biquad */ + { 0xb8 ,0x00 }, /* B8 DAI1 biquad */ + { 0xb9, 0x00 }, /* B9 DAI1 biquad */ + { 0xba, 0x00 }, /* BA DAI1 biquad */ + { 0xbb, 0x00 }, /* BB DAI1 biquad */ + { 0xbc, 0x00 }, /* BC DAI1 biquad */ + { 0xbd, 0x00 }, /* BD DAI1 biquad */ + { 0xbe, 0x00 }, /* BE DAI1 biquad */ + { 0xbf, 0x00 }, /* BF DAI1 biquad */ + + { 0xc0, 0x00 }, /* C0 DAI2 biquad */ + { 0xc1, 0x00 }, /* C1 DAI2 biquad */ + { 0xc2, 0x00 }, /* C2 DAI2 biquad */ + { 0xc3, 0x00 }, /* C3 DAI2 biquad */ + { 0xc4, 0x00 }, /* C4 DAI2 biquad */ + { 0xc5, 0x00 }, /* C5 DAI2 biquad */ + { 0xc6, 0x00 }, /* C6 DAI2 biquad */ + { 0xc7, 0x00 }, /* C7 DAI2 biquad */ + { 0xc8, 0x00 }, /* C8 DAI2 biquad */ + { 0xc9, 0x00 }, /* C9 DAI2 biquad */ }; static struct { @@ -606,11 +536,28 @@ static struct { { 0xFF, 0x00, 1 }, /* FF */ }; -static int max98088_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool max98088_readable_register(struct device *dev, unsigned int reg) +{ + return max98088_access[reg].readable; +} + +static bool max98088_volatile_register(struct device *dev, unsigned int reg) { return max98088_access[reg].vol; } +static const struct regmap_config max98088_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .readable_reg = max98088_readable_register, + .volatile_reg = max98088_volatile_register, + .max_register = 0xff, + + .reg_defaults = max98088_reg, + .num_reg_defaults = ARRAY_SIZE(max98088_reg), + .cache_type = REGCACHE_RBTREE, +}; /* * Load equalizer DSP coefficient configurations registers @@ -1612,58 +1559,34 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } -static void max98088_sync_cache(struct snd_soc_codec *codec) -{ - u8 *reg_cache = codec->reg_cache; - int i; - - if (!codec->cache_sync) - return; - - codec->cache_only = 0; - - /* write back cached values if they're writeable and - * different from the hardware default. - */ - for (i = 1; i < codec->driver->reg_cache_size; i++) { - if (!max98088_access[i].writable) - continue; - - if (reg_cache[i] == max98088_reg[i]) - continue; - - snd_soc_write(codec, i, reg_cache[i]); - } - - codec->cache_sync = 0; -} - static int max98088_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - switch (level) { - case SND_SOC_BIAS_ON: - break; - - case SND_SOC_BIAS_PREPARE: - break; - - case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) - max98088_sync_cache(codec); - - snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, - M98088_MBEN, M98088_MBEN); - break; - - case SND_SOC_BIAS_OFF: - snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, - M98088_MBEN, 0); - codec->cache_sync = 1; - break; - } - codec->dapm.bias_level = level; - return 0; + struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(max98088->regmap); + + snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, + M98088_MBEN, M98088_MBEN); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, M98088_REG_4C_PWR_EN_IN, + M98088_MBEN, 0); + regcache_mark_dirty(max98088->regmap); + break; + } + codec->dapm.bias_level = level; + return 0; } #define MAX98088_RATES SNDRV_PCM_RATE_8000_96000 @@ -1990,9 +1913,9 @@ static int max98088_probe(struct snd_soc_codec *codec) struct max98088_cdata *cdata; int ret = 0; - codec->cache_sync = 1; + regcache_mark_dirty(max98088->regmap); - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -2050,9 +1973,6 @@ static int max98088_probe(struct snd_soc_codec *codec) max98088_handle_pdata(codec); - snd_soc_add_codec_controls(codec, max98088_snd_controls, - ARRAY_SIZE(max98088_snd_controls)); - err_access: return ret; } @@ -2068,15 +1988,13 @@ static int max98088_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_max98088 = { - .probe = max98088_probe, - .remove = max98088_remove, - .suspend = max98088_suspend, - .resume = max98088_resume, - .set_bias_level = max98088_set_bias_level, - .reg_cache_size = ARRAY_SIZE(max98088_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = max98088_reg, - .volatile_register = max98088_volatile_register, + .probe = max98088_probe, + .remove = max98088_remove, + .suspend = max98088_suspend, + .resume = max98088_resume, + .set_bias_level = max98088_set_bias_level, + .controls = max98088_snd_controls, + .num_controls = ARRAY_SIZE(max98088_snd_controls), .dapm_widgets = max98088_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98088_dapm_widgets), .dapm_routes = max98088_audio_map, @@ -2084,7 +2002,7 @@ static struct snd_soc_codec_driver soc_codec_dev_max98088 = { }; static int max98088_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) + const struct i2c_device_id *id) { struct max98088_priv *max98088; int ret; @@ -2094,6 +2012,10 @@ static int max98088_i2c_probe(struct i2c_client *i2c, if (max98088 == NULL) return -ENOMEM; + max98088->regmap = devm_regmap_init_i2c(i2c, &max98088_regmap); + if (IS_ERR(max98088->regmap)) + return PTR_ERR(max98088->regmap); + max98088->devtype = id->driver_data; i2c_set_clientdata(i2c, max98088); diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 04618a5f2a6..67244315c72 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -39,6 +39,7 @@ struct max98095_cdata { }; struct max98095_priv { + struct regmap *regmap; enum max98095_type devtype; struct max98095_pdata *pdata; unsigned int sysclk; @@ -56,263 +57,145 @@ struct max98095_priv { struct snd_soc_jack *mic_jack; }; -static const u8 max98095_reg_def[M98095_REG_CNT] = { - 0x00, /* 00 */ - 0x00, /* 01 */ - 0x00, /* 02 */ - 0x00, /* 03 */ - 0x00, /* 04 */ - 0x00, /* 05 */ - 0x00, /* 06 */ - 0x00, /* 07 */ - 0x00, /* 08 */ - 0x00, /* 09 */ - 0x00, /* 0A */ - 0x00, /* 0B */ - 0x00, /* 0C */ - 0x00, /* 0D */ - 0x00, /* 0E */ - 0x00, /* 0F */ - 0x00, /* 10 */ - 0x00, /* 11 */ - 0x00, /* 12 */ - 0x00, /* 13 */ - 0x00, /* 14 */ - 0x00, /* 15 */ - 0x00, /* 16 */ - 0x00, /* 17 */ - 0x00, /* 18 */ - 0x00, /* 19 */ - 0x00, /* 1A */ - 0x00, /* 1B */ - 0x00, /* 1C */ - 0x00, /* 1D */ - 0x00, /* 1E */ - 0x00, /* 1F */ - 0x00, /* 20 */ - 0x00, /* 21 */ - 0x00, /* 22 */ - 0x00, /* 23 */ - 0x00, /* 24 */ - 0x00, /* 25 */ - 0x00, /* 26 */ - 0x00, /* 27 */ - 0x00, /* 28 */ - 0x00, /* 29 */ - 0x00, /* 2A */ - 0x00, /* 2B */ - 0x00, /* 2C */ - 0x00, /* 2D */ - 0x00, /* 2E */ - 0x00, /* 2F */ - 0x00, /* 30 */ - 0x00, /* 31 */ - 0x00, /* 32 */ - 0x00, /* 33 */ - 0x00, /* 34 */ - 0x00, /* 35 */ - 0x00, /* 36 */ - 0x00, /* 37 */ - 0x00, /* 38 */ - 0x00, /* 39 */ - 0x00, /* 3A */ - 0x00, /* 3B */ - 0x00, /* 3C */ - 0x00, /* 3D */ - 0x00, /* 3E */ - 0x00, /* 3F */ - 0x00, /* 40 */ - 0x00, /* 41 */ - 0x00, /* 42 */ - 0x00, /* 43 */ - 0x00, /* 44 */ - 0x00, /* 45 */ - 0x00, /* 46 */ - 0x00, /* 47 */ - 0x00, /* 48 */ - 0x00, /* 49 */ - 0x00, /* 4A */ - 0x00, /* 4B */ - 0x00, /* 4C */ - 0x00, /* 4D */ - 0x00, /* 4E */ - 0x00, /* 4F */ - 0x00, /* 50 */ - 0x00, /* 51 */ - 0x00, /* 52 */ - 0x00, /* 53 */ - 0x00, /* 54 */ - 0x00, /* 55 */ - 0x00, /* 56 */ - 0x00, /* 57 */ - 0x00, /* 58 */ - 0x00, /* 59 */ - 0x00, /* 5A */ - 0x00, /* 5B */ - 0x00, /* 5C */ - 0x00, /* 5D */ - 0x00, /* 5E */ - 0x00, /* 5F */ - 0x00, /* 60 */ - 0x00, /* 61 */ - 0x00, /* 62 */ - 0x00, /* 63 */ - 0x00, /* 64 */ - 0x00, /* 65 */ - 0x00, /* 66 */ - 0x00, /* 67 */ - 0x00, /* 68 */ - 0x00, /* 69 */ - 0x00, /* 6A */ - 0x00, /* 6B */ - 0x00, /* 6C */ - 0x00, /* 6D */ - 0x00, /* 6E */ - 0x00, /* 6F */ - 0x00, /* 70 */ - 0x00, /* 71 */ - 0x00, /* 72 */ - 0x00, /* 73 */ - 0x00, /* 74 */ - 0x00, /* 75 */ - 0x00, /* 76 */ - 0x00, /* 77 */ - 0x00, /* 78 */ - 0x00, /* 79 */ - 0x00, /* 7A */ - 0x00, /* 7B */ - 0x00, /* 7C */ - 0x00, /* 7D */ - 0x00, /* 7E */ - 0x00, /* 7F */ - 0x00, /* 80 */ - 0x00, /* 81 */ - 0x00, /* 82 */ - 0x00, /* 83 */ - 0x00, /* 84 */ - 0x00, /* 85 */ - 0x00, /* 86 */ - 0x00, /* 87 */ - 0x00, /* 88 */ - 0x00, /* 89 */ - 0x00, /* 8A */ - 0x00, /* 8B */ - 0x00, /* 8C */ - 0x00, /* 8D */ - 0x00, /* 8E */ - 0x00, /* 8F */ - 0x00, /* 90 */ - 0x00, /* 91 */ - 0x30, /* 92 */ - 0xF0, /* 93 */ - 0x00, /* 94 */ - 0x00, /* 95 */ - 0x3F, /* 96 */ - 0x00, /* 97 */ - 0x00, /* 98 */ - 0x00, /* 99 */ - 0x00, /* 9A */ - 0x00, /* 9B */ - 0x00, /* 9C */ - 0x00, /* 9D */ - 0x00, /* 9E */ - 0x00, /* 9F */ - 0x00, /* A0 */ - 0x00, /* A1 */ - 0x00, /* A2 */ - 0x00, /* A3 */ - 0x00, /* A4 */ - 0x00, /* A5 */ - 0x00, /* A6 */ - 0x00, /* A7 */ - 0x00, /* A8 */ - 0x00, /* A9 */ - 0x00, /* AA */ - 0x00, /* AB */ - 0x00, /* AC */ - 0x00, /* AD */ - 0x00, /* AE */ - 0x00, /* AF */ - 0x00, /* B0 */ - 0x00, /* B1 */ - 0x00, /* B2 */ - 0x00, /* B3 */ - 0x00, /* B4 */ - 0x00, /* B5 */ - 0x00, /* B6 */ - 0x00, /* B7 */ - 0x00, /* B8 */ - 0x00, /* B9 */ - 0x00, /* BA */ - 0x00, /* BB */ - 0x00, /* BC */ - 0x00, /* BD */ - 0x00, /* BE */ - 0x00, /* BF */ - 0x00, /* C0 */ - 0x00, /* C1 */ - 0x00, /* C2 */ - 0x00, /* C3 */ - 0x00, /* C4 */ - 0x00, /* C5 */ - 0x00, /* C6 */ - 0x00, /* C7 */ - 0x00, /* C8 */ - 0x00, /* C9 */ - 0x00, /* CA */ - 0x00, /* CB */ - 0x00, /* CC */ - 0x00, /* CD */ - 0x00, /* CE */ - 0x00, /* CF */ - 0x00, /* D0 */ - 0x00, /* D1 */ - 0x00, /* D2 */ - 0x00, /* D3 */ - 0x00, /* D4 */ - 0x00, /* D5 */ - 0x00, /* D6 */ - 0x00, /* D7 */ - 0x00, /* D8 */ - 0x00, /* D9 */ - 0x00, /* DA */ - 0x00, /* DB */ - 0x00, /* DC */ - 0x00, /* DD */ - 0x00, /* DE */ - 0x00, /* DF */ - 0x00, /* E0 */ - 0x00, /* E1 */ - 0x00, /* E2 */ - 0x00, /* E3 */ - 0x00, /* E4 */ - 0x00, /* E5 */ - 0x00, /* E6 */ - 0x00, /* E7 */ - 0x00, /* E8 */ - 0x00, /* E9 */ - 0x00, /* EA */ - 0x00, /* EB */ - 0x00, /* EC */ - 0x00, /* ED */ - 0x00, /* EE */ - 0x00, /* EF */ - 0x00, /* F0 */ - 0x00, /* F1 */ - 0x00, /* F2 */ - 0x00, /* F3 */ - 0x00, /* F4 */ - 0x00, /* F5 */ - 0x00, /* F6 */ - 0x00, /* F7 */ - 0x00, /* F8 */ - 0x00, /* F9 */ - 0x00, /* FA */ - 0x00, /* FB */ - 0x00, /* FC */ - 0x00, /* FD */ - 0x00, /* FE */ - 0x00, /* FF */ +static const struct reg_default max98095_reg_def[] = { + { 0xf, 0x00 }, /* 0F */ + { 0x10, 0x00 }, /* 10 */ + { 0x11, 0x00 }, /* 11 */ + { 0x12, 0x00 }, /* 12 */ + { 0x13, 0x00 }, /* 13 */ + { 0x14, 0x00 }, /* 14 */ + { 0x15, 0x00 }, /* 15 */ + { 0x16, 0x00 }, /* 16 */ + { 0x17, 0x00 }, /* 17 */ + { 0x18, 0x00 }, /* 18 */ + { 0x19, 0x00 }, /* 19 */ + { 0x1a, 0x00 }, /* 1A */ + { 0x1b, 0x00 }, /* 1B */ + { 0x1c, 0x00 }, /* 1C */ + { 0x1d, 0x00 }, /* 1D */ + { 0x1e, 0x00 }, /* 1E */ + { 0x1f, 0x00 }, /* 1F */ + { 0x20, 0x00 }, /* 20 */ + { 0x21, 0x00 }, /* 21 */ + { 0x22, 0x00 }, /* 22 */ + { 0x23, 0x00 }, /* 23 */ + { 0x24, 0x00 }, /* 24 */ + { 0x25, 0x00 }, /* 25 */ + { 0x26, 0x00 }, /* 26 */ + { 0x27, 0x00 }, /* 27 */ + { 0x28, 0x00 }, /* 28 */ + { 0x29, 0x00 }, /* 29 */ + { 0x2a, 0x00 }, /* 2A */ + { 0x2b, 0x00 }, /* 2B */ + { 0x2c, 0x00 }, /* 2C */ + { 0x2d, 0x00 }, /* 2D */ + { 0x2e, 0x00 }, /* 2E */ + { 0x2f, 0x00 }, /* 2F */ + { 0x30, 0x00 }, /* 30 */ + { 0x31, 0x00 }, /* 31 */ + { 0x32, 0x00 }, /* 32 */ + { 0x33, 0x00 }, /* 33 */ + { 0x34, 0x00 }, /* 34 */ + { 0x35, 0x00 }, /* 35 */ + { 0x36, 0x00 }, /* 36 */ + { 0x37, 0x00 }, /* 37 */ + { 0x38, 0x00 }, /* 38 */ + { 0x39, 0x00 }, /* 39 */ + { 0x3a, 0x00 }, /* 3A */ + { 0x3b, 0x00 }, /* 3B */ + { 0x3c, 0x00 }, /* 3C */ + { 0x3d, 0x00 }, /* 3D */ + { 0x3e, 0x00 }, /* 3E */ + { 0x3f, 0x00 }, /* 3F */ + { 0x40, 0x00 }, /* 40 */ + { 0x41, 0x00 }, /* 41 */ + { 0x42, 0x00 }, /* 42 */ + { 0x43, 0x00 }, /* 43 */ + { 0x44, 0x00 }, /* 44 */ + { 0x45, 0x00 }, /* 45 */ + { 0x46, 0x00 }, /* 46 */ + { 0x47, 0x00 }, /* 47 */ + { 0x48, 0x00 }, /* 48 */ + { 0x49, 0x00 }, /* 49 */ + { 0x4a, 0x00 }, /* 4A */ + { 0x4b, 0x00 }, /* 4B */ + { 0x4c, 0x00 }, /* 4C */ + { 0x4d, 0x00 }, /* 4D */ + { 0x4e, 0x00 }, /* 4E */ + { 0x4f, 0x00 }, /* 4F */ + { 0x50, 0x00 }, /* 50 */ + { 0x51, 0x00 }, /* 51 */ + { 0x52, 0x00 }, /* 52 */ + { 0x53, 0x00 }, /* 53 */ + { 0x54, 0x00 }, /* 54 */ + { 0x55, 0x00 }, /* 55 */ + { 0x56, 0x00 }, /* 56 */ + { 0x57, 0x00 }, /* 57 */ + { 0x58, 0x00 }, /* 58 */ + { 0x59, 0x00 }, /* 59 */ + { 0x5a, 0x00 }, /* 5A */ + { 0x5b, 0x00 }, /* 5B */ + { 0x5c, 0x00 }, /* 5C */ + { 0x5d, 0x00 }, /* 5D */ + { 0x5e, 0x00 }, /* 5E */ + { 0x5f, 0x00 }, /* 5F */ + { 0x60, 0x00 }, /* 60 */ + { 0x61, 0x00 }, /* 61 */ + { 0x62, 0x00 }, /* 62 */ + { 0x63, 0x00 }, /* 63 */ + { 0x64, 0x00 }, /* 64 */ + { 0x65, 0x00 }, /* 65 */ + { 0x66, 0x00 }, /* 66 */ + { 0x67, 0x00 }, /* 67 */ + { 0x68, 0x00 }, /* 68 */ + { 0x69, 0x00 }, /* 69 */ + { 0x6a, 0x00 }, /* 6A */ + { 0x6b, 0x00 }, /* 6B */ + { 0x6c, 0x00 }, /* 6C */ + { 0x6d, 0x00 }, /* 6D */ + { 0x6e, 0x00 }, /* 6E */ + { 0x6f, 0x00 }, /* 6F */ + { 0x70, 0x00 }, /* 70 */ + { 0x71, 0x00 }, /* 71 */ + { 0x72, 0x00 }, /* 72 */ + { 0x73, 0x00 }, /* 73 */ + { 0x74, 0x00 }, /* 74 */ + { 0x75, 0x00 }, /* 75 */ + { 0x76, 0x00 }, /* 76 */ + { 0x77, 0x00 }, /* 77 */ + { 0x78, 0x00 }, /* 78 */ + { 0x79, 0x00 }, /* 79 */ + { 0x7a, 0x00 }, /* 7A */ + { 0x7b, 0x00 }, /* 7B */ + { 0x7c, 0x00 }, /* 7C */ + { 0x7d, 0x00 }, /* 7D */ + { 0x7e, 0x00 }, /* 7E */ + { 0x7f, 0x00 }, /* 7F */ + { 0x80, 0x00 }, /* 80 */ + { 0x81, 0x00 }, /* 81 */ + { 0x82, 0x00 }, /* 82 */ + { 0x83, 0x00 }, /* 83 */ + { 0x84, 0x00 }, /* 84 */ + { 0x85, 0x00 }, /* 85 */ + { 0x86, 0x00 }, /* 86 */ + { 0x87, 0x00 }, /* 87 */ + { 0x88, 0x00 }, /* 88 */ + { 0x89, 0x00 }, /* 89 */ + { 0x8a, 0x00 }, /* 8A */ + { 0x8b, 0x00 }, /* 8B */ + { 0x8c, 0x00 }, /* 8C */ + { 0x8d, 0x00 }, /* 8D */ + { 0x8e, 0x00 }, /* 8E */ + { 0x8f, 0x00 }, /* 8F */ + { 0x90, 0x00 }, /* 90 */ + { 0x91, 0x00 }, /* 91 */ + { 0x92, 0x30 }, /* 92 */ + { 0x93, 0xF0 }, /* 93 */ + { 0x94, 0x00 }, /* 94 */ + { 0x95, 0x00 }, /* 95 */ + { 0x96, 0x3F }, /* 96 */ + { 0x97, 0x00 }, /* 97 */ + { 0xff, 0x00 }, /* FF */ }; static struct { @@ -577,14 +460,14 @@ static struct { { 0xFF, 0x00 }, /* FF */ }; -static int max98095_readable(struct snd_soc_codec *codec, unsigned int reg) +static bool max98095_readable(struct device *dev, unsigned int reg) { if (reg >= M98095_REG_CNT) return 0; return max98095_access[reg].readable != 0; } -static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) +static bool max98095_volatile(struct device *dev, unsigned int reg) { if (reg > M98095_REG_MAX_CACHED) return 1; @@ -611,22 +494,18 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg) return 0; } -/* - * Filter coefficients are in a separate register segment - * and they share the address space of the normal registers. - * The coefficient registers do not need or share the cache. - */ -static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - int ret; +static const struct regmap_config max98095_regmap = { + .reg_bits = 8, + .val_bits = 8, - codec->cache_bypass = 1; - ret = snd_soc_write(codec, reg, value); - codec->cache_bypass = 0; + .reg_defaults = max98095_reg_def, + .num_reg_defaults = ARRAY_SIZE(max98095_reg_def), + .max_register = M98095_0FF_REV_ID, + .cache_type = REGCACHE_RBTREE, - return ret ? -EIO : 0; -} + .readable_reg = max98095_readable, + .volatile_reg = max98095_volatile, +}; /* * Load equalizer DSP coefficient configurations registers @@ -649,8 +528,8 @@ static void m98095_eq_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, eq_reg++, M98095_BYTE0(coefs[i])); } } @@ -675,8 +554,8 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai, /* Step through the registers and coefs */ for (i = 0; i < M98095_COEFS_PER_BAND; i++) { - max98095_hw_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); - max98095_hw_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE1(coefs[i])); + snd_soc_write(codec, bq_reg++, M98095_BYTE0(coefs[i])); } } @@ -1288,14 +1167,6 @@ static const struct snd_soc_dapm_route max98095_audio_map[] = { {"MIC2 Input", NULL, "MIC2"}, }; -static int max98095_add_widgets(struct snd_soc_codec *codec) -{ - snd_soc_add_codec_controls(codec, max98095_snd_controls, - ARRAY_SIZE(max98095_snd_controls)); - - return 0; -} - /* codec mclk clock divider coefficients */ static const struct { u32 rate; @@ -1751,6 +1622,7 @@ static int max98095_dai3_set_fmt(struct snd_soc_dai *codec_dai, static int max98095_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { @@ -1762,7 +1634,7 @@ static int max98095_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(max98095->regmap); if (ret != 0) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); @@ -1777,7 +1649,7 @@ static int max98095_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_update_bits(codec, M98095_090_PWR_EN_IN, M98095_MBEN, 0); - codec->cache_sync = 1; + regcache_mark_dirty(max98095->regmap); break; } codec->dapm.bias_level = level; @@ -2345,7 +2217,7 @@ static int max98095_reset(struct snd_soc_codec *codec) /* Reset to hardware default for registers, as there is not * a soft reset hardware control register */ for (i = M98095_010_HOST_INT_CFG; i < M98095_REG_MAX_CACHED; i++) { - ret = snd_soc_write(codec, i, max98095_reg_def[i]); + ret = snd_soc_write(codec, i, snd_soc_read(codec, i)); if (ret < 0) { dev_err(codec->dev, "Failed to reset: %d\n", ret); return ret; @@ -2362,7 +2234,7 @@ static int max98095_probe(struct snd_soc_codec *codec) struct i2c_client *client; int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -2451,8 +2323,6 @@ static int max98095_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, M98095_097_PWR_SYS, M98095_SHDNRUN, M98095_SHDNRUN); - max98095_add_widgets(codec); - return 0; err_irq: @@ -2484,11 +2354,8 @@ static struct snd_soc_codec_driver soc_codec_dev_max98095 = { .suspend = max98095_suspend, .resume = max98095_resume, .set_bias_level = max98095_set_bias_level, - .reg_cache_size = ARRAY_SIZE(max98095_reg_def), - .reg_word_size = sizeof(u8), - .reg_cache_default = max98095_reg_def, - .readable_register = max98095_readable, - .volatile_register = max98095_volatile, + .controls = max98095_snd_controls, + .num_controls = ARRAY_SIZE(max98095_snd_controls), .dapm_widgets = max98095_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98095_dapm_widgets), .dapm_routes = max98095_audio_map, @@ -2506,6 +2373,13 @@ static int max98095_i2c_probe(struct i2c_client *i2c, if (max98095 == NULL) return -ENOMEM; + max98095->regmap = devm_regmap_init_i2c(i2c, &max98095_regmap); + if (IS_ERR(max98095->regmap)) { + ret = PTR_ERR(max98095->regmap); + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return ret; + } + max98095->devtype = id->driver_data; i2c_set_clientdata(i2c, max98095); max98095->pdata = i2c->dev.platform_data; diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 58c38a5b481..c5dd61785f8 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -18,6 +18,7 @@ #include <linux/module.h> #include <linux/init.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -27,18 +28,26 @@ #include "max9850.h" struct max9850_priv { + struct regmap *regmap; unsigned int sysclk; }; /* max9850 register cache */ -static const u8 max9850_reg[MAX9850_CACHEREGNUM] = { - 0x00, 0x00, 0x0c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 +static const struct reg_default max9850_reg[] = { + { 2, 0x0c }, + { 3, 0x00 }, + { 4, 0x00 }, + { 5, 0x00 }, + { 6, 0x00 }, + { 7, 0x00 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x00 }, }; /* these registers are not used at the moment but provided for the sake of * completeness */ -static int max9850_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) +static bool max9850_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case MAX9850_STATUSA: @@ -49,6 +58,15 @@ static int max9850_volatile_register(struct snd_soc_codec *codec, } } +static const struct regmap_config max9850_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = MAX9850_DIGITAL_AUDIO, + .volatile_reg = max9850_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; + static const unsigned int max9850_tlv[] = { TLV_DB_RANGE_HEAD(4), 0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0), @@ -225,6 +243,7 @@ static int max9850_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) static int max9850_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec); int ret; switch (level) { @@ -234,7 +253,7 @@ static int max9850_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(max9850->regmap); if (ret) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); @@ -295,7 +314,7 @@ static int max9850_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -316,10 +335,6 @@ static struct snd_soc_codec_driver soc_codec_dev_max9850 = { .suspend = max9850_suspend, .resume = max9850_resume, .set_bias_level = max9850_set_bias_level, - .reg_cache_size = ARRAY_SIZE(max9850_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = max9850_reg, - .volatile_register = max9850_volatile_register, .controls = max9850_controls, .num_controls = ARRAY_SIZE(max9850_controls), @@ -340,6 +355,10 @@ static int max9850_i2c_probe(struct i2c_client *i2c, if (max9850 == NULL) return -ENOMEM; + max9850->regmap = devm_regmap_init_i2c(i2c, &max9850_regmap); + if (IS_ERR(max9850->regmap)) + return PTR_ERR(max9850->regmap); + i2c_set_clientdata(i2c, max9850); ret = snd_soc_register_codec(&i2c->dev, diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index ea141e1d6f2..f5472adee67 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -30,16 +30,10 @@ #include <sound/soc.h> #include <sound/initval.h> #include <sound/soc-dapm.h> +#include <linux/regmap.h> #include "mc13783.h" -#define MC13783_AUDIO_RX0 36 -#define MC13783_AUDIO_RX1 37 -#define MC13783_AUDIO_TX 38 -#define MC13783_SSI_NETWORK 39 -#define MC13783_AUDIO_CODEC 40 -#define MC13783_AUDIO_DAC 41 - #define AUDIO_RX0_ALSPEN (1 << 5) #define AUDIO_RX0_ALSPSEL (1 << 7) #define AUDIO_RX0_ADDCDC (1 << 21) @@ -95,45 +89,12 @@ struct mc13783_priv { struct mc13xxx *mc13xxx; + struct regmap *regmap; enum mc13783_ssi_port adc_ssi_port; enum mc13783_ssi_port dac_ssi_port; }; -static unsigned int mc13783_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); - unsigned int value = 0; - - mc13xxx_lock(priv->mc13xxx); - - mc13xxx_reg_read(priv->mc13xxx, reg, &value); - - mc13xxx_unlock(priv->mc13xxx); - - return value; -} - -static int mc13783_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); - int ret; - - mc13xxx_lock(priv->mc13xxx); - - ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); - - /* include errata fix for spi audio problems */ - if (reg == MC13783_AUDIO_CODEC || reg == MC13783_AUDIO_DAC) - ret = mc13xxx_reg_write(priv->mc13xxx, reg, value); - - mc13xxx_unlock(priv->mc13xxx); - - return ret; -} - /* Mapping between sample rates and register value */ static unsigned int mc13783_rates[] = { 8000, 11025, 12000, 16000, @@ -466,6 +427,29 @@ static const struct snd_kcontrol_new right_input_mux = static const struct snd_kcontrol_new samp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 3, 1, 0); +static const char * const speaker_amp_source_text[] = { + "CODEC", "Right" +}; +static const SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4, + speaker_amp_source_text); +static const struct snd_kcontrol_new speaker_amp_source_mux = + SOC_DAPM_ENUM("Speaker Amp Source MUX", speaker_amp_source); + +static const char * const headset_amp_source_text[] = { + "CODEC", "Mixer" +}; + +static const SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11, + headset_amp_source_text); +static const struct snd_kcontrol_new headset_amp_source_mux = + SOC_DAPM_ENUM("Headset Amp Source MUX", headset_amp_source); + +static const struct snd_kcontrol_new cdcout_ctl = + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 18, 1, 0); + +static const struct snd_kcontrol_new adc_bypass_ctl = + SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_CODEC, 16, 1, 0); + static const struct snd_kcontrol_new lamp_ctl = SOC_DAPM_SINGLE("Switch", MC13783_AUDIO_RX0, 5, 1, 0); @@ -503,12 +487,22 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_VIRT_MUX("PGA Right Input Mux", SND_SOC_NOPM, 0, 0, &right_input_mux), + SND_SOC_DAPM_MUX("Speaker Amp Source MUX", SND_SOC_NOPM, 0, 0, + &speaker_amp_source_mux), + + SND_SOC_DAPM_MUX("Headset Amp Source MUX", SND_SOC_NOPM, 0, 0, + &headset_amp_source_mux), + SND_SOC_DAPM_PGA("PGA Left Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("PGA Right Input", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_ADC("ADC", "Capture", MC13783_AUDIO_CODEC, 11, 0), SND_SOC_DAPM_SUPPLY("ADC_Reset", MC13783_AUDIO_CODEC, 15, 0, NULL, 0), + SND_SOC_DAPM_PGA("Voice CODEC PGA", MC13783_AUDIO_RX1, 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("Voice CODEC Bypass", MC13783_AUDIO_CODEC, 16, 0, + &adc_bypass_ctl), + /* Output */ SND_SOC_DAPM_SUPPLY("DAC_E", MC13783_AUDIO_DAC, 11, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DAC_Reset", MC13783_AUDIO_DAC, 15, 0, NULL, 0), @@ -516,10 +510,15 @@ static const struct snd_soc_dapm_widget mc13783_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("RXOUTR"), SND_SOC_DAPM_OUTPUT("HSL"), SND_SOC_DAPM_OUTPUT("HSR"), + SND_SOC_DAPM_OUTPUT("LSPL"), SND_SOC_DAPM_OUTPUT("LSP"), SND_SOC_DAPM_OUTPUT("SP"), + SND_SOC_DAPM_OUTPUT("CDCOUT"), - SND_SOC_DAPM_SWITCH("Speaker Amp", MC13783_AUDIO_RX0, 3, 0, &samp_ctl), + SND_SOC_DAPM_SWITCH("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 0, + &cdcout_ctl), + SND_SOC_DAPM_SWITCH("Speaker Amp Switch", MC13783_AUDIO_RX0, 3, 0, + &samp_ctl), SND_SOC_DAPM_SWITCH("Loudspeaker Amp", SND_SOC_NOPM, 0, 0, &lamp_ctl), SND_SOC_DAPM_SWITCH("Headset Amp Left", MC13783_AUDIO_RX0, 10, 0, &hlamp_ctl), @@ -554,20 +553,28 @@ static struct snd_soc_dapm_route mc13783_routes[] = { { "ADC", NULL, "PGA Right Input"}, { "ADC", NULL, "ADC_Reset"}, + { "Voice CODEC PGA", "Voice CODEC Bypass", "ADC" }, + + { "Speaker Amp Source MUX", "CODEC", "Voice CODEC PGA"}, + { "Speaker Amp Source MUX", "Right", "DAC PGA"}, + + { "Headset Amp Source MUX", "CODEC", "Voice CODEC PGA"}, + { "Headset Amp Source MUX", "Mixer", "DAC PGA"}, + /* Output */ { "HSL", NULL, "Headset Amp Left" }, { "HSR", NULL, "Headset Amp Right"}, { "RXOUTL", NULL, "Line out Amp Left"}, { "RXOUTR", NULL, "Line out Amp Right"}, - { "SP", NULL, "Speaker Amp"}, - { "Speaker Amp", NULL, "DAC PGA"}, - { "LSP", NULL, "DAC PGA"}, - { "Headset Amp Left", NULL, "DAC PGA"}, - { "Headset Amp Right", NULL, "DAC PGA"}, + { "SP", "Speaker Amp Switch", "Speaker Amp Source MUX"}, + { "LSP", "Loudspeaker Amp", "Speaker Amp Source MUX"}, + { "HSL", "Headset Amp Left", "Headset Amp Source MUX"}, + { "HSR", "Headset Amp Right", "Headset Amp Source MUX"}, { "Line out Amp Left", NULL, "DAC PGA"}, { "Line out Amp Right", NULL, "DAC PGA"}, { "DAC PGA", NULL, "DAC"}, { "DAC", NULL, "DAC_E"}, + { "CDCOUT", "CDCOUT Switch", "Voice CODEC PGA"}, }; static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix", @@ -580,15 +587,39 @@ static const struct soc_enum mc13783_enum_3d_mixer = static struct snd_kcontrol_new mc13783_control_list[] = { SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), SOC_SINGLE("PCM Playback Volume", MC13783_AUDIO_RX1, 6, 15, 0), + SOC_SINGLE("PCM Playback Switch", MC13783_AUDIO_RX1, 5, 1, 0), SOC_DOUBLE("PCM Capture Volume", MC13783_AUDIO_TX, 19, 14, 31, 0), SOC_ENUM("3D Control", mc13783_enum_3d_mixer), + + SOC_SINGLE("CDCOUT Switch", MC13783_AUDIO_RX0, 18, 1, 0), + SOC_SINGLE("Earpiece Amp Switch", MC13783_AUDIO_RX0, 3, 1, 0), + SOC_DOUBLE("Headset Amp Switch", MC13783_AUDIO_RX0, 10, 9, 1, 0), + SOC_DOUBLE("Line out Amp Switch", MC13783_AUDIO_RX0, 16, 15, 1, 0), + + SOC_SINGLE("PCM Capture Mixin Switch", MC13783_AUDIO_RX0, 22, 1, 0), + SOC_SINGLE("Line in Capture Mixin Switch", MC13783_AUDIO_RX0, 23, 1, 0), + + SOC_SINGLE("CODEC Capture Volume", MC13783_AUDIO_RX1, 1, 15, 0), + SOC_SINGLE("CODEC Capture Mixin Switch", MC13783_AUDIO_RX0, 21, 1, 0), + + SOC_SINGLE("Line in Capture Volume", MC13783_AUDIO_RX1, 12, 15, 0), + SOC_SINGLE("Line in Capture Switch", MC13783_AUDIO_RX1, 10, 1, 0), + + SOC_SINGLE("MC1 Capture Bias Switch", MC13783_AUDIO_TX, 0, 1, 0), + SOC_SINGLE("MC2 Capture Bias Switch", MC13783_AUDIO_TX, 1, 1, 0), }; static int mc13783_probe(struct snd_soc_codec *codec) { struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; - mc13xxx_lock(priv->mc13xxx); + codec->control_data = dev_get_regmap(codec->dev->parent, NULL); + ret = snd_soc_codec_set_cache_io(codec, 8, 24, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } /* these are the reset values */ mc13xxx_reg_write(priv->mc13xxx, MC13783_AUDIO_RX0, 0x25893); @@ -612,8 +643,6 @@ static int mc13783_probe(struct snd_soc_codec *codec) mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_DAC, 0, AUDIO_SSI_SEL); - mc13xxx_unlock(priv->mc13xxx); - return 0; } @@ -621,13 +650,9 @@ static int mc13783_remove(struct snd_soc_codec *codec) { struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); - mc13xxx_lock(priv->mc13xxx); - /* Make sure VAUDIOON is off */ mc13xxx_reg_rmw(priv->mc13xxx, MC13783_AUDIO_RX0, 0x3, 0); - mc13xxx_unlock(priv->mc13xxx); - return 0; } @@ -717,8 +742,6 @@ static struct snd_soc_dai_driver mc13783_dai_sync[] = { static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { .probe = mc13783_probe, .remove = mc13783_remove, - .read = mc13783_read, - .write = mc13783_write, .controls = mc13783_control_list, .num_controls = ARRAY_SIZE(mc13783_control_list), .dapm_widgets = mc13783_dapm_widgets, diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 26118828782..185fa3bc305 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -342,6 +342,8 @@ static int ml26124_hw_params(struct snd_pcm_substream *substream, struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); int i = get_coeff(priv->mclk, params_rate(hw_params)); + if (i < 0) + return i; priv->substream = substream; priv->rate = params_rate(hw_params); diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index c91eba504f9..73f9c3630e2 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -21,6 +21,7 @@ #include <linux/gpio.h> #include <linux/i2c.h> #include <linux/regmap.h> +#include <linux/of.h> #include <linux/of_device.h> #include <linux/of_gpio.h> #include <sound/pcm.h> diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 7613181123f..7146653a8e1 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -28,6 +28,7 @@ #include <sound/initval.h> #include <sound/soc.h> #include <sound/tlv.h> +#include <linux/of.h> #include <linux/of_device.h> #include "pcm1792a.h" diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index c26a8f814b1..a3fb4117963 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -21,6 +21,7 @@ #include <linux/of_gpio.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> +#include <linux/acpi.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -926,7 +927,7 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, return 0; } -void hp_amp_power_on(struct snd_soc_codec *codec) +static void hp_amp_power_on(struct snd_soc_codec *codec) { struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); @@ -1603,13 +1604,14 @@ static int rt5640_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - unsigned int val_len = 0, val_clk, mask_clk, dai_sel; - int pre_div, bclk_ms, frame_size; + unsigned int val_len = 0, val_clk, mask_clk; + int dai_sel, pre_div, bclk_ms, frame_size; rt5640->lrck[dai->id] = params_rate(params); pre_div = get_clk_info(rt5640->sysclk, rt5640->lrck[dai->id]); if (pre_div < 0) { - dev_err(codec->dev, "Unsupported clock setting\n"); + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5640->lrck[dai->id], dai->id); return -EINVAL; } frame_size = snd_soc_params_to_frame_size(params); @@ -1673,7 +1675,8 @@ static int rt5640_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct snd_soc_codec *codec = dai->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - unsigned int reg_val = 0, dai_sel; + unsigned int reg_val = 0; + int dai_sel; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: @@ -1977,13 +1980,20 @@ static int rt5640_suspend(struct snd_soc_codec *codec) rt5640_reset(codec); regcache_cache_only(rt5640->regmap, true); regcache_mark_dirty(rt5640->regmap); + if (gpio_is_valid(rt5640->pdata.ldo1_en)) + gpio_set_value_cansleep(rt5640->pdata.ldo1_en, 0); return 0; } static int rt5640_resume(struct snd_soc_codec *codec) { - rt5640_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(rt5640->pdata.ldo1_en)) { + gpio_set_value_cansleep(rt5640->pdata.ldo1_en, 1); + msleep(400); + } return 0; } @@ -2080,6 +2090,14 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +#ifdef CONFIG_ACPI +static struct acpi_device_id rt5640_acpi_match[] = { + { "INT33CA", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); +#endif + static int rt5640_parse_dt(struct rt5640_priv *rt5640, struct device_node *np) { rt5640->pdata.in1_diff = of_property_read_bool(np, @@ -2199,6 +2217,7 @@ static struct i2c_driver rt5640_i2c_driver = { .driver = { .name = "rt5640", .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5640_acpi_match), }, .probe = rt5640_i2c_probe, .remove = rt5640_i2c_remove, diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 38f3b105c17..52e7cb08434 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -60,48 +60,6 @@ enum si476x_pcm_format { SI476X_PCM_FORMAT_S24_LE = 6, }; -static unsigned int si476x_codec_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - int err; - unsigned int val; - struct si476x_core *core = codec->control_data; - - si476x_core_lock(core); - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, true); - - err = regmap_read(core->regmap, reg, &val); - - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, false); - si476x_core_unlock(core); - - if (err < 0) - return err; - - return val; -} - -static int si476x_codec_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int val) -{ - int err; - struct si476x_core *core = codec->control_data; - - si476x_core_lock(core); - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, true); - - err = regmap_write(core->regmap, reg, val); - - if (!si476x_core_is_powered_up(core)) - regcache_cache_only(core->regmap, false); - si476x_core_unlock(core); - - return err; -} - static const struct snd_soc_dapm_widget si476x_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("LOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), @@ -115,6 +73,7 @@ static const struct snd_soc_dapm_route si476x_dapm_routes[] = { static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { + struct si476x_core *core = i2c_mfd_cell_to_core(codec_dai->dev); int err; u16 format = 0; @@ -178,9 +137,14 @@ static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } + si476x_core_lock(core); + err = snd_soc_update_bits(codec_dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT, SI476X_DIGITAL_IO_OUTPUT_FORMAT_MASK, format); + + si476x_core_unlock(core); + if (err < 0) { dev_err(codec_dai->codec->dev, "Failed to set output format\n"); return err; @@ -193,6 +157,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct si476x_core *core = i2c_mfd_cell_to_core(dai->dev); int rate, width, err; rate = params_rate(params); @@ -218,11 +183,13 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + si476x_core_lock(core); + err = snd_soc_write(dai->codec, SI476X_DIGITAL_IO_OUTPUT_SAMPLE_RATE, rate); if (err < 0) { dev_err(dai->codec->dev, "Failed to set sample rate\n"); - return err; + goto out; } err = snd_soc_update_bits(dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT, @@ -231,15 +198,18 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, (width << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT)); if (err < 0) { dev_err(dai->codec->dev, "Failed to set output width\n"); - return err; + goto out; } - return 0; +out: + si476x_core_unlock(core); + + return err; } static int si476x_codec_probe(struct snd_soc_codec *codec) { - codec->control_data = i2c_mfd_cell_to_core(codec->dev); + codec->control_data = dev_get_regmap(codec->dev->parent, NULL); return 0; } @@ -268,8 +238,6 @@ static struct snd_soc_dai_driver si476x_dai = { static struct snd_soc_codec_driver soc_codec_dev_si476x = { .probe = si476x_codec_probe, - .read = si476x_codec_read, - .write = si476x_codec_write, .dapm_widgets = si476x_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(si476x_dapm_widgets), .dapm_routes = si476x_dapm_routes, diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index dba26e63844..13045f2af4d 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -164,30 +164,28 @@ static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec) } /*end - adc helper functions */ -static inline unsigned int sn95031_read(struct snd_soc_codec *codec, - unsigned int reg) +static int sn95031_read(void *ctx, unsigned int reg, unsigned int *val) { u8 value = 0; int ret; ret = intel_scu_ipc_ioread8(reg, &value); - if (ret) - pr_err("read of %x failed, err %d\n", reg, ret); - return value; + if (ret == 0) + *val = value; + return ret; } -static inline int sn95031_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) +static int sn95031_write(void *ctx, unsigned int reg, unsigned int value) { - int ret; - - ret = intel_scu_ipc_iowrite8(reg, value); - if (ret) - pr_err("write of %x failed, err %d\n", reg, ret); - return ret; + return intel_scu_ipc_iowrite8(reg, value); } +static const struct regmap_config sn95031_regmap = { + .reg_read = sn95031_read, + .reg_write = sn95031_write, +}; + static int sn95031_set_vaud_bias(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -827,6 +825,8 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); + snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots * for max 4 dais (2 stereo and 2 mono) @@ -886,8 +886,6 @@ static int sn95031_codec_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver sn95031_codec = { .probe = sn95031_codec_probe, .remove = sn95031_codec_remove, - .read = sn95031_read, - .write = sn95031_write, .set_bias_level = sn95031_set_vaud_bias, .idle_bias_off = true, .dapm_widgets = sn95031_dapm_widgets, @@ -898,7 +896,14 @@ static struct snd_soc_codec_driver sn95031_codec = { static int sn95031_device_probe(struct platform_device *pdev) { + struct regmap *regmap; + pr_debug("codec device probe called for %s\n", dev_name(&pdev->dev)); + + regmap = devm_regmap_init(&pdev->dev, NULL, NULL, &sn95031_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + return snd_soc_register_codec(&pdev->dev, &sn95031_codec, sn95031_dais, ARRAY_SIZE(sn95031_dais)); } diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 6d31d88f720..fe4d29d8856 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -37,6 +37,7 @@ #include <linux/i2c.h> #include <linux/regmap.h> #include <linux/spi/spi.h> +#include <linux/of.h> #include <linux/of_device.h> #include <linux/of_gpio.h> #include <sound/pcm.h> @@ -244,6 +245,8 @@ struct tas5086_private { unsigned int mclk, sclk; unsigned int format; bool deemph; + unsigned int charge_period; + unsigned int pwm_start_mid_z; /* Current sample rate for de-emphasis control */ int rate; /* GPIO driving Reset pin, if any */ @@ -456,6 +459,75 @@ static int tas5086_mute_stream(struct snd_soc_dai *dai, int mute, int stream) return regmap_write(priv->regmap, TAS5086_SOFT_MUTE, val); } +static void tas5086_reset(struct tas5086_private *priv) +{ + if (gpio_is_valid(priv->gpio_nreset)) { + /* Reset codec - minimum assertion time is 400ns */ + gpio_direction_output(priv->gpio_nreset, 0); + udelay(1); + gpio_set_value(priv->gpio_nreset, 1); + + /* Codec needs ~15ms to wake up */ + msleep(15); + } +} + +/* charge period values in microseconds */ +static const int tas5086_charge_period[] = { + 13000, 16900, 23400, 31200, 41600, 54600, 72800, 96200, + 130000, 156000, 234000, 312000, 416000, 546000, 728000, 962000, + 1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000, +}; + +static int tas5086_init(struct device *dev, struct tas5086_private *priv) +{ + int ret, i; + + /* + * If any of the channels is configured to start in Mid-Z mode, + * configure 'part 1' of the PWM starts to use Mid-Z, and tell + * all configured mid-z channels to start start under 'part 1'. + */ + if (priv->pwm_start_mid_z) + regmap_write(priv->regmap, TAS5086_PWM_START, + TAS5086_PWM_START_MIDZ_FOR_START_1 | + priv->pwm_start_mid_z); + + /* lookup and set split-capacitor charge period */ + if (priv->charge_period == 0) { + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); + } else { + i = index_in_array(tas5086_charge_period, + ARRAY_SIZE(tas5086_charge_period), + priv->charge_period); + if (i >= 0) + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, + i + 0x08); + else + dev_warn(dev, + "Invalid split-cap charge period of %d ns.\n", + priv->charge_period); + } + + /* enable factory trim */ + ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00); + if (ret < 0) + return ret; + + /* start all channels */ + ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20); + if (ret < 0) + return ret; + + /* mute all channels for now */ + ret = regmap_write(priv->regmap, TAS5086_SOFT_MUTE, + TAS5086_SOFT_MUTE_ALL); + if (ret < 0) + return ret; + + return 0; +} + /* TAS5086 controls */ static const DECLARE_TLV_DB_SCALE(tas5086_dac_tlv, -10350, 50, 1); @@ -691,14 +763,39 @@ static struct snd_soc_dai_driver tas5086_dai = { }; #ifdef CONFIG_PM +static int tas5086_soc_suspend(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + /* Shut down all channels */ + ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x60); + if (ret < 0) + return ret; + + return 0; +} + static int tas5086_soc_resume(struct snd_soc_codec *codec) { struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + tas5086_reset(priv); + regcache_mark_dirty(priv->regmap); + + ret = tas5086_init(codec->dev, priv); + if (ret < 0) + return ret; + + ret = regcache_sync(priv->regmap); + if (ret < 0) + return ret; - /* Restore codec state */ - return regcache_sync(priv->regmap); + return 0; } #else +#define tas5086_soc_suspend NULL #define tas5086_soc_resume NULL #endif /* CONFIG_PM */ @@ -710,23 +807,19 @@ static const struct of_device_id tas5086_dt_ids[] = { MODULE_DEVICE_TABLE(of, tas5086_dt_ids); #endif -/* charge period values in microseconds */ -static const int tas5086_charge_period[] = { - 13000, 16900, 23400, 31200, 41600, 54600, 72800, 96200, - 130000, 156000, 234000, 312000, 416000, 546000, 728000, 962000, - 1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000, -}; - static int tas5086_probe(struct snd_soc_codec *codec) { struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); - int charge_period = 1300000; /* hardware default is 1300 ms */ - u8 pwm_start_mid_z = 0; int i, ret; + priv->pwm_start_mid_z = 0; + priv->charge_period = 1300000; /* hardware default is 1300 ms */ + if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { struct device_node *of_node = codec->dev->of_node; - of_property_read_u32(of_node, "ti,charge-period", &charge_period); + + of_property_read_u32(of_node, "ti,charge-period", + &priv->charge_period); for (i = 0; i < 6; i++) { char name[25]; @@ -735,43 +828,11 @@ static int tas5086_probe(struct snd_soc_codec *codec) "ti,mid-z-channel-%d", i + 1); if (of_get_property(of_node, name, NULL) != NULL) - pwm_start_mid_z |= 1 << i; + priv->pwm_start_mid_z |= 1 << i; } } - /* - * If any of the channels is configured to start in Mid-Z mode, - * configure 'part 1' of the PWM starts to use Mid-Z, and tell - * all configured mid-z channels to start start under 'part 1'. - */ - if (pwm_start_mid_z) - regmap_write(priv->regmap, TAS5086_PWM_START, - TAS5086_PWM_START_MIDZ_FOR_START_1 | - pwm_start_mid_z); - - /* lookup and set split-capacitor charge period */ - if (charge_period == 0) { - regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); - } else { - i = index_in_array(tas5086_charge_period, - ARRAY_SIZE(tas5086_charge_period), - charge_period); - if (i >= 0) - regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, - i + 0x08); - else - dev_warn(codec->dev, - "Invalid split-cap charge period of %d ns.\n", - charge_period); - } - - /* enable factory trim */ - ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00); - if (ret < 0) - return ret; - - /* start all channels */ - ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20); + ret = tas5086_init(codec->dev, priv); if (ret < 0) return ret; @@ -780,12 +841,6 @@ static int tas5086_probe(struct snd_soc_codec *codec) if (ret < 0) return ret; - /* mute all channels for now */ - ret = regmap_write(priv->regmap, TAS5086_SOFT_MUTE, - TAS5086_SOFT_MUTE_ALL); - if (ret < 0) - return ret; - return 0; } @@ -803,6 +858,7 @@ static int tas5086_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_tas5086 = { .probe = tas5086_probe, .remove = tas5086_remove, + .suspend = tas5086_soc_suspend, .resume = tas5086_soc_resume, .controls = tas5086_controls, .num_controls = ARRAY_SIZE(tas5086_controls), @@ -862,17 +918,8 @@ static int tas5086_i2c_probe(struct i2c_client *i2c, if (devm_gpio_request(dev, gpio_nreset, "TAS5086 Reset")) gpio_nreset = -EINVAL; - if (gpio_is_valid(gpio_nreset)) { - /* Reset codec - minimum assertion time is 400ns */ - gpio_direction_output(gpio_nreset, 0); - udelay(1); - gpio_set_value(gpio_nreset, 1); - - /* Codec needs ~15ms to wake up */ - msleep(15); - } - priv->gpio_nreset = gpio_nreset; + tas5086_reset(priv); /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */ ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 31762ebdd77..5d430cc56f5 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -24,6 +24,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -37,11 +38,27 @@ /* * AIC23 register cache */ -static const u16 tlv320aic23_reg[] = { - 0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */ - 0x001A, 0x0004, 0x0007, 0x0001, /* 4 */ - 0x0020, 0x0000, 0x0000, 0x0000, /* 8 */ - 0x0000, 0x0000, 0x0000, 0x0000, /* 12 */ +static const struct reg_default tlv320aic23_reg[] = { + { 0, 0x0097 }, + { 1, 0x0097 }, + { 2, 0x00F9 }, + { 3, 0x00F9 }, + { 4, 0x001A }, + { 5, 0x0004 }, + { 6, 0x0007 }, + { 7, 0x0001 }, + { 8, 0x0020 }, + { 9, 0x0000 }, +}; + +static const struct regmap_config tlv320aic23_regmap = { + .reg_bits = 7, + .val_bits = 9, + + .max_register = TLV320AIC23_RESET, + .reg_defaults = tlv320aic23_reg, + .num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg), + .cache_type = REGCACHE_RBTREE, }; static const char *rec_src_text[] = { "Line", "Mic" }; @@ -171,7 +188,7 @@ static const struct snd_soc_dapm_route tlv320aic23_intercon[] = { /* AIC23 driver data */ struct aic23 { - enum snd_soc_control_type control_type; + struct regmap *regmap; int mclk; int requested_adc; int requested_dac; @@ -532,7 +549,9 @@ static int tlv320aic23_suspend(struct snd_soc_codec *codec) static int tlv320aic23_resume(struct snd_soc_codec *codec) { - snd_soc_cache_sync(codec); + struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); + regcache_mark_dirty(aic23->regmap); + regcache_sync(aic23->regmap); tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; @@ -540,10 +559,9 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec) static int tlv320aic23_probe(struct snd_soc_codec *codec) { - struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type); + ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -552,16 +570,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) /* Reset codec */ snd_soc_write(codec, TLV320AIC23_RESET, 0); - /* Write the register default value to cache for reserved registers, - * so the write to the these registers are suppressed by the cache - * restore code when it skips writes of default registers. - */ - snd_soc_cache_write(codec, 0x0A, 0); - snd_soc_cache_write(codec, 0x0B, 0); - snd_soc_cache_write(codec, 0x0C, 0); - snd_soc_cache_write(codec, 0x0D, 0); - snd_soc_cache_write(codec, 0x0E, 0); - /* power on device */ tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -586,9 +594,6 @@ static int tlv320aic23_probe(struct snd_soc_codec *codec) snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1); - snd_soc_add_codec_controls(codec, tlv320aic23_snd_controls, - ARRAY_SIZE(tlv320aic23_snd_controls)); - return 0; } @@ -599,21 +604,19 @@ static int tlv320aic23_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { - .reg_cache_size = ARRAY_SIZE(tlv320aic23_reg), - .reg_word_size = sizeof(u16), - .reg_cache_default = tlv320aic23_reg, .probe = tlv320aic23_probe, .remove = tlv320aic23_remove, .suspend = tlv320aic23_suspend, .resume = tlv320aic23_resume, .set_bias_level = tlv320aic23_set_bias_level, + .controls = tlv320aic23_snd_controls, + .num_controls = ARRAY_SIZE(tlv320aic23_snd_controls), .dapm_widgets = tlv320aic23_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), .dapm_routes = tlv320aic23_intercon, .num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon), }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * If the i2c layer weren't so broken, we could pass this kind of data * around @@ -631,8 +634,11 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c, if (aic23 == NULL) return -ENOMEM; + aic23->regmap = devm_regmap_init_i2c(i2c, &tlv320aic23_regmap); + if (IS_ERR(aic23->regmap)) + return PTR_ERR(aic23->regmap); + i2c_set_clientdata(i2c, aic23); - aic23->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1); @@ -660,29 +666,7 @@ static struct i2c_driver tlv320aic23_i2c_driver = { .id_table = tlv320aic23_id, }; -#endif - -static int __init tlv320aic23_modinit(void) -{ - int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - ret = i2c_add_driver(&tlv320aic23_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register TLV320AIC23 I2C driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(tlv320aic23_modinit); - -static void __exit tlv320aic23_exit(void) -{ -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_del_driver(&tlv320aic23_i2c_driver); -#endif -} -module_exit(tlv320aic23_exit); +module_i2c_driver(tlv320aic23_i2c_driver); MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 7b8f3d965f4..94a658fa6d9 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -29,6 +29,7 @@ MODULE_LICENSE("GPL"); /* AIC26 driver private data */ struct aic26 { struct spi_device *spi; + struct regmap *regmap; struct snd_soc_codec *codec; int master; int datfm; @@ -40,85 +41,6 @@ struct aic26 { int keyclick_len; }; -/* --------------------------------------------------------------------- - * Register access routines - */ -static unsigned int aic26_reg_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - u16 cmd, value; - u8 buffer[2]; - int rc; - - if (reg >= AIC26_NUM_REGS) { - WARN_ON_ONCE(1); - return 0; - } - - /* Do SPI transfer; first 16bits are command; remaining is - * register contents */ - cmd = AIC26_READ_COMMAND_WORD(reg); - buffer[0] = (cmd >> 8) & 0xff; - buffer[1] = cmd & 0xff; - rc = spi_write_then_read(aic26->spi, buffer, 2, buffer, 2); - if (rc) { - dev_err(&aic26->spi->dev, "AIC26 reg read error\n"); - return -EIO; - } - value = (buffer[0] << 8) | buffer[1]; - - /* Update the cache before returning with the value */ - cache[reg] = value; - return value; -} - -static unsigned int aic26_reg_read_cache(struct snd_soc_codec *codec, - unsigned int reg) -{ - u16 *cache = codec->reg_cache; - - if (reg >= AIC26_NUM_REGS) { - WARN_ON_ONCE(1); - return 0; - } - - return cache[reg]; -} - -static int aic26_reg_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); - u16 *cache = codec->reg_cache; - u16 cmd; - u8 buffer[4]; - int rc; - - if (reg >= AIC26_NUM_REGS) { - WARN_ON_ONCE(1); - return -EINVAL; - } - - /* Do SPI transfer; first 16bits are command; remaining is data - * to write into register */ - cmd = AIC26_WRITE_COMMAND_WORD(reg); - buffer[0] = (cmd >> 8) & 0xff; - buffer[1] = cmd & 0xff; - buffer[2] = value >> 8; - buffer[3] = value; - rc = spi_write(aic26->spi, buffer, 4); - if (rc) { - dev_err(&aic26->spi->dev, "AIC26 reg read error\n"); - return -EIO; - } - - /* update cache before returning */ - cache[reg] = value; - return 0; -} - static const struct snd_soc_dapm_widget tlv320aic26_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MICIN"), SND_SOC_DAPM_INPUT("AUX"), @@ -195,19 +117,15 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, snd_soc_write(codec, AIC26_REG_PLL_PROG2, reg); /* Audio Control 3 (master mode, fsref rate) */ - reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL3); - reg &= ~0xf800; if (aic26->master) - reg |= 0x0800; + reg = 0x0800; if (fsref == 48000) - reg |= 0x2000; - snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); + reg = 0x2000; + snd_soc_update_bits(codec, AIC26_REG_AUDIO_CTRL3, 0xf800, reg); /* Audio Control 1 (FSref divisor) */ - reg = aic26_reg_read_cache(codec, AIC26_REG_AUDIO_CTRL1); - reg &= ~0x0fff; - reg |= wlen | aic26->datfm | (divisor << 3) | divisor; - snd_soc_write(codec, AIC26_REG_AUDIO_CTRL1, reg); + reg = wlen | aic26->datfm | (divisor << 3) | divisor; + snd_soc_update_bits(codec, AIC26_REG_AUDIO_CTRL1, 0xfff, reg); return 0; } @@ -219,16 +137,16 @@ static int aic26_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; struct aic26 *aic26 = snd_soc_codec_get_drvdata(codec); - u16 reg = aic26_reg_read_cache(codec, AIC26_REG_DAC_GAIN); + u16 reg; dev_dbg(&aic26->spi->dev, "aic26_mute(dai=%p, mute=%i)\n", dai, mute); if (mute) - reg |= 0x8080; + reg = 0x8080; else - reg &= ~0x8080; - snd_soc_write(codec, AIC26_REG_DAC_GAIN, reg); + reg = 0; + snd_soc_update_bits(codec, AIC26_REG_DAC_GAIN, 0x8000, reg); return 0; } @@ -346,7 +264,7 @@ static ssize_t aic26_keyclick_show(struct device *dev, struct aic26 *aic26 = dev_get_drvdata(dev); int val, amp, freq, len; - val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); + val = snd_soc_read(aic26->codec, AIC26_REG_AUDIO_CTRL2); amp = (val >> 12) & 0x7; freq = (125 << ((val >> 8) & 0x7)) >> 1; len = 2 * (1 + ((val >> 4) & 0xf)); @@ -360,11 +278,9 @@ static ssize_t aic26_keyclick_set(struct device *dev, const char *buf, size_t count) { struct aic26 *aic26 = dev_get_drvdata(dev); - int val; - val = aic26_reg_read_cache(aic26->codec, AIC26_REG_AUDIO_CTRL2); - val |= 0x8000; - snd_soc_write(aic26->codec, AIC26_REG_AUDIO_CTRL2, val); + snd_soc_update_bits(aic26->codec, AIC26_REG_AUDIO_CTRL2, + 0x8000, 0x800); return count; } @@ -377,7 +293,9 @@ static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); static int aic26_probe(struct snd_soc_codec *codec) { struct aic26 *aic26 = dev_get_drvdata(codec->dev); - int ret, err, i, reg; + int ret, reg; + + snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); aic26->codec = codec; @@ -393,37 +311,30 @@ static int aic26_probe(struct snd_soc_codec *codec) reg |= 0x0800; /* set master mode */ snd_soc_write(codec, AIC26_REG_AUDIO_CTRL3, reg); - /* Fill register cache */ - for (i = 0; i < codec->driver->reg_cache_size; i++) - snd_soc_read(codec, i); - /* Register the sysfs files for debugging */ /* Create SysFS files */ ret = device_create_file(codec->dev, &dev_attr_keyclick); if (ret) dev_info(codec->dev, "error creating sysfs files\n"); - /* register controls */ - dev_dbg(codec->dev, "Registering controls\n"); - err = snd_soc_add_codec_controls(codec, aic26_snd_controls, - ARRAY_SIZE(aic26_snd_controls)); - WARN_ON(err < 0); - return 0; } static struct snd_soc_codec_driver aic26_soc_codec_dev = { .probe = aic26_probe, - .read = aic26_reg_read, - .write = aic26_reg_write, - .reg_cache_size = AIC26_NUM_REGS, - .reg_word_size = sizeof(u16), + .controls = aic26_snd_controls, + .num_controls = ARRAY_SIZE(aic26_snd_controls), .dapm_widgets = tlv320aic26_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tlv320aic26_dapm_widgets), .dapm_routes = tlv320aic26_dapm_routes, .num_dapm_routes = ARRAY_SIZE(tlv320aic26_dapm_routes), }; +static const struct regmap_config aic26_regmap = { + .reg_bits = 16, + .val_bits = 16, +}; + /* --------------------------------------------------------------------- * SPI device portion of driver: probe and release routines and SPI * driver registration. @@ -440,6 +351,10 @@ static int aic26_spi_probe(struct spi_device *spi) if (!aic26) return -ENOMEM; + aic26->regmap = devm_regmap_init_spi(spi, &aic26_regmap); + if (IS_ERR(aic26->regmap)) + return PTR_ERR(aic26->regmap); + /* Initialize the driver data */ aic26->spi = spi; dev_set_drvdata(&spi->dev, aic26); diff --git a/sound/soc/codecs/tlv320aic26.h b/sound/soc/codecs/tlv320aic26.h index 67f19c3bebe..629b85e7540 100644 --- a/sound/soc/codecs/tlv320aic26.h +++ b/sound/soc/codecs/tlv320aic26.h @@ -9,10 +9,7 @@ #define _TLV320AIC16_H_ /* AIC26 Registers */ -#define AIC26_READ_COMMAND_WORD(addr) ((1 << 15) | (addr << 5)) -#define AIC26_WRITE_COMMAND_WORD(addr) ((0 << 15) | (addr << 5)) -#define AIC26_PAGE_ADDR(page, offset) ((page << 6) | offset) -#define AIC26_NUM_REGS AIC26_PAGE_ADDR(3, 0) +#define AIC26_PAGE_ADDR(page, offset) ((page << 11) | offset << 5) /* Page 0: Auxiliary data registers */ #define AIC26_REG_BAT1 AIC26_PAGE_ADDR(0, 0x05) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 2ed57d4aa44..18cdcca9014 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -60,9 +60,8 @@ struct aic32x4_rate_divs { }; struct aic32x4_priv { + struct regmap *regmap; u32 sysclk; - u8 page_no; - void *control_data; u32 power_cfg; u32 micpga_routing; bool swapdacs; @@ -262,67 +261,25 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { {"Right ADC", NULL, "Right Input Mixer"}, }; -static inline int aic32x4_change_page(struct snd_soc_codec *codec, - unsigned int new_page) -{ - struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - u8 data[2]; - int ret; - - data[0] = 0x00; - data[1] = new_page & 0xff; - - ret = codec->hw_write(codec->control_data, data, 2); - if (ret == 2) { - aic32x4->page_no = new_page; - return 0; - } else { - return ret; - } -} - -static int aic32x4_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int val) -{ - struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - unsigned int page = reg / 128; - unsigned int fixed_reg = reg % 128; - u8 data[2]; - int ret; - - /* A write to AIC32X4_PSEL is really a non-explicit page change */ - if (reg == AIC32X4_PSEL) - return aic32x4_change_page(codec, val); - - if (aic32x4->page_no != page) { - ret = aic32x4_change_page(codec, page); - if (ret != 0) - return ret; - } - - data[0] = fixed_reg & 0xff; - data[1] = val & 0xff; - - if (codec->hw_write(codec->control_data, data, 2) == 2) - return 0; - else - return -EIO; -} +static const struct regmap_range_cfg aic32x4_regmap_pages[] = { + { + .selector_reg = 0, + .selector_mask = 0xff, + .window_start = 0, + .window_len = 128, + .range_min = AIC32X4_PAGE1, + .range_max = AIC32X4_PAGE1 + 127, + }, +}; -static unsigned int aic32x4_read(struct snd_soc_codec *codec, unsigned int reg) -{ - struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); - unsigned int page = reg / 128; - unsigned int fixed_reg = reg % 128; - int ret; +static const struct regmap_config aic32x4_regmap = { + .reg_bits = 8, + .val_bits = 8, - if (aic32x4->page_no != page) { - ret = aic32x4_change_page(codec, page); - if (ret != 0) - return ret; - } - return i2c_smbus_read_byte_data(codec->control_data, fixed_reg & 0xff); -} + .max_register = AIC32X4_RMICPGAVOL, + .ranges = aic32x4_regmap_pages, + .num_ranges = ARRAY_SIZE(aic32x4_regmap_pages), +}; static inline int aic32x4_get_divs(int mclk, int rate) { @@ -617,16 +574,10 @@ static int aic32x4_probe(struct snd_soc_codec *codec) { struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u32 tmp_reg; - int ret; - codec->hw_write = (hw_write_t) i2c_master_send; - codec->control_data = aic32x4->control_data; + snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (aic32x4->rstn_gpio >= 0) { - ret = devm_gpio_request_one(codec->dev, aic32x4->rstn_gpio, - GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); - if (ret != 0) - return ret; ndelay(10); gpio_set_value(aic32x4->rstn_gpio, 1); } @@ -692,8 +643,6 @@ static int aic32x4_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { - .read = aic32x4_read, - .write = aic32x4_write, .probe = aic32x4_probe, .remove = aic32x4_remove, .suspend = aic32x4_suspend, @@ -720,7 +669,10 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, if (aic32x4 == NULL) return -ENOMEM; - aic32x4->control_data = i2c; + aic32x4->regmap = devm_regmap_init_i2c(i2c, &aic32x4_regmap); + if (IS_ERR(aic32x4->regmap)) + return PTR_ERR(aic32x4->regmap); + i2c_set_clientdata(i2c, aic32x4); if (pdata) { @@ -735,6 +687,13 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->rstn_gpio = -1; } + if (aic32x4->rstn_gpio >= 0) { + ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio, + GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); + if (ret != 0) + return ret; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic32x4, &aic32x4_dai, 1); return ret; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 64ad84d8a30..546d16b7d38 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -40,6 +40,7 @@ #include <linux/i2c.h> #include <linux/gpio.h> #include <linux/regulator/consumer.h> +#include <linux/of.h> #include <linux/of_gpio.h> #include <linux/slab.h> #include <sound/core.h> @@ -72,9 +73,9 @@ struct aic3x_disable_nb { /* codec private data */ struct aic3x_priv { struct snd_soc_codec *codec; + struct regmap *regmap; struct regulator_bulk_data supplies[AIC3X_NUM_SUPPLIES]; struct aic3x_disable_nb disable_nb[AIC3X_NUM_SUPPLIES]; - enum snd_soc_control_type control_type; struct aic3x_setup_data *setup; unsigned int sysclk; struct list_head list; @@ -90,41 +91,45 @@ struct aic3x_priv { enum aic3x_micbias_voltage micbias_vg; }; -/* - * AIC3X register cache - * We can't read the AIC3X register space when we are - * using 2 wire for device control, so we cache them instead. - * There is no point in caching the reset register - */ -static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { - 0x00, 0x00, 0x00, 0x10, /* 0 */ - 0x04, 0x00, 0x00, 0x00, /* 4 */ - 0x00, 0x00, 0x00, 0x01, /* 8 */ - 0x00, 0x00, 0x00, 0x80, /* 12 */ - 0x80, 0xff, 0xff, 0x78, /* 16 */ - 0x78, 0x78, 0x78, 0x78, /* 20 */ - 0x78, 0x00, 0x00, 0xfe, /* 24 */ - 0x00, 0x00, 0xfe, 0x00, /* 28 */ - 0x18, 0x18, 0x00, 0x00, /* 32 */ - 0x00, 0x00, 0x00, 0x00, /* 36 */ - 0x00, 0x00, 0x00, 0x80, /* 40 */ - 0x80, 0x00, 0x00, 0x00, /* 44 */ - 0x00, 0x00, 0x00, 0x04, /* 48 */ - 0x00, 0x00, 0x00, 0x00, /* 52 */ - 0x00, 0x00, 0x04, 0x00, /* 56 */ - 0x00, 0x00, 0x00, 0x00, /* 60 */ - 0x00, 0x04, 0x00, 0x00, /* 64 */ - 0x00, 0x00, 0x00, 0x00, /* 68 */ - 0x04, 0x00, 0x00, 0x00, /* 72 */ - 0x00, 0x00, 0x00, 0x00, /* 76 */ - 0x00, 0x00, 0x00, 0x00, /* 80 */ - 0x00, 0x00, 0x00, 0x00, /* 84 */ - 0x00, 0x00, 0x00, 0x00, /* 88 */ - 0x00, 0x00, 0x00, 0x00, /* 92 */ - 0x00, 0x00, 0x00, 0x00, /* 96 */ - 0x00, 0x00, 0x02, 0x00, /* 100 */ - 0x00, 0x00, 0x00, 0x00, /* 104 */ - 0x00, 0x00, /* 108 */ +static const struct reg_default aic3x_reg[] = { + { 0, 0x00 }, { 1, 0x00 }, { 2, 0x00 }, { 3, 0x10 }, + { 4, 0x04 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 }, + { 8, 0x00 }, { 9, 0x00 }, { 10, 0x00 }, { 11, 0x01 }, + { 12, 0x00 }, { 13, 0x00 }, { 14, 0x00 }, { 15, 0x80 }, + { 16, 0x80 }, { 17, 0xff }, { 18, 0xff }, { 19, 0x78 }, + { 20, 0x78 }, { 21, 0x78 }, { 22, 0x78 }, { 23, 0x78 }, + { 24, 0x78 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0xfe }, + { 28, 0x00 }, { 29, 0x00 }, { 30, 0xfe }, { 31, 0x00 }, + { 32, 0x18 }, { 33, 0x18 }, { 34, 0x00 }, { 35, 0x00 }, + { 36, 0x00 }, { 37, 0x00 }, { 38, 0x00 }, { 39, 0x00 }, + { 40, 0x00 }, { 41, 0x00 }, { 42, 0x00 }, { 43, 0x80 }, + { 44, 0x80 }, { 45, 0x00 }, { 46, 0x00 }, { 47, 0x00 }, + { 48, 0x00 }, { 49, 0x00 }, { 50, 0x00 }, { 51, 0x04 }, + { 52, 0x00 }, { 53, 0x00 }, { 54, 0x00 }, { 55, 0x00 }, + { 56, 0x00 }, { 57, 0x00 }, { 58, 0x04 }, { 59, 0x00 }, + { 60, 0x00 }, { 61, 0x00 }, { 62, 0x00 }, { 63, 0x00 }, + { 64, 0x00 }, { 65, 0x04 }, { 66, 0x00 }, { 67, 0x00 }, + { 68, 0x00 }, { 69, 0x00 }, { 70, 0x00 }, { 71, 0x00 }, + { 72, 0x04 }, { 73, 0x00 }, { 74, 0x00 }, { 75, 0x00 }, + { 76, 0x00 }, { 77, 0x00 }, { 78, 0x00 }, { 79, 0x00 }, + { 80, 0x00 }, { 81, 0x00 }, { 82, 0x00 }, { 83, 0x00 }, + { 84, 0x00 }, { 85, 0x00 }, { 86, 0x00 }, { 87, 0x00 }, + { 88, 0x00 }, { 89, 0x00 }, { 90, 0x00 }, { 91, 0x00 }, + { 92, 0x00 }, { 93, 0x00 }, { 94, 0x00 }, { 95, 0x00 }, + { 96, 0x00 }, { 97, 0x00 }, { 98, 0x00 }, { 99, 0x00 }, + { 100, 0x00 }, { 101, 0x00 }, { 102, 0x02 }, { 103, 0x00 }, + { 104, 0x00 }, { 105, 0x00 }, { 106, 0x00 }, { 107, 0x00 }, + { 108, 0x00 }, { 109, 0x00 }, +}; + +static const struct regmap_config aic3x_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = DAC_ICC_ADJ, + .reg_defaults = aic3x_reg, + .num_reg_defaults = ARRAY_SIZE(aic3x_reg), + .cache_type = REGCACHE_RBTREE, }; #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ @@ -828,12 +833,6 @@ static int aic3x_add_widgets(struct snd_soc_codec *codec) struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, - ARRAY_SIZE(aic3x_dapm_widgets)); - - /* set up audio path interconnects */ - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - if (aic3x->model == AIC3X_MODEL_3007) { snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets, ARRAY_SIZE(aic3007_dapm_widgets)); @@ -1082,29 +1081,6 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } -static int aic3x_init_3007(struct snd_soc_codec *codec) -{ - u8 tmp1, tmp2, *cache = codec->reg_cache; - - /* - * There is no need to cache writes to undocumented page 0xD but - * respective page 0 register cache entries must be preserved - */ - tmp1 = cache[0xD]; - tmp2 = cache[0x8]; - /* Class-D speaker driver init; datasheet p. 46 */ - snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x0D); - snd_soc_write(codec, 0xD, 0x0D); - snd_soc_write(codec, 0x8, 0x5C); - snd_soc_write(codec, 0x8, 0x5D); - snd_soc_write(codec, 0x8, 0x5C); - snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x00); - cache[0xD] = tmp1; - cache[0x8] = tmp2; - - return 0; -} - static int aic3x_regulator_event(struct notifier_block *nb, unsigned long event, void *data) { @@ -1119,7 +1095,7 @@ static int aic3x_regulator_event(struct notifier_block *nb, */ if (gpio_is_valid(aic3x->gpio_reset)) gpio_set_value(aic3x->gpio_reset, 0); - aic3x->codec->cache_sync = 1; + regcache_mark_dirty(aic3x->regmap); } return 0; @@ -1128,8 +1104,7 @@ static int aic3x_regulator_event(struct notifier_block *nb, static int aic3x_set_power(struct snd_soc_codec *codec, int power) { struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); - int i, ret; - u8 *cache = codec->reg_cache; + int ret; if (power) { ret = regulator_bulk_enable(ARRAY_SIZE(aic3x->supplies), @@ -1137,12 +1112,6 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) if (ret) goto out; aic3x->power = 1; - /* - * Reset release and cache sync is necessary only if some - * supply was off or if there were cached writes - */ - if (!codec->cache_sync) - goto out; if (gpio_is_valid(aic3x->gpio_reset)) { udelay(1); @@ -1150,12 +1119,8 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) } /* Sync reg_cache with the hardware */ - codec->cache_only = 0; - for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++) - snd_soc_write(codec, i, cache[i]); - if (aic3x->model == AIC3X_MODEL_3007) - aic3x_init_3007(codec); - codec->cache_sync = 0; + regcache_cache_only(aic3x->regmap, false); + regcache_sync(aic3x->regmap); } else { /* * Do soft reset to this codec instance in order to clear @@ -1163,10 +1128,10 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) * remain on */ snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); - codec->cache_sync = 1; + regcache_mark_dirty(aic3x->regmap); aic3x->power = 0; /* HW writes are needless when bias is off */ - codec->cache_only = 1; + regcache_cache_only(aic3x->regmap, true); ret = regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); } @@ -1321,7 +1286,6 @@ static int aic3x_init(struct snd_soc_codec *codec) snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); if (aic3x->model == AIC3X_MODEL_3007) { - aic3x_init_3007(codec); snd_soc_write(codec, CLASSD_CTRL, 0); } @@ -1349,29 +1313,12 @@ static int aic3x_probe(struct snd_soc_codec *codec) INIT_LIST_HEAD(&aic3x->list); aic3x->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } - if (gpio_is_valid(aic3x->gpio_reset) && - !aic3x_is_shared_reset(aic3x)) { - ret = gpio_request(aic3x->gpio_reset, "tlv320aic3x reset"); - if (ret != 0) - goto err_gpio; - gpio_direction_output(aic3x->gpio_reset, 0); - } - - for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) - aic3x->supplies[i].supply = aic3x_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(aic3x->supplies), - aic3x->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - goto err_get; - } for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { aic3x->disable_nb[i].nb.notifier_call = aic3x_regulator_event; aic3x->disable_nb[i].aic3x = aic3x; @@ -1385,7 +1332,7 @@ static int aic3x_probe(struct snd_soc_codec *codec) } } - codec->cache_only = 1; + regcache_mark_dirty(aic3x->regmap); aic3x_init(codec); if (aic3x->setup) { @@ -1396,8 +1343,6 @@ static int aic3x_probe(struct snd_soc_codec *codec) (aic3x->setup->gpio_func[1] & 0xf) << 4); } - snd_soc_add_codec_controls(codec, aic3x_snd_controls, - ARRAY_SIZE(aic3x_snd_controls)); if (aic3x->model == AIC3X_MODEL_3007) snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); @@ -1428,12 +1373,6 @@ err_notif: while (i--) regulator_unregister_notifier(aic3x->supplies[i].consumer, &aic3x->disable_nb[i].nb); - regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); -err_get: - if (gpio_is_valid(aic3x->gpio_reset) && - !aic3x_is_shared_reset(aic3x)) - gpio_free(aic3x->gpio_reset); -err_gpio: return ret; } @@ -1444,15 +1383,9 @@ static int aic3x_remove(struct snd_soc_codec *codec) aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF); list_del(&aic3x->list); - if (gpio_is_valid(aic3x->gpio_reset) && - !aic3x_is_shared_reset(aic3x)) { - gpio_set_value(aic3x->gpio_reset, 0); - gpio_free(aic3x->gpio_reset); - } for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) regulator_unregister_notifier(aic3x->supplies[i].consumer, &aic3x->disable_nb[i].nb); - regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); return 0; } @@ -1460,13 +1393,16 @@ static int aic3x_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_aic3x = { .set_bias_level = aic3x_set_bias_level, .idle_bias_off = true, - .reg_cache_size = ARRAY_SIZE(aic3x_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = aic3x_reg, .probe = aic3x_probe, .remove = aic3x_remove, .suspend = aic3x_suspend, .resume = aic3x_resume, + .controls = aic3x_snd_controls, + .num_controls = ARRAY_SIZE(aic3x_snd_controls), + .dapm_widgets = aic3x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; /* @@ -1483,6 +1419,16 @@ static const struct i2c_device_id aic3x_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); +static const struct reg_default aic3007_class_d[] = { + /* Class-D speaker driver init; datasheet p. 46 */ + { AIC3X_PAGE_SELECT, 0x0D }, + { 0xD, 0x0D }, + { 0x8, 0x5C }, + { 0x8, 0x5D }, + { 0x8, 0x5C }, + { AIC3X_PAGE_SELECT, 0x00 }, +}; + /* * If the i2c layer weren't so broken, we could pass this kind of data * around @@ -1494,7 +1440,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, struct aic3x_priv *aic3x; struct aic3x_setup_data *ai3x_setup; struct device_node *np = i2c->dev.of_node; - int ret; + int ret, i; u32 value; aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL); @@ -1503,7 +1449,13 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, return -ENOMEM; } - aic3x->control_type = SND_SOC_I2C; + aic3x->regmap = devm_regmap_init_i2c(i2c, &aic3x_regmap); + if (IS_ERR(aic3x->regmap)) { + ret = PTR_ERR(aic3x->regmap); + return ret; + } + + regcache_cache_only(aic3x->regmap, true); i2c_set_clientdata(i2c, aic3x); if (pdata) { @@ -1555,14 +1507,54 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, aic3x->model = id->driver_data; + if (gpio_is_valid(aic3x->gpio_reset) && + !aic3x_is_shared_reset(aic3x)) { + ret = gpio_request(aic3x->gpio_reset, "tlv320aic3x reset"); + if (ret != 0) + goto err; + gpio_direction_output(aic3x->gpio_reset, 0); + } + + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) + aic3x->supplies[i].supply = aic3x_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(aic3x->supplies), + aic3x->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + goto err_gpio; + } + + if (aic3x->model == AIC3X_MODEL_3007) { + ret = regmap_register_patch(aic3x->regmap, aic3007_class_d, + ARRAY_SIZE(aic3007_class_d)); + if (ret != 0) + dev_err(&i2c->dev, "Failed to init class D: %d\n", + ret); + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic3x, &aic3x_dai, 1); return ret; + +err_gpio: + if (gpio_is_valid(aic3x->gpio_reset) && + !aic3x_is_shared_reset(aic3x)) + gpio_free(aic3x->gpio_reset); +err: + return ret; } static int aic3x_i2c_remove(struct i2c_client *client) { + struct aic3x_priv *aic3x = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); + if (gpio_is_valid(aic3x->gpio_reset) && + !aic3x_is_shared_reset(aic3x)) { + gpio_set_value(aic3x->gpio_reset, 0); + gpio_free(aic3x->gpio_reset); + } return 0; } diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 348552e1771..b27c396037d 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -30,6 +30,7 @@ #include <sound/tpa6130a2-plat.h> #include <sound/soc.h> #include <sound/tlv.h> +#include <linux/of_gpio.h> #include "tpa6130a2.h" @@ -371,30 +372,33 @@ static int tpa6130a2_probe(struct i2c_client *client, { struct device *dev; struct tpa6130a2_data *data; - struct tpa6130a2_platform_data *pdata; + struct tpa6130a2_platform_data *pdata = client->dev.platform_data; + struct device_node *np = client->dev.of_node; const char *regulator; int ret; dev = &client->dev; - if (client->dev.platform_data == NULL) { - dev_err(dev, "Platform data not set\n"); - dump_stack(); - return -ENODEV; - } - data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); if (data == NULL) { dev_err(dev, "Can not allocate memory\n"); return -ENOMEM; } + if (pdata) { + data->power_gpio = pdata->power_gpio; + } else if (np) { + data->power_gpio = of_get_named_gpio(np, "power-gpio", 0); + } else { + dev_err(dev, "Platform data not set\n"); + dump_stack(); + return -ENODEV; + } + tpa6130a2_client = client; i2c_set_clientdata(tpa6130a2_client, data); - pdata = client->dev.platform_data; - data->power_gpio = pdata->power_gpio; data->id = id->driver_data; mutex_init(&data->mutex); @@ -473,10 +477,20 @@ static const struct i2c_device_id tpa6130a2_id[] = { }; MODULE_DEVICE_TABLE(i2c, tpa6130a2_id); +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tpa6130a2_of_match[] = { + { .compatible = "ti,tpa6130a2", }, + { .compatible = "ti,tpa6140a2" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tpa6130a2_of_match); +#endif + static struct i2c_driver tpa6130a2_i2c_driver = { .driver = { .name = "tpa6130a2", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tpa6130a2_of_match), }, .probe = tpa6130a2_probe, .remove = tpa6130a2_remove, diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 1e3884d6b3f..dfc51bb425d 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -46,13 +46,7 @@ /* TWL4030 PMBR1 Register GPIO6 mux bits */ #define TWL4030_GPIO6_PWM0_MUTE(value) ((value & 0x03) << 2) -/* Shadow register used by the audio driver */ -#define TWL4030_REG_SW_SHADOW 0x4A -#define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1) - -/* TWL4030_REG_SW_SHADOW (0x4A) Fields */ -#define TWL4030_HFL_EN 0x01 -#define TWL4030_HFR_EN 0x02 +#define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1) /* * twl4030 register cache & default register settings @@ -132,7 +126,6 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_VIBRA_PWM_SET (0x47) */ 0x00, /* REG_ANAMIC_GAIN (0x48) */ 0x00, /* REG_MISC_SET_2 (0x49) */ - 0x00, /* REG_SW_SHADOW (0x4A) - Shadow, non HW register */ }; /* codec private data */ @@ -198,42 +191,41 @@ static int twl4030_write(struct snd_soc_codec *codec, int write_to_reg = 0; twl4030_write_reg_cache(codec, reg, value); - if (likely(reg < TWL4030_REG_SW_SHADOW)) { - /* Decide if the given register can be written */ - switch (reg) { - case TWL4030_REG_EAR_CTL: - if (twl4030->earpiece_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_PREDL_CTL: - if (twl4030->predrivel_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_PREDR_CTL: - if (twl4030->predriver_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_PRECKL_CTL: - if (twl4030->carkitl_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_PRECKR_CTL: - if (twl4030->carkitr_enabled) - write_to_reg = 1; - break; - case TWL4030_REG_HS_GAIN_SET: - if (twl4030->hsl_enabled || twl4030->hsr_enabled) - write_to_reg = 1; - break; - default: - /* All other register can be written */ + /* Decide if the given register can be written */ + switch (reg) { + case TWL4030_REG_EAR_CTL: + if (twl4030->earpiece_enabled) write_to_reg = 1; - break; - } - if (write_to_reg) - return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, - value, reg); + break; + case TWL4030_REG_PREDL_CTL: + if (twl4030->predrivel_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PREDR_CTL: + if (twl4030->predriver_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKL_CTL: + if (twl4030->carkitl_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_PRECKR_CTL: + if (twl4030->carkitr_enabled) + write_to_reg = 1; + break; + case TWL4030_REG_HS_GAIN_SET: + if (twl4030->hsl_enabled || twl4030->hsr_enabled) + write_to_reg = 1; + break; + default: + /* All other register can be written */ + write_to_reg = 1; + break; } + if (write_to_reg) + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + value, reg); + return 0; } @@ -532,7 +524,7 @@ SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); /* Handsfree Left virtual mute */ static const struct snd_kcontrol_new twl4030_dapm_handsfreelmute_control = - SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 0, 1, 0); + SOC_DAPM_SINGLE_VIRT("Switch", 1); /* Handsfree Right */ static const char *twl4030_handsfreer_texts[] = @@ -548,7 +540,7 @@ SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum); /* Handsfree Right virtual mute */ static const struct snd_kcontrol_new twl4030_dapm_handsfreermute_control = - SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 1, 1, 0); + SOC_DAPM_SINGLE_VIRT("Switch", 1); /* Vibra */ /* Vibra audio path selection */ diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 3c79dbb6c32..f2f4bcb2ff7 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -54,12 +54,7 @@ enum twl6040_dai_id { #define TWL6040_OUTHF_0dB 0x03 #define TWL6040_OUTHF_M52dB 0x1D -/* Shadow register used by the driver */ -#define TWL6040_REG_SW_SHADOW 0x2F -#define TWL6040_CACHEREGNUM (TWL6040_REG_SW_SHADOW + 1) - -/* TWL6040_REG_SW_SHADOW (0x2F) fields */ -#define TWL6040_EAR_PATH_ENABLE 0x01 +#define TWL6040_CACHEREGNUM (TWL6040_REG_STATUS + 1) struct twl6040_jack_data { struct snd_soc_jack *jack; @@ -135,8 +130,6 @@ static const u8 twl6040_reg[TWL6040_CACHEREGNUM] = { 0x00, /* REG_HFOTRIM 0x2C */ 0x09, /* REG_ACCCTL 0x2D */ 0x00, /* REG_STATUS 0x2E (ro) */ - - 0x00, /* REG_SW_SHADOW 0x2F - Shadow, non HW register */ }; /* List of registers to be restored after power up */ @@ -220,12 +213,8 @@ static int twl6040_read_reg_volatile(struct snd_soc_codec *codec, if (reg >= TWL6040_CACHEREGNUM) return -EIO; - if (likely(reg < TWL6040_REG_SW_SHADOW)) { - value = twl6040_reg_read(twl6040, reg); - twl6040_write_reg_cache(codec, reg, value); - } else { - value = twl6040_read_reg_cache(codec, reg); - } + value = twl6040_reg_read(twl6040, reg); + twl6040_write_reg_cache(codec, reg, value); return value; } @@ -246,7 +235,7 @@ static bool twl6040_is_path_unmuted(struct snd_soc_codec *codec, return priv->dl2_unmuted; default: return 1; - }; + } } /* @@ -261,8 +250,7 @@ static int twl6040_write(struct snd_soc_codec *codec, return -EIO; twl6040_write_reg_cache(codec, reg, value); - if (likely(reg < TWL6040_REG_SW_SHADOW) && - twl6040_is_path_unmuted(codec, reg)) + if (twl6040_is_path_unmuted(codec, reg)) return twl6040_reg_write(twl6040, reg, value); else return 0; @@ -555,7 +543,7 @@ static const struct snd_kcontrol_new hfr_mux_controls = SOC_DAPM_ENUM("Route", twl6040_hf_enum[1]); static const struct snd_kcontrol_new ep_path_enable_control = - SOC_DAPM_SINGLE("Switch", TWL6040_REG_SW_SHADOW, 0, 1, 0); + SOC_DAPM_SINGLE_VIRT("Switch", 1); static const struct snd_kcontrol_new auxl_switch_control = SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFLCTL, 6, 1, 0); @@ -1100,7 +1088,7 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i break; default: break; - }; + } } static int twl6040_digital_mute(struct snd_soc_dai *dai, int mute) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index bbd64384ca1..8c91be5d67e 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -983,24 +983,36 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { ARIZONA_MUX_ROUTES("ASRC2L", "ASRC2L"), ARIZONA_MUX_ROUTES("ASRC2R", "ASRC2R"), + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, + { "AEC Loopback", "HPOUT2L", "OUT2L" }, + { "AEC Loopback", "HPOUT2R", "OUT2R" }, { "HPOUT2L", NULL, "OUT2L" }, { "HPOUT2R", NULL, "OUT2R" }, + { "AEC Loopback", "HPOUT3L", "OUT3L" }, + { "AEC Loopback", "HPOUT3R", "OUT3R" }, { "HPOUT3L", NULL, "OUT3L" }, { "HPOUT3R", NULL, "OUT3L" }, + { "AEC Loopback", "SPKOUTL", "OUT4L" }, { "SPKOUTLN", NULL, "OUT4L" }, { "SPKOUTLP", NULL, "OUT4L" }, + { "AEC Loopback", "SPKOUTR", "OUT4R" }, { "SPKOUTRN", NULL, "OUT4R" }, { "SPKOUTRP", NULL, "OUT4R" }, + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, { "SPKDAT1L", NULL, "OUT5L" }, { "SPKDAT1R", NULL, "OUT5R" }, + { "AEC Loopback", "SPKDAT2L", "OUT6L" }, + { "AEC Loopback", "SPKDAT2R", "OUT6R" }, { "SPKDAT2L", NULL, "OUT6L" }, { "SPKDAT2R", NULL, "OUT6R" }, diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 8dfce8f1ad2..46ec0e9744d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -398,11 +398,12 @@ static int wm_coeff_write_control(struct snd_kcontrol *kcontrol, ret = regmap_raw_write(adsp->regmap, reg, scratch, ctl->len); if (ret) { - adsp_err(adsp, "Failed to write %zu bytes to %x\n", - ctl->len, reg); + adsp_err(adsp, "Failed to write %zu bytes to %x: %d\n", + ctl->len, reg, ret); kfree(scratch); return ret; } + adsp_dbg(adsp, "Wrote %zu bytes to %x\n", ctl->len, reg); kfree(scratch); @@ -452,11 +453,12 @@ static int wm_coeff_read_control(struct snd_kcontrol *kcontrol, ret = regmap_raw_read(adsp->regmap, reg, scratch, ctl->len); if (ret) { - adsp_err(adsp, "Failed to read %zu bytes from %x\n", - ctl->len, reg); + adsp_err(adsp, "Failed to read %zu bytes from %x: %d\n", + ctl->len, reg, ret); kfree(scratch); return ret; } + adsp_dbg(adsp, "Read %zu bytes from %x\n", ctl->len, reg); memcpy(buf, scratch, ctl->len); kfree(scratch); @@ -570,6 +572,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) file, header->ver); goto out_fw; } + adsp_info(dsp, "Firmware version: %d\n", header->ver); if (header->core != dsp->type) { adsp_err(dsp, "%s: invalid core %d != %d\n", @@ -686,7 +689,8 @@ static int wm_adsp_load(struct wm_adsp *dsp) &buf_list); if (!buf) { adsp_err(dsp, "Out of memory\n"); - return -ENOMEM; + ret = -ENOMEM; + goto out_fw; } ret = regmap_raw_write_async(regmap, reg, buf->buf, @@ -1057,6 +1061,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].dm); region->len -= be32_to_cpu(adsp1_alg[i].dm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region DM with ID %x\n", @@ -1074,6 +1079,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].zm); region->len -= be32_to_cpu(adsp1_alg[i].zm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", @@ -1103,6 +1109,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].xm); region->len -= be32_to_cpu(adsp2_alg[i].xm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region XM with ID %x\n", @@ -1120,6 +1127,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].ym); region->len -= be32_to_cpu(adsp2_alg[i].ym); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region YM with ID %x\n", @@ -1137,6 +1145,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].zm); region->len -= be32_to_cpu(adsp2_alg[i].zm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", @@ -1308,8 +1317,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) le32_to_cpu(blk->len)); if (ret != 0) { adsp_err(dsp, - "%s.%d: Failed to write to %x in %s\n", - file, blocks, reg, region_name); + "%s.%d: Failed to write to %x in %s: %d\n", + file, blocks, reg, region_name, ret); } } @@ -1353,6 +1362,7 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, struct snd_soc_codec *codec = w->codec; struct wm_adsp *dsps = snd_soc_codec_get_drvdata(codec); struct wm_adsp *dsp = &dsps[w->shift]; + struct wm_adsp_alg_region *alg_region; struct wm_coeff_ctl *ctl; int ret; int val; @@ -1430,6 +1440,14 @@ int wm_adsp1_event(struct snd_soc_dapm_widget *w, list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; + + while (!list_empty(&dsp->alg_regions)) { + alg_region = list_first_entry(&dsp->alg_regions, + struct wm_adsp_alg_region, + list); + list_del(&alg_region->list); + kfree(alg_region); + } break; default: diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index c82f89c9475..95970f5db3e 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -1,9 +1,10 @@ config SND_DAVINCI_SOC - tristate "SoC Audio for the TI DAVINCI chip" - depends on ARCH_DAVINCI + tristate "SoC Audio for the TI DAVINCI or AM33XX chip" + depends on ARCH_DAVINCI || SOC_AM33XX help + Platform driver for daVinci or AM33xx Say Y or M if you want to add support for codecs attached to - the DAVINCI AC97 or I2S interface. You will also need + the DAVINCI AC97, I2S, or McASP interface. You will also need to select the audio interfaces to support below. config SND_DAVINCI_SOC_I2S @@ -15,6 +16,17 @@ config SND_DAVINCI_SOC_MCASP config SND_DAVINCI_SOC_VCIF tristate +config SND_AM33XX_SOC_EVM + tristate "SoC Audio for the AM33XX chip based boards" + depends on SND_DAVINCI_SOC && SOC_AM33XX + select SND_SOC_TLV320AIC3X + select SND_DAVINCI_SOC_MCASP + help + Say Y or M if you want to add support for SoC audio on AM33XX + boards using McASP and TLV320AIC3X codec. For example AM335X-EVM, + AM335X-EVMSK, and BeagelBone with AudioCape boards have this + setup. + config SND_DAVINCI_SOC_EVM tristate "SoC Audio support for DaVinci DM6446, DM355 or DM365 EVM" depends on SND_DAVINCI_SOC diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index a396ab6d6d5..bc81e79fc30 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -13,6 +13,7 @@ obj-$(CONFIG_SND_DAVINCI_SOC_VCIF) += snd-soc-davinci-vcif.o snd-soc-evm-objs := davinci-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_AM33XX_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index fd7c45b9ed5..623eb5e7c08 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -16,6 +16,7 @@ #include <linux/platform_device.h> #include <linux/platform_data/edma.h> #include <linux/i2c.h> +#include <linux/of_platform.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -23,10 +24,16 @@ #include <asm/dma.h> #include <asm/mach-types.h> +#include <linux/edma.h> + #include "davinci-pcm.h" #include "davinci-i2s.h" #include "davinci-mcasp.h" +struct snd_soc_card_drvdata_davinci { + unsigned sysclk; +}; + #define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \ SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF) static int evm_hw_params(struct snd_pcm_substream *substream, @@ -35,27 +42,11 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *soc_card = codec->card; int ret = 0; - unsigned sysclk; - - /* ASP1 on DM355 EVM is clocked by an external oscillator */ - if (machine_is_davinci_dm355_evm() || machine_is_davinci_dm6467_evm() || - machine_is_davinci_dm365_evm()) - sysclk = 27000000; - - /* ASP0 in DM6446 EVM is clocked by U55, as configured by - * board-dm644x-evm.c using GPIOs from U18. There are six - * options; here we "know" we use a 48 KHz sample rate. - */ - else if (machine_is_davinci_evm()) - sysclk = 12288000; - - else if (machine_is_davinci_da830_evm() || - machine_is_davinci_da850_evm()) - sysclk = 24576000; - - else - return -EINVAL; + unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *) + snd_soc_card_get_drvdata(soc_card))->sysclk; /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT); @@ -133,13 +124,22 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; + struct device_node *np = codec->card->dev->of_node; + int ret; /* Add davinci-evm specific widgets */ snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); - /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + if (np) { + ret = snd_soc_of_parse_audio_routing(codec->card, + "ti,audio-routing"); + if (ret) + return ret; + } else { + /* Set up davinci-evm specific audio path audio_map */ + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + } /* not connected */ snd_soc_dapm_disable_pin(dapm, "MONO_LOUT"); @@ -243,35 +243,65 @@ static struct snd_soc_dai_link da850_evm_dai = { }; /* davinci dm6446 evm audio machine driver */ +/* + * ASP0 in DM6446 EVM is clocked by U55, as configured by + * board-dm644x-evm.c using GPIOs from U18. There are six + * options; here we "know" we use a 48 KHz sample rate. + */ +static struct snd_soc_card_drvdata_davinci dm6446_snd_soc_card_drvdata = { + .sysclk = 12288000, +}; + static struct snd_soc_card dm6446_snd_soc_card_evm = { .name = "DaVinci DM6446 EVM", .owner = THIS_MODULE, .dai_link = &dm6446_evm_dai, .num_links = 1, + .drvdata = &dm6446_snd_soc_card_drvdata, }; /* davinci dm355 evm audio machine driver */ +/* ASP1 on DM355 EVM is clocked by an external oscillator */ +static struct snd_soc_card_drvdata_davinci dm355_snd_soc_card_drvdata = { + .sysclk = 27000000, +}; + static struct snd_soc_card dm355_snd_soc_card_evm = { .name = "DaVinci DM355 EVM", .owner = THIS_MODULE, .dai_link = &dm355_evm_dai, .num_links = 1, + .drvdata = &dm355_snd_soc_card_drvdata, }; /* davinci dm365 evm audio machine driver */ +static struct snd_soc_card_drvdata_davinci dm365_snd_soc_card_drvdata = { + .sysclk = 27000000, +}; + static struct snd_soc_card dm365_snd_soc_card_evm = { .name = "DaVinci DM365 EVM", .owner = THIS_MODULE, .dai_link = &dm365_evm_dai, .num_links = 1, + .drvdata = &dm365_snd_soc_card_drvdata, }; /* davinci dm6467 evm audio machine driver */ +static struct snd_soc_card_drvdata_davinci dm6467_snd_soc_card_drvdata = { + .sysclk = 27000000, +}; + static struct snd_soc_card dm6467_snd_soc_card_evm = { .name = "DaVinci DM6467 EVM", .owner = THIS_MODULE, .dai_link = dm6467_evm_dai, .num_links = ARRAY_SIZE(dm6467_evm_dai), + .drvdata = &dm6467_snd_soc_card_drvdata, +}; + +static struct snd_soc_card_drvdata_davinci da830_snd_soc_card_drvdata = { + .sysclk = 24576000, }; static struct snd_soc_card da830_snd_soc_card = { @@ -279,6 +309,11 @@ static struct snd_soc_card da830_snd_soc_card = { .owner = THIS_MODULE, .dai_link = &da830_evm_dai, .num_links = 1, + .drvdata = &da830_snd_soc_card_drvdata, +}; + +static struct snd_soc_card_drvdata_davinci da850_snd_soc_card_drvdata = { + .sysclk = 24576000, }; static struct snd_soc_card da850_snd_soc_card = { @@ -286,8 +321,101 @@ static struct snd_soc_card da850_snd_soc_card = { .owner = THIS_MODULE, .dai_link = &da850_evm_dai, .num_links = 1, + .drvdata = &da850_snd_soc_card_drvdata, +}; + +#if defined(CONFIG_OF) + +/* + * The struct is used as place holder. It will be completely + * filled with data from dt node. + */ +static struct snd_soc_dai_link evm_dai_tlv320aic3x = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .codec_dai_name = "tlv320aic3x-hifi", + .ops = &evm_ops, + .init = evm_aic3x_init, +}; + +static const struct of_device_id davinci_evm_dt_ids[] = { + { + .compatible = "ti,da830-evm-audio", + .data = (void *) &evm_dai_tlv320aic3x, + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, davinci_evm_dt_ids); + +/* davinci evm audio machine driver */ +static struct snd_soc_card evm_soc_card = { + .owner = THIS_MODULE, + .num_links = 1, }; +static int davinci_evm_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + const struct of_device_id *match = + of_match_device(of_match_ptr(davinci_evm_dt_ids), &pdev->dev); + struct snd_soc_dai_link *dai = (struct snd_soc_dai_link *) match->data; + struct snd_soc_card_drvdata_davinci *drvdata = NULL; + int ret = 0; + + evm_soc_card.dai_link = dai; + + dai->codec_of_node = of_parse_phandle(np, "ti,audio-codec", 0); + if (!dai->codec_of_node) + return -EINVAL; + + dai->cpu_of_node = of_parse_phandle(np, "ti,mcasp-controller", 0); + if (!dai->cpu_of_node) + return -EINVAL; + + dai->platform_of_node = dai->cpu_of_node; + + evm_soc_card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&evm_soc_card, "ti,model"); + if (ret) + return ret; + + drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); + if (!drvdata) + return -ENOMEM; + + ret = of_property_read_u32(np, "ti,codec-clock-rate", &drvdata->sysclk); + if (ret < 0) + return -EINVAL; + + snd_soc_card_set_drvdata(&evm_soc_card, drvdata); + ret = devm_snd_soc_register_card(&pdev->dev, &evm_soc_card); + + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + + return ret; +} + +static int davinci_evm_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver davinci_evm_driver = { + .probe = davinci_evm_probe, + .remove = davinci_evm_remove, + .driver = { + .name = "davinci_evm", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(davinci_evm_dt_ids), + }, +}; +#endif + static struct platform_device *evm_snd_device; static int __init evm_init(void) @@ -296,6 +424,15 @@ static int __init evm_init(void) int index; int ret; + /* + * If dtb is there, the devices will be created dynamically. + * Only register platfrom driver structure. + */ +#if defined(CONFIG_OF) + if (of_have_populated_dt()) + return platform_driver_register(&davinci_evm_driver); +#endif + if (machine_is_davinci_evm()) { evm_snd_dev_data = &dm6446_snd_soc_card_evm; index = 0; @@ -331,6 +468,13 @@ static int __init evm_init(void) static void __exit evm_exit(void) { +#if defined(CONFIG_OF) + if (of_have_populated_dt()) { + platform_driver_unregister(&davinci_evm_driver); + return; + } +#endif + platform_device_unregister(evm_snd_device); } diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 32ddb7fe503..71e14bb3a8c 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1001,18 +1001,40 @@ static const struct snd_soc_component_driver davinci_mcasp_component = { .name = "davinci-mcasp", }; +/* Some HW specific values and defaults. The rest is filled in from DT. */ +static struct snd_platform_data dm646x_mcasp_pdata = { + .tx_dma_offset = 0x400, + .rx_dma_offset = 0x400, + .asp_chan_q = EVENTQ_0, + .version = MCASP_VERSION_1, +}; + +static struct snd_platform_data da830_mcasp_pdata = { + .tx_dma_offset = 0x2000, + .rx_dma_offset = 0x2000, + .asp_chan_q = EVENTQ_0, + .version = MCASP_VERSION_2, +}; + +static struct snd_platform_data omap2_mcasp_pdata = { + .tx_dma_offset = 0, + .rx_dma_offset = 0, + .asp_chan_q = EVENTQ_0, + .version = MCASP_VERSION_3, +}; + static const struct of_device_id mcasp_dt_ids[] = { { .compatible = "ti,dm646x-mcasp-audio", - .data = (void *)MCASP_VERSION_1, + .data = &dm646x_mcasp_pdata, }, { .compatible = "ti,da830-mcasp-audio", - .data = (void *)MCASP_VERSION_2, + .data = &da830_mcasp_pdata, }, { - .compatible = "ti,omap2-mcasp-audio", - .data = (void *)MCASP_VERSION_3, + .compatible = "ti,am33xx-mcasp-audio", + .data = &omap2_mcasp_pdata, }, { /* sentinel */ } }; @@ -1025,9 +1047,9 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( struct snd_platform_data *pdata = NULL; const struct of_device_id *match = of_match_device(mcasp_dt_ids, &pdev->dev); + struct of_phandle_args dma_spec; const u32 *of_serial_dir32; - u8 *of_serial_dir; u32 val; int i, ret = 0; @@ -1035,20 +1057,13 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( pdata = pdev->dev.platform_data; return pdata; } else if (match) { - pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); - if (!pdata) { - ret = -ENOMEM; - goto nodata; - } + pdata = (struct snd_platform_data *) match->data; } else { /* control shouldn't reach here. something is wrong */ ret = -EINVAL; goto nodata; } - if (match->data) - pdata->version = (u8)((int)match->data); - ret = of_property_read_u32(np, "op-mode", &val); if (ret >= 0) pdata->op_mode = val; @@ -1065,35 +1080,46 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( pdata->tdm_slots = val; } - ret = of_property_read_u32(np, "num-serializer", &val); - if (ret >= 0) - pdata->num_serializer = val; - of_serial_dir32 = of_get_property(np, "serial-dir", &val); val /= sizeof(u32); - if (val != pdata->num_serializer) { - dev_err(&pdev->dev, - "num-serializer(%d) != serial-dir size(%d)\n", - pdata->num_serializer, val); - ret = -EINVAL; - goto nodata; - } - if (of_serial_dir32) { - of_serial_dir = devm_kzalloc(&pdev->dev, - (sizeof(*of_serial_dir) * val), - GFP_KERNEL); + u8 *of_serial_dir = devm_kzalloc(&pdev->dev, + (sizeof(*of_serial_dir) * val), + GFP_KERNEL); if (!of_serial_dir) { ret = -ENOMEM; goto nodata; } - for (i = 0; i < pdata->num_serializer; i++) + for (i = 0; i < val; i++) of_serial_dir[i] = be32_to_cpup(&of_serial_dir32[i]); + pdata->num_serializer = val; pdata->serial_dir = of_serial_dir; } + ret = of_property_match_string(np, "dma-names", "tx"); + if (ret < 0) + goto nodata; + + ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", ret, + &dma_spec); + if (ret < 0) + goto nodata; + + pdata->tx_dma_channel = dma_spec.args[0]; + + ret = of_property_match_string(np, "dma-names", "rx"); + if (ret < 0) + goto nodata; + + ret = of_parse_phandle_with_args(np, "dmas", "#dma-cells", ret, + &dma_spec); + if (ret < 0) + goto nodata; + + pdata->rx_dma_channel = dma_spec.args[0]; + ret = of_property_read_u32(np, "tx-num-evt", &val); if (ret >= 0) pdata->txnumevt = val; @@ -1124,7 +1150,7 @@ nodata: static int davinci_mcasp_probe(struct platform_device *pdev) { struct davinci_pcm_dma_params *dma_data; - struct resource *mem, *ioarea, *res; + struct resource *mem, *ioarea, *res, *dat; struct snd_platform_data *pdata; struct davinci_audio_dev *dev; int ret; @@ -1145,10 +1171,15 @@ static int davinci_mcasp_probe(struct platform_device *pdev) return -EINVAL; } - mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); if (!mem) { - dev_err(&pdev->dev, "no mem resource?\n"); - return -ENODEV; + dev_warn(dev->dev, + "\"mpu\" mem resource not found, using index 0\n"); + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + return -ENODEV; + } } ioarea = devm_request_mem_region(&pdev->dev, mem->start, @@ -1182,40 +1213,36 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->rxnumevt = pdata->rxnumevt; dev->dev = &pdev->dev; + dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat"); + if (!dat) + dat = mem; + dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->sram_pool = pdata->sram_pool; dma_data->sram_size = pdata->sram_size_playback; - dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset + - mem->start); + dma_data->dma_addr = dat->start + pdata->tx_dma_offset; - /* first TX, then RX */ res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { - dev_err(&pdev->dev, "no DMA resource\n"); - ret = -ENODEV; - goto err_release_clk; - } - - dma_data->channel = res->start; + if (res) + dma_data->channel = res->start; + else + dma_data->channel = pdata->tx_dma_channel; dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]; dma_data->asp_chan_q = pdata->asp_chan_q; dma_data->ram_chan_q = pdata->ram_chan_q; dma_data->sram_pool = pdata->sram_pool; dma_data->sram_size = pdata->sram_size_capture; - dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset + - mem->start); + dma_data->dma_addr = dat->start + pdata->rx_dma_offset; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!res) { - dev_err(&pdev->dev, "no DMA resource\n"); - ret = -ENODEV; - goto err_release_clk; - } + if (res) + dma_data->channel = res->start; + else + dma_data->channel = pdata->rx_dma_channel; - dma_data->channel = res->start; dev_set_drvdata(&pdev->dev, dev); ret = snd_soc_register_component(&pdev->dev, &davinci_mcasp_component, &davinci_mcasp_dai[pdata->op_mode], 1); @@ -1251,12 +1278,51 @@ static int davinci_mcasp_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_PM_SLEEP +static int davinci_mcasp_suspend(struct device *dev) +{ + struct davinci_audio_dev *a = dev_get_drvdata(dev); + void __iomem *base = a->base; + + a->context.txfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_TXFMCTL_REG); + a->context.rxfmtctl = mcasp_get_reg(base + DAVINCI_MCASP_RXFMCTL_REG); + a->context.txfmt = mcasp_get_reg(base + DAVINCI_MCASP_TXFMT_REG); + a->context.rxfmt = mcasp_get_reg(base + DAVINCI_MCASP_RXFMT_REG); + a->context.aclkxctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKXCTL_REG); + a->context.aclkrctl = mcasp_get_reg(base + DAVINCI_MCASP_ACLKRCTL_REG); + a->context.pdir = mcasp_get_reg(base + DAVINCI_MCASP_PDIR_REG); + + return 0; +} + +static int davinci_mcasp_resume(struct device *dev) +{ + struct davinci_audio_dev *a = dev_get_drvdata(dev); + void __iomem *base = a->base; + + mcasp_set_reg(base + DAVINCI_MCASP_TXFMCTL_REG, a->context.txfmtctl); + mcasp_set_reg(base + DAVINCI_MCASP_RXFMCTL_REG, a->context.rxfmtctl); + mcasp_set_reg(base + DAVINCI_MCASP_TXFMT_REG, a->context.txfmt); + mcasp_set_reg(base + DAVINCI_MCASP_RXFMT_REG, a->context.rxfmt); + mcasp_set_reg(base + DAVINCI_MCASP_ACLKXCTL_REG, a->context.aclkxctl); + mcasp_set_reg(base + DAVINCI_MCASP_ACLKRCTL_REG, a->context.aclkrctl); + mcasp_set_reg(base + DAVINCI_MCASP_PDIR_REG, a->context.pdir); + + return 0; +} +#endif + +SIMPLE_DEV_PM_OPS(davinci_mcasp_pm_ops, + davinci_mcasp_suspend, + davinci_mcasp_resume); + static struct platform_driver davinci_mcasp_driver = { .probe = davinci_mcasp_probe, .remove = davinci_mcasp_remove, .driver = { .name = "davinci-mcasp", .owner = THIS_MODULE, + .pm = &davinci_mcasp_pm_ops, .of_match_table = mcasp_dt_ids, }, }; @@ -1266,4 +1332,3 @@ module_platform_driver(davinci_mcasp_driver); MODULE_AUTHOR("Steve Chen"); MODULE_DESCRIPTION("TI DAVINCI McASP SoC Interface"); MODULE_LICENSE("GPL"); - diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a9ac0c11da7..a2e27e1c32f 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -43,6 +43,18 @@ struct davinci_audio_dev { /* McASP FIFO related */ u8 txnumevt; u8 rxnumevt; + +#ifdef CONFIG_PM_SLEEP + struct { + u32 txfmtctl; + u32 rxfmtctl; + u32 txfmt; + u32 rxfmt; + u32 aclkxctl; + u32 aclkrctl; + u32 pdir; + } context; +#endif }; #endif /* DAVINCI_MCASP_H */ diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 9a4a0ca2c1d..5983740be12 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -42,7 +42,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { - pr_err("%s: failed set cpu dai format\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the cpu dai format.\n"); return ret; } @@ -50,14 +51,16 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret) { - pr_err("%s: failed set codec dai format\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the codec format.\n"); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_OUT); if (ret) { - pr_err("%s: failed setting codec sysclk\n", __func__); + dev_err(cpu_dai->dev, + "Failed to set the codec sysclk.\n"); return ret; } snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); @@ -65,7 +68,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, SND_SOC_CLOCK_IN); if (ret) { - pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); + dev_err(cpu_dai->dev, + "Can't set the IMX_SSP_SYS_CLK CPU system clock.\n"); return ret; } @@ -155,7 +159,8 @@ static struct platform_driver eukrea_tlv320_driver = { .owner = THIS_MODULE, }, .probe = eukrea_tlv320_probe, - .remove = eukrea_tlv320_remove,}; + .remove = eukrea_tlv320_remove, +}; module_platform_driver(eukrea_tlv320_driver); diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 3920c3e849c..55193a5596c 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -963,7 +963,7 @@ static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) @@ -982,7 +982,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) return true; default: return false; - }; + } } static const struct regmap_config fsl_spdif_regmap_config = { @@ -1107,11 +1107,6 @@ static int fsl_spdif_probe(struct platform_device *pdev) /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (IS_ERR(res)) { - dev_err(&pdev->dev, "could not determine device resources\n"); - return PTR_ERR(res); - } - regs = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(regs)) return PTR_ERR(regs); @@ -1172,23 +1167,16 @@ static int fsl_spdif_probe(struct platform_device *pdev) /* Register with ASoC */ dev_set_drvdata(&pdev->dev, spdif_priv); - ret = snd_soc_register_component(&pdev->dev, &fsl_spdif_component, - &spdif_priv->cpu_dai_drv, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component, + &spdif_priv->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); return ret; } ret = imx_pcm_dma_init(pdev); - if (ret) { + if (ret) dev_err(&pdev->dev, "imx_pcm_dma_init failed: %d\n", ret); - goto error_component; - } - - return ret; - -error_component: - snd_soc_unregister_component(&pdev->dev); return ret; } @@ -1196,7 +1184,6 @@ error_component: static int fsl_spdif_remove(struct platform_device *pdev) { imx_pcm_dma_exit(pdev); - snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6b81d0ce2c4..35e277379b8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -469,19 +469,12 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, * parameters, then the second stream may be * constrained to the wrong sample rate or size. */ - if (!first_runtime->sample_bits) { - dev_err(substream->pcm->card->dev, - "set sample size in %s stream first\n", - substream->stream == - SNDRV_PCM_STREAM_PLAYBACK - ? "capture" : "playback"); - return -EAGAIN; - } - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + if (first_runtime->sample_bits) { + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, first_runtime->sample_bits, first_runtime->sample_bits); + } } ssi_private->second_stream = substream; @@ -748,7 +741,7 @@ static void fsl_ssi_ac97_init(void) fsl_ssi_setup(fsl_ac97_data); } -void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, +static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { struct ccsr_ssi *ssi = fsl_ac97_data->ssi; @@ -770,7 +763,7 @@ void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, udelay(100); } -unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, +static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { struct ccsr_ssi *ssi = fsl_ac97_data->ssi; @@ -936,7 +929,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi_private->ssi_phys = res.start; ssi_private->irq = irq_of_parse_and_map(np, 0); - if (ssi_private->irq == 0) { + if (!ssi_private->irq) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); return -ENXIO; } @@ -1135,7 +1128,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->ssi_on_imx) imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, &ssi_private->dev_attr); if (ssi_private->ssi_on_imx) clk_disable_unprepare(ssi_private->clk); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index d3bf71a0ec5..ac869931d7f 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -66,13 +66,10 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, size_t count, loff_t *ppos) { ssize_t ret; - char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + char *buf; int port = (int)file->private_data; u32 pdcr, ptcr; - if (!buf) - return -ENOMEM; - if (audmux_clk) { ret = clk_prepare_enable(audmux_clk); if (ret) @@ -85,6 +82,10 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (audmux_clk) clk_disable_unprepare(audmux_clk); + buf = kmalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + ret = snprintf(buf, PAGE_SIZE, "PDCR: %08x\nPTCR: %08x\n", pdcr, ptcr); diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index a2fd7321b5a..79cee782dbb 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -160,6 +160,7 @@ static struct platform_driver imx_mc13783_audio_driver = { .driver = { .name = "imx_mc13783", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = imx_mc13783_probe, .remove = imx_mc13783_remove diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 4dc1296688e..aee23077080 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -25,12 +25,10 @@ static bool filter(struct dma_chan *chan, void *param) { - struct snd_dmaengine_dai_dma_data *dma_data = param; - if (!imx_dma_is_general_purpose(chan)) return false; - chan->private = dma_data->filter_data; + chan->private = param; return true; } diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 34043c55f2a..10e330514ed 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -39,8 +39,6 @@ struct imx_pcm_runtime_data { unsigned int period; int periods; unsigned long offset; - unsigned long last_offset; - unsigned long size; struct hrtimer hrt; int poll_time_ns; struct snd_pcm_substream *substream; @@ -52,9 +50,7 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) struct imx_pcm_runtime_data *iprtd = container_of(hrt, struct imx_pcm_runtime_data, hrt); struct snd_pcm_substream *substream = iprtd->substream; - struct snd_pcm_runtime *runtime = substream->runtime; struct pt_regs regs; - unsigned long delta; if (!atomic_read(&iprtd->running)) return HRTIMER_NORESTART; @@ -66,19 +62,7 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) else iprtd->offset = regs.ARM_r9 & 0xffff; - /* How much data have we transferred since the last period report? */ - if (iprtd->offset >= iprtd->last_offset) - delta = iprtd->offset - iprtd->last_offset; - else - delta = runtime->buffer_size + iprtd->offset - - iprtd->last_offset; - - /* If we've transferred at least a period then report it and - * reset our poll time */ - if (delta >= iprtd->period) { - snd_pcm_period_elapsed(substream); - iprtd->last_offset = iprtd->offset; - } + snd_pcm_period_elapsed(substream); hrtimer_forward_now(hrt, ns_to_ktime(iprtd->poll_time_ns)); @@ -95,11 +79,9 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; - iprtd->size = params_buffer_bytes(params); iprtd->periods = params_periods(params); - iprtd->period = params_period_bytes(params) ; + iprtd->period = params_period_bytes(params); iprtd->offset = 0; - iprtd->last_offset = 0; iprtd->poll_time_ns = 1000000000 / params_rate(params) * params_period_size(params); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index ca1be1d9dcf..f2beae78969 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -159,7 +159,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dapm_widgets = imx_sgtl5000_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets); - ret = snd_soc_register_card(&data->card); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto fail; @@ -186,7 +186,6 @@ static int imx_sgtl5000_remove(struct platform_device *pdev) { struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); - snd_soc_unregister_card(&data->card); clk_put(data->codec_clk); return 0; @@ -202,6 +201,7 @@ static struct platform_driver imx_sgtl5000_driver = { .driver = { .name = "imx-sgtl5000", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = imx_sgtl5000_dt_ids, }, .probe = imx_sgtl5000_probe, diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c index 816013b0ebb..8499d5292f0 100644 --- a/sound/soc/fsl/imx-spdif.c +++ b/sound/soc/fsl/imx-spdif.c @@ -87,7 +87,7 @@ static int imx_spdif_audio_probe(struct platform_device *pdev) if (ret) goto error_dir; - ret = snd_soc_register_card(&data->card); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); goto error_dir; @@ -119,8 +119,6 @@ static int imx_spdif_audio_remove(struct platform_device *pdev) if (data->txdev) platform_device_unregister(data->txdev); - snd_soc_unregister_card(&data->card); - return 0; } diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 57d6941676f..f5f248c91c1 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -613,7 +613,6 @@ static int imx_ssi_probe(struct platform_device *pdev) failed_pcm: snd_soc_unregister_component(&pdev->dev); failed_register: - release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); failed_clk: snd_soc_set_ac97_ops(NULL); @@ -623,7 +622,6 @@ failed_clk: static int imx_ssi_remove(struct platform_device *pdev) { - struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct imx_ssi *ssi = platform_get_drvdata(pdev); if (!ssi->dma_init) @@ -637,7 +635,6 @@ static int imx_ssi_remove(struct platform_device *pdev) if (ssi->flags & IMX_SSI_USE_AC97) ac97_ssi = NULL; - release_mem_region(res->start, resource_size(res)); clk_disable_unprepare(ssi->clk); snd_soc_set_ac97_ops(NULL); diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 722afe69169..361f94f09b1 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -266,7 +266,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->card.late_probe = imx_wm8962_late_probe; data->card.set_bias_level = imx_wm8962_set_bias_level; - ret = snd_soc_register_card(&data->card); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto clk_fail; @@ -279,8 +279,7 @@ static int imx_wm8962_probe(struct platform_device *pdev) return 0; clk_fail: - if (!IS_ERR(data->codec_clk)) - clk_disable_unprepare(data->codec_clk); + clk_disable_unprepare(data->codec_clk); fail: if (ssi_np) of_node_put(ssi_np); @@ -296,7 +295,6 @@ static int imx_wm8962_remove(struct platform_device *pdev) if (!IS_ERR(data->codec_clk)) clk_disable_unprepare(data->codec_clk); - snd_soc_unregister_card(&data->card); return 0; } @@ -311,6 +309,7 @@ static struct platform_driver imx_wm8962_driver = { .driver = { .name = "imx-wm8962", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = imx_wm8962_dt_ids, }, .probe = imx_wm8962_probe, diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 8c49147db84..b2fbb7075a6 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -27,6 +27,11 @@ static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, if (!ret && daifmt) ret = snd_soc_dai_set_fmt(dai, daifmt); + if (ret == -ENOTSUPP) { + dev_dbg(dai->dev, "ASoC: set_fmt is not supported\n"); + ret = 0; + } + if (!ret && set->sysclk) ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index b238434f92b..55d0d9d3a9f 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -29,9 +29,7 @@ #define KIRKWOOD_FORMATS \ (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE | \ - SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE | \ - SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE) + SNDRV_PCM_FMTBIT_S32_LE) static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs) { @@ -161,7 +159,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) * Enable Error interrupts. We're only ack'ing them but * it's useful for diagnostics */ - writel((unsigned long)-1, priv->io + KIRKWOOD_ERR_MASK); + writel((unsigned int)-1, priv->io + KIRKWOOD_ERR_MASK); } dram = mv_mbus_dram_info(); diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0f3d73d4ef4..d34d91743e3 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -103,7 +103,7 @@ static void kirkwood_set_rate(struct snd_soc_dai *dai, { uint32_t clks_ctrl; - if (rate == 44100 || rate == 48000 || rate == 96000) { + if (IS_ERR(priv->extclk)) { /* use internal dco for the supported rates * defined in kirkwood_i2s_dai */ dev_dbg(dai->dev, "%s: dco set rate = %lu\n", @@ -160,9 +160,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S16_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_16; ctl_play = KIRKWOOD_PLAYCTL_SIZE_16_C | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_16_C | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; /* * doesn't work... S20_3LE != kirkwood 20bit format ? @@ -178,9 +180,11 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S24_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_24; ctl_play = KIRKWOOD_PLAYCTL_SIZE_24 | - KIRKWOOD_PLAYCTL_I2S_EN; + KIRKWOOD_PLAYCTL_I2S_EN | + KIRKWOOD_PLAYCTL_SPDIF_EN; ctl_rec = KIRKWOOD_RECCTL_SIZE_24 | - KIRKWOOD_RECCTL_I2S_EN; + KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN; break; case SNDRV_PCM_FORMAT_S32_LE: i2s_value |= KIRKWOOD_I2S_CTL_SIZE_32; @@ -240,6 +244,11 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl); } + if (dai->id == 0) + ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ + switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* configure */ @@ -258,7 +267,8 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: /* stop audio, disable interrupts */ - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_INT_MASK); @@ -272,13 +282,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE; + ctl |= KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE; writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE); + ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | + KIRKWOOD_PLAYCTL_SPDIF_MUTE); writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; @@ -301,7 +313,13 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_rec; - value = ctl & ~KIRKWOOD_RECCTL_I2S_EN; + if (dai->id == 0) + ctl &= ~KIRKWOOD_RECCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif */ + + value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN | + KIRKWOOD_RECCTL_SPDIF_EN); writel(value, priv->io + KIRKWOOD_RECCTL); /* enable interrupts */ @@ -361,9 +379,8 @@ static int kirkwood_i2s_trigger(struct snd_pcm_substream *substream, int cmd, return 0; } -static int kirkwood_i2s_probe(struct snd_soc_dai *dai) +static int kirkwood_i2s_init(struct kirkwood_dma_data *priv) { - struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); unsigned long value; unsigned int reg_data; @@ -404,9 +421,10 @@ static const struct snd_soc_dai_ops kirkwood_i2s_dai_ops = { .set_fmt = kirkwood_i2s_set_fmt, }; - -static struct snd_soc_dai_driver kirkwood_i2s_dai = { - .probe = kirkwood_i2s_probe, +static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = { + { + .name = "i2s", + .id = 0, .playback = { .channels_min = 1, .channels_max = 2, @@ -422,10 +440,53 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai = { .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, + }, + { + .name = "spdif", + .id = 1, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .ops = &kirkwood_i2s_dai_ops, + }, }; -static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { - .probe = kirkwood_i2s_probe, +static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { + { + .name = "i2s", + .id = 0, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000 | + SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000 | + SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_KNOT, + .formats = KIRKWOOD_I2S_FORMATS, + }, + .ops = &kirkwood_i2s_dai_ops, + }, + { + .name = "spdif", + .id = 1, .playback = { .channels_min = 1, .channels_max = 2, @@ -443,6 +504,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk = { .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, + }, }; static const struct snd_soc_component_driver kirkwood_i2s_component = { @@ -452,7 +514,7 @@ static const struct snd_soc_component_driver kirkwood_i2s_component = { static int kirkwood_i2s_dev_probe(struct platform_device *pdev) { struct kirkwood_asoc_platform_data *data = pdev->dev.platform_data; - struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; + struct snd_soc_dai_driver *soc_dai = kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; struct device_node *np = pdev->dev.of_node; @@ -496,14 +558,17 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return err; priv->extclk = devm_clk_get(&pdev->dev, "extclk"); - if (!IS_ERR(priv->extclk)) { + if (IS_ERR(priv->extclk)) { + if (PTR_ERR(priv->extclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } else { if (priv->extclk == priv->clk) { devm_clk_put(&pdev->dev, priv->extclk); priv->extclk = ERR_PTR(-EINVAL); } else { dev_info(&pdev->dev, "found external clock\n"); clk_prepare_enable(priv->extclk); - soc_dai = &kirkwood_i2s_dai_extclk; + soc_dai = kirkwood_i2s_dai_extclk; } } @@ -521,7 +586,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) } err = snd_soc_register_component(&pdev->dev, &kirkwood_i2s_component, - soc_dai, 1); + soc_dai, 2); if (err) { dev_err(&pdev->dev, "snd_soc_register_component failed\n"); goto err_component; @@ -532,6 +597,9 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "snd_soc_register_platform failed\n"); goto err_platform; } + + kirkwood_i2s_init(priv); + return 0; err_platform: snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index 025be0e9716..65f2a5b9ec3 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -52,7 +52,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { { .name = "CS42L51", .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "cs42l51-hifi", .codec_name = "cs42l51-codec.0-004a", diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c index 27545b0c485..d213832b0c7 100644 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ b/sound/soc/kirkwood/kirkwood-t5325.c @@ -68,7 +68,7 @@ static struct snd_soc_dai_link t5325_dai[] = { { .name = "ALC5621", .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "mvebu-audio", + .cpu_dai_name = "i2s", .platform_name = "mvebu-audio", .codec_dai_name = "alc5621-hifi", .codec_name = "alc562x-codec.0-001a", diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index f8e1ccc1c58..bf23afbba1d 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -123,8 +123,8 @@ /* need to find where they come from */ #define KIRKWOOD_SND_MIN_PERIODS 8 #define KIRKWOOD_SND_MAX_PERIODS 16 -#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x4000 -#define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x4000 +#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x800 +#define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x8000 #define KIRKWOOD_SND_MAX_BUFFER_BYTES (KIRKWOOD_SND_MAX_PERIOD_BYTES \ * KIRKWOOD_SND_MAX_PERIODS) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index ee363845759..d3d4c32434f 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -400,7 +400,7 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) } /* register the soc card */ snd_soc_card_mfld.dev = &pdev->dev; - ret_val = snd_soc_register_card(&snd_soc_card_mfld); + ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld); if (ret_val) { pr_debug("snd_soc_register_card failed %d\n", ret_val); return ret_val; @@ -410,20 +410,12 @@ static int snd_mfld_mc_probe(struct platform_device *pdev) return 0; } -static int snd_mfld_mc_remove(struct platform_device *pdev) -{ - pr_debug("snd_mfld_mc_remove called\n"); - snd_soc_unregister_card(&snd_soc_card_mfld); - return 0; -} - static struct platform_driver snd_mfld_mc_driver = { .driver = { .owner = THIS_MODULE, .name = "msic_audio", }, .probe = snd_mfld_mc_probe, - .remove = snd_mfld_mc_remove, }; module_platform_driver(snd_mfld_mc_driver); diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index b56b8a0e8de..54e622acac3 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -494,6 +494,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); struct mxs_saif *master_saif; u32 delay; + int ret; master_saif = mxs_saif_get_master(saif); if (!master_saif) @@ -503,23 +504,37 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (saif->state == MXS_SAIF_STATE_RUNNING) + return 0; + dev_dbg(cpu_dai->dev, "start\n"); - clk_enable(master_saif->clk); - if (!master_saif->mclk_in_use) - __raw_writel(BM_SAIF_CTRL_RUN, - master_saif->base + SAIF_CTRL + MXS_SET_ADDR); + ret = clk_enable(master_saif->clk); + if (ret) { + dev_err(saif->dev, "Failed to enable master clock\n"); + return ret; + } /* * If the saif's master is not himself, we also need to enable * itself clk for its internal basic logic to work. */ if (saif != master_saif) { - clk_enable(saif->clk); + ret = clk_enable(saif->clk); + if (ret) { + dev_err(saif->dev, "Failed to enable master clock\n"); + clk_disable(master_saif->clk); + return ret; + } + __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_SET_ADDR); } + if (!master_saif->mclk_in_use) + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_SET_ADDR); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* * write data to saif data register to trigger @@ -543,6 +558,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, } master_saif->ongoing = 1; + saif->state = MXS_SAIF_STATE_RUNNING; dev_dbg(saif->dev, "CTRL 0x%x STAT 0x%x\n", __raw_readl(saif->base + SAIF_CTRL), @@ -555,6 +571,9 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (saif->state == MXS_SAIF_STATE_STOPPED) + return 0; + dev_dbg(cpu_dai->dev, "stop\n"); /* wait a while for the current sample to complete */ @@ -575,6 +594,7 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, } master_saif->ongoing = 0; + saif->state = MXS_SAIF_STATE_STOPPED; break; default: @@ -768,8 +788,8 @@ static int mxs_saif_probe(struct platform_device *pdev) dev_warn(&pdev->dev, "failed to init clocks\n"); } - ret = snd_soc_register_component(&pdev->dev, &mxs_saif_component, - &mxs_saif_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &mxs_saif_component, + &mxs_saif_dai, 1); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); return ret; @@ -778,21 +798,15 @@ static int mxs_saif_probe(struct platform_device *pdev) ret = mxs_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "register PCM failed: %d\n", ret); - goto failed_pdev_alloc; + return ret; } return 0; - -failed_pdev_alloc: - snd_soc_unregister_component(&pdev->dev); - - return ret; } static int mxs_saif_remove(struct platform_device *pdev) { mxs_pcm_platform_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 53eaa4bf0e2..fbaf7badfdf 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -124,6 +124,11 @@ struct mxs_saif { u32 fifo_underrun; u32 fifo_overrun; + + enum { + MXS_SAIF_STATE_STOPPED, + MXS_SAIF_STATE_RUNNING, + } state; }; extern int mxs_saif_put_mclk(unsigned int saif_id); diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 4bb273786ff..61822cc53bd 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -122,14 +122,12 @@ static struct snd_soc_card mxs_sgtl5000 = { .num_links = ARRAY_SIZE(mxs_sgtl5000_dai), }; -static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) +static int mxs_sgtl5000_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &mxs_sgtl5000; + int ret, i; struct device_node *np = pdev->dev.of_node; struct device_node *saif_np[2], *codec_np; - int i; - - if (!np) - return 1; /* no device tree */ saif_np[0] = of_parse_phandle(np, "saif-controllers", 0); saif_np[1] = of_parse_phandle(np, "saif-controllers", 1); @@ -152,18 +150,6 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) of_node_put(saif_np[0]); of_node_put(saif_np[1]); - return 0; -} - -static int mxs_sgtl5000_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &mxs_sgtl5000; - int ret; - - ret = mxs_sgtl5000_probe_dt(pdev); - if (ret < 0) - return ret; - /* * Set an init clock(11.28Mhz) for sgtl5000 initialization(i2c r/w). * The Sgtl5000 sysclk is derived from saif0 mclk and it's range diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 90d2a7cd256..cd9ee167959 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -490,14 +490,9 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dev = &pdev->dev; - return snd_soc_register_component(&pdev->dev, &omap_mcpdm_component, - &omap_mcpdm_dai, 1); -} - -static int asoc_mcpdm_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - return 0; + return devm_snd_soc_register_component(&pdev->dev, + &omap_mcpdm_component, + &omap_mcpdm_dai, 1); } static const struct of_device_id omap_mcpdm_of_match[] = { @@ -514,7 +509,6 @@ static struct platform_driver asoc_mcpdm_driver = { }, .probe = asoc_mcpdm_probe, - .remove = asoc_mcpdm_remove, }; module_platform_driver(asoc_mcpdm_driver); diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 2a9324f794d..6a8d6b5f160 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -338,9 +338,9 @@ static int omap_twl4030_probe(struct platform_device *pdev) } snd_soc_card_set_drvdata(card, priv); - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n", ret); return ret; } @@ -357,7 +357,6 @@ static int omap_twl4030_remove(struct platform_device *pdev) snd_soc_jack_free_gpios(&priv->hs_jack, ARRAY_SIZE(hs_jack_gpios), hs_jack_gpios); - snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 5b7d969f89a..08acdc236bf 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -163,6 +163,7 @@ static struct platform_driver mmp_driver = { .driver = { .name = "brownstone-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = brownstone_probe, .remove = brownstone_remove, diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index f4cce1e8011..1853d41034b 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -329,6 +329,7 @@ static struct platform_driver corgi_driver = { .driver = { .name = "corgi-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = corgi_probe, .remove = corgi_remove, diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 70d799b13f0..44b5c09d296 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -178,6 +178,7 @@ static struct platform_driver e740_driver = { .driver = { .name = "e740-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = e740_probe, .remove = e740_remove, diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index f94d2ab5135..c34e447eb99 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -160,6 +160,7 @@ static struct platform_driver e750_driver = { .driver = { .name = "e750-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = e750_probe, .remove = e750_remove, diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 8768a640dd7..3137f800b43 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -150,6 +150,7 @@ static struct platform_driver e800_driver = { .driver = { .name = "e800-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = e800_probe, .remove = e800_remove, diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index eef1f7b7b38..fd2f4eda1fd 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -91,6 +91,7 @@ static struct platform_driver imote2_driver = { .driver = { .name = "imote2-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = imote2_probe, .remove = imote2_remove, diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index bbea7780eac..160c5245448 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -215,6 +215,7 @@ static struct platform_driver mioa701_wm9713_driver = { .driver = { .name = "mioa701-wm9713", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, }; diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 41752a5fe3b..5bf5f1f7cac 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -455,8 +455,8 @@ static int asoc_mmp_sspa_probe(struct platform_device *pdev) priv->dai_fmt = (unsigned int) -1; platform_set_drvdata(pdev, priv); - return snd_soc_register_component(&pdev->dev, &mmp_sspa_component, - &mmp_sspa_dai, 1); + return devm_snd_soc_register_component(&pdev->dev, &mmp_sspa_component, + &mmp_sspa_dai, 1); } static int asoc_mmp_sspa_remove(struct platform_device *pdev) @@ -466,7 +466,6 @@ static int asoc_mmp_sspa_remove(struct platform_device *pdev) clk_disable(priv->audio_clk); clk_put(priv->audio_clk); clk_put(priv->sysclk); - snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index e1ffcdd9a64..3284c4b901c 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -181,6 +181,7 @@ static struct platform_driver palm27x_wm9712_driver = { .driver = { .name = "palm27x-asoc", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, }; diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index fafe46355c3..c93e138d8dc 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -303,6 +303,7 @@ static struct platform_driver poodle_driver = { .driver = { .name = "poodle-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = poodle_probe, .remove = poodle_remove, diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index f1059d999de..ae956e3f4b9 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -89,33 +89,6 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_mic_mono_in = { .filter_data = &pxa2xx_ac97_pcm_aux_mic_mono_req, }; -#ifdef CONFIG_PM -static int pxa2xx_ac97_suspend(struct snd_soc_dai *dai) -{ - return pxa2xx_ac97_hw_suspend(); -} - -static int pxa2xx_ac97_resume(struct snd_soc_dai *dai) -{ - return pxa2xx_ac97_hw_resume(); -} - -#else -#define pxa2xx_ac97_suspend NULL -#define pxa2xx_ac97_resume NULL -#endif - -static int pxa2xx_ac97_probe(struct snd_soc_dai *dai) -{ - return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev)); -} - -static int pxa2xx_ac97_remove(struct snd_soc_dai *dai) -{ - pxa2xx_ac97_hw_remove(to_platform_device(dai->dev)); - return 0; -} - static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) @@ -185,10 +158,6 @@ static struct snd_soc_dai_driver pxa_ac97_dai_driver[] = { { .name = "pxa2xx-ac97", .ac97_control = 1, - .probe = pxa2xx_ac97_probe, - .remove = pxa2xx_ac97_remove, - .suspend = pxa2xx_ac97_suspend, - .resume = pxa2xx_ac97_resume, .playback = { .stream_name = "AC97 Playback", .channels_min = 2, @@ -246,6 +215,12 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return -ENXIO; } + ret = pxa2xx_ac97_hw_probe(pdev); + if (ret) { + dev_err(&pdev->dev, "PXA2xx AC97 hw probe error (%d)\n", ret); + return ret; + } + ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); if (ret != 0) return ret; @@ -262,15 +237,34 @@ static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { snd_soc_unregister_component(&pdev->dev); snd_soc_set_ac97_ops(NULL); + pxa2xx_ac97_hw_remove(pdev); return 0; } +#ifdef CONFIG_PM_SLEEP +static int pxa2xx_ac97_dev_suspend(struct device *dev) +{ + return pxa2xx_ac97_hw_suspend(); +} + +static int pxa2xx_ac97_dev_resume(struct device *dev) +{ + return pxa2xx_ac97_hw_resume(); +} + +static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, + pxa2xx_ac97_dev_suspend, pxa2xx_ac97_dev_resume); +#endif + static struct platform_driver pxa2xx_ac97_driver = { .probe = pxa2xx_ac97_dev_probe, .remove = pxa2xx_ac97_dev_remove, .driver = { .name = "pxa2xx-ac97", .owner = THIS_MODULE, +#ifdef CONFIG_PM_SLEEP + .pm = &pxa2xx_ac97_pm_ops, +#endif }, }; diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index a3fe19123f0..1d9c2ed223b 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -275,6 +275,7 @@ static struct platform_driver tosa_driver = { .driver = { .name = "tosa-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = tosa_probe, .remove = tosa_remove, diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index 13c9ee0cb83..0b535b57062 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -160,6 +160,7 @@ static struct platform_driver ttc_dkb_driver = { .driver = { .name = "ttc-dkb-audio", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, }, .probe = ttc_dkb_probe, .remove = ttc_dkb_remove, diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c index 29e24680362..84f5d8b7667 100644 --- a/sound/soc/samsung/bells.c +++ b/sound/soc/samsung/bells.c @@ -356,6 +356,7 @@ static struct snd_soc_dapm_widget bells_widgets[] = { static struct snd_soc_dapm_route bells_routes[] = { { "Sub CLK_SYS", NULL, "OPCLK" }, + { "CLKIN", NULL, "OPCLK" }, { "DMIC", NULL, "MICBIAS2" }, { "IN2L", NULL, "DMIC" }, diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index b302f3b7a58..a5cbdb4f165 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -702,13 +702,6 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, } writel(mod, i2s->addr + I2SMOD); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dai_set_dma_data(dai, substream, - (void *)&i2s->dma_playback); - else - snd_soc_dai_set_dma_data(dai, substream, - (void *)&i2s->dma_capture); - i2s->frmclk = params_rate(params); return 0; @@ -970,6 +963,8 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); + snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + if (other) { other->addr = i2s->addr; other->clk = i2s->clk; @@ -1060,7 +1055,7 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) i2s->i2s_dai_drv.ops = &samsung_i2s_dai_ops; i2s->i2s_dai_drv.suspend = i2s_suspend; i2s->i2s_dai_drv.resume = i2s_resume; - i2s->i2s_dai_drv.playback.channels_min = 2; + i2s->i2s_dai_drv.playback.channels_min = 1; i2s->i2s_dai_drv.playback.channels_max = 2; i2s->i2s_dai_drv.playback.rates = SAMSUNG_I2S_RATES; i2s->i2s_dai_drv.playback.formats = SAMSUNG_I2S_FMTS; @@ -1073,7 +1068,7 @@ static struct i2s_dai *i2s_alloc_dai(struct platform_device *pdev, bool sec) dev_set_drvdata(&i2s->pdev->dev, i2s); } else { /* Create a new platform_device for Secondary */ i2s->pdev = platform_device_alloc("samsung-i2s-sec", -1); - if (IS_ERR(i2s->pdev)) + if (!i2s->pdev) return NULL; i2s->pdev->dev.parent = &pdev->dev; @@ -1143,9 +1138,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Unable to get drvdata\n"); return -EFAULT; } - snd_soc_register_component(&sec_dai->pdev->dev, - &samsung_i2s_component, - &sec_dai->i2s_dai_drv, 1); + devm_snd_soc_register_component(&sec_dai->pdev->dev, + &samsung_i2s_component, + &sec_dai->i2s_dai_drv, 1); samsung_asoc_dma_platform_register(&pdev->dev); return 0; } @@ -1258,8 +1253,9 @@ static int samsung_i2s_probe(struct platform_device *pdev) goto err; } - snd_soc_register_component(&pri_dai->pdev->dev, &samsung_i2s_component, - &pri_dai->i2s_dai_drv, 1); + devm_snd_soc_register_component(&pri_dai->pdev->dev, + &samsung_i2s_component, + &pri_dai->i2s_dai_drv, 1); pm_runtime_enable(&pdev->dev); @@ -1294,7 +1290,6 @@ static int samsung_i2s_remove(struct platform_device *pdev) i2s->sec_dai = NULL; samsung_asoc_dma_platform_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 5fd7a05a9b9..b072bd107b3 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -9,6 +9,7 @@ #include "../codecs/wm8994.h" #include <sound/pcm_params.h> +#include <sound/soc.h> #include <linux/module.h> #include <linux/of.h> #include <linux/of_device.h> @@ -193,7 +194,7 @@ static int smdk_audio_probe(struct platform_device *pdev) platform_set_drvdata(pdev, board); - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); @@ -201,23 +202,14 @@ static int smdk_audio_probe(struct platform_device *pdev) return ret; } -static int smdk_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static struct platform_driver smdk_audio_driver = { .driver = { .name = "smdk-audio-wm8894", .owner = THIS_MODULE, .of_match_table = of_match_ptr(samsung_wm8994_of_match), + .pm = &snd_soc_pm_ops, }, .probe = smdk_audio_probe, - .remove = smdk_audio_remove, }; module_platform_driver(smdk_audio_driver); diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 56d8ff6a402..14011d90d70 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -37,7 +37,6 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" select SND_SIMPLE_CARD - select RCAR_CLK_ADG help This option enables R-Car SUR/SCU/SSIU/SSI sound support diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index d80deb7ccf1..9430097979a 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -8,7 +8,6 @@ * for more details. */ #include <linux/sh_clk.h> -#include <mach/clock.h> #include "rsnd.h" #define CLKA 0 @@ -22,6 +21,7 @@ struct rsnd_adg { int rate_of_441khz_div_6; int rate_of_48khz_div_6; + u32 ckr; }; #define for_each_rsnd_clk(pos, adg, i) \ @@ -116,6 +116,11 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *mod, unsigned int rate) found_clock: + /* see rsnd_adg_ssi_clk_init() */ + rsnd_mod_bset(mod, SSICKR, 0x00FF0000, adg->ckr); + rsnd_mod_write(mod, BRRA, 0x00000002); /* 1/6 */ + rsnd_mod_write(mod, BRRB, 0x00000002); /* 1/6 */ + /* * This "mod" = "ssi" here. * we can get "ssi id" from mod @@ -182,9 +187,7 @@ static void rsnd_adg_ssi_clk_init(struct rsnd_priv *priv, struct rsnd_adg *adg) } } - rsnd_priv_bset(priv, SSICKR, 0x00FF0000, ckr); - rsnd_priv_write(priv, BRRA, 0x00000002); /* 1/6 */ - rsnd_priv_write(priv, BRRB, 0x00000002); /* 1/6 */ + adg->ckr = ckr; } int rsnd_adg_probe(struct platform_device *pdev, diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index a3570602851..78c35b44fc0 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -94,6 +94,7 @@ * */ #include <linux/pm_runtime.h> +#include <linux/shdma-base.h> #include "rsnd.h" #define RSND_RATES SNDRV_PCM_RATE_8000_96000 @@ -103,54 +104,9 @@ * rsnd_platform functions */ #define rsnd_platform_call(priv, dai, func, param...) \ - (!(priv->info->func) ? -ENODEV : \ + (!(priv->info->func) ? 0 : \ priv->info->func(param)) - -/* - * basic function - */ -u32 rsnd_read(struct rsnd_priv *priv, - struct rsnd_mod *mod, enum rsnd_reg reg) -{ - void __iomem *base = rsnd_gen_reg_get(priv, mod, reg); - - BUG_ON(!base); - - return ioread32(base); -} - -void rsnd_write(struct rsnd_priv *priv, - struct rsnd_mod *mod, - enum rsnd_reg reg, u32 data) -{ - void __iomem *base = rsnd_gen_reg_get(priv, mod, reg); - struct device *dev = rsnd_priv_to_dev(priv); - - BUG_ON(!base); - - dev_dbg(dev, "w %p : %08x\n", base, data); - - iowrite32(data, base); -} - -void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, - enum rsnd_reg reg, u32 mask, u32 data) -{ - void __iomem *base = rsnd_gen_reg_get(priv, mod, reg); - struct device *dev = rsnd_priv_to_dev(priv); - u32 val; - - BUG_ON(!base); - - val = ioread32(base); - val &= ~mask; - val |= data & mask; - iowrite32(val, base); - - dev_dbg(dev, "s %p : %08x\n", base, val); -} - /* * rsnd_mod functions */ @@ -254,13 +210,6 @@ int rsnd_dma_available(struct rsnd_dma *dma) return !!dma->chan; } -static bool rsnd_dma_filter(struct dma_chan *chan, void *param) -{ - chan->private = param; - - return true; -} - int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int is_play, int id, int (*inquiry)(struct rsnd_dma *dma, @@ -268,7 +217,9 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, int (*complete)(struct rsnd_dma *dma)) { struct device *dev = rsnd_priv_to_dev(priv); + struct dma_slave_config cfg; dma_cap_mask_t mask; + int ret; if (dma->chan) { dev_err(dev, "it already has dma channel\n"); @@ -278,15 +229,23 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, dma_cap_zero(mask); dma_cap_set(DMA_SLAVE, mask); - dma->slave.shdma_slave.slave_id = id; - - dma->chan = dma_request_channel(mask, rsnd_dma_filter, - &dma->slave.shdma_slave); + dma->chan = dma_request_slave_channel_compat(mask, shdma_chan_filter, + (void *)id, dev, + is_play ? "tx" : "rx"); if (!dma->chan) { dev_err(dev, "can't get dma channel\n"); return -EIO; } + cfg.slave_id = id; + cfg.dst_addr = 0; /* use default addr when playback */ + cfg.src_addr = 0; /* use default addr when capture */ + cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + + ret = dmaengine_slave_config(dma->chan, &cfg); + if (ret < 0) + goto rsnd_dma_init_err; + dma->dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; dma->priv = priv; dma->inquiry = inquiry; @@ -294,6 +253,11 @@ int rsnd_dma_init(struct rsnd_priv *priv, struct rsnd_dma *dma, INIT_WORK(&dma->work, rsnd_dma_do_work); return 0; + +rsnd_dma_init_err: + rsnd_dma_quit(priv, dma); + + return ret; } void rsnd_dma_quit(struct rsnd_priv *priv, @@ -363,6 +327,9 @@ int rsnd_dai_id(struct rsnd_priv *priv, struct rsnd_dai *rdai) struct rsnd_dai *rsnd_dai_get(struct rsnd_priv *priv, int id) { + if ((id < 0) || (id >= rsnd_dai_nr(priv))) + return NULL; + return priv->rdai + id; } diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index babb203b43b..61212ee97c2 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -11,6 +11,11 @@ #include "rsnd.h" struct rsnd_gen_ops { + int (*probe)(struct platform_device *pdev, + struct rcar_snd_info *info, + struct rsnd_priv *priv); + void (*remove)(struct platform_device *pdev, + struct rsnd_priv *priv); int (*path_init)(struct rsnd_priv *priv, struct rsnd_dai *rdai, struct rsnd_dai_stream *io); @@ -19,21 +24,97 @@ struct rsnd_gen_ops { struct rsnd_dai_stream *io); }; -struct rsnd_gen_reg_map { - int index; /* -1 : not supported */ - u32 offset_id; /* offset of ssi0, ssi1, ssi2... */ - u32 offset_adr; /* offset of SSICR, SSISR, ... */ -}; - struct rsnd_gen { void __iomem *base[RSND_BASE_MAX]; - struct rsnd_gen_reg_map reg_map[RSND_REG_MAX]; struct rsnd_gen_ops *ops; + + struct regmap *regmap; + struct regmap_field *regs[RSND_REG_MAX]; }; #define rsnd_priv_to_gen(p) ((struct rsnd_gen *)(p)->gen) +#define RSND_REG_SET(gen, id, reg_id, offset, _id_offset, _id_size) \ + [id] = { \ + .reg = (unsigned int)gen->base[reg_id] + offset, \ + .lsb = 0, \ + .msb = 31, \ + .id_size = _id_size, \ + .id_offset = _id_offset, \ + } + +/* + * basic function + */ +static int rsnd_regmap_write32(void *context, const void *_data, size_t count) +{ + struct rsnd_priv *priv = context; + struct device *dev = rsnd_priv_to_dev(priv); + u32 *data = (u32 *)_data; + u32 val = data[1]; + void __iomem *reg = (void *)data[0]; + + iowrite32(val, reg); + + dev_dbg(dev, "w %p : %08x\n", reg, val); + + return 0; +} + +static int rsnd_regmap_read32(void *context, + const void *_data, size_t reg_size, + void *_val, size_t val_size) +{ + struct rsnd_priv *priv = context; + struct device *dev = rsnd_priv_to_dev(priv); + u32 *data = (u32 *)_data; + u32 *val = (u32 *)_val; + void __iomem *reg = (void *)data[0]; + + *val = ioread32(reg); + + dev_dbg(dev, "r %p : %08x\n", reg, *val); + + return 0; +} + +static struct regmap_bus rsnd_regmap_bus = { + .write = rsnd_regmap_write32, + .read = rsnd_regmap_read32, + .reg_format_endian_default = REGMAP_ENDIAN_NATIVE, + .val_format_endian_default = REGMAP_ENDIAN_NATIVE, +}; + +u32 rsnd_read(struct rsnd_priv *priv, + struct rsnd_mod *mod, enum rsnd_reg reg) +{ + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + u32 val; + + regmap_fields_read(gen->regs[reg], rsnd_mod_id(mod), &val); + + return val; +} + +void rsnd_write(struct rsnd_priv *priv, + struct rsnd_mod *mod, + enum rsnd_reg reg, u32 data) +{ + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + + regmap_fields_write(gen->regs[reg], rsnd_mod_id(mod), data); +} + +void rsnd_bset(struct rsnd_priv *priv, struct rsnd_mod *mod, + enum rsnd_reg reg, u32 mask, u32 data) +{ + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + + regmap_fields_update_bits(gen->regs[reg], rsnd_mod_id(mod), + mask, data); +} + /* * Gen2 * will be filled in the future @@ -98,44 +179,64 @@ static int rsnd_gen1_path_exit(struct rsnd_priv *priv, return ret; } -static struct rsnd_gen_ops rsnd_gen1_ops = { - .path_init = rsnd_gen1_path_init, - .path_exit = rsnd_gen1_path_exit, -}; +/* single address mapping */ +#define RSND_GEN1_S_REG(gen, reg, id, offset) \ + RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, 0, 9) -#define RSND_GEN1_REG_MAP(g, s, i, oi, oa) \ - do { \ - (g)->reg_map[RSND_REG_##i].index = RSND_GEN1_##s; \ - (g)->reg_map[RSND_REG_##i].offset_id = oi; \ - (g)->reg_map[RSND_REG_##i].offset_adr = oa; \ - } while (0) +/* multi address mapping */ +#define RSND_GEN1_M_REG(gen, reg, id, offset, _id_offset) \ + RSND_REG_SET(gen, RSND_REG_##id, RSND_GEN1_##reg, offset, _id_offset, 9) -static void rsnd_gen1_reg_map_init(struct rsnd_gen *gen) +static int rsnd_gen1_regmap_init(struct rsnd_priv *priv, struct rsnd_gen *gen) { - RSND_GEN1_REG_MAP(gen, SRU, SRC_ROUTE_SEL, 0x0, 0x00); - RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL0, 0x0, 0x08); - RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL1, 0x0, 0x0c); - RSND_GEN1_REG_MAP(gen, SRU, SRC_TMG_SEL2, 0x0, 0x10); - RSND_GEN1_REG_MAP(gen, SRU, SRC_CTRL, 0x0, 0xc0); - RSND_GEN1_REG_MAP(gen, SRU, SSI_MODE0, 0x0, 0xD0); - RSND_GEN1_REG_MAP(gen, SRU, SSI_MODE1, 0x0, 0xD4); - RSND_GEN1_REG_MAP(gen, SRU, BUSIF_MODE, 0x4, 0x20); - RSND_GEN1_REG_MAP(gen, SRU, BUSIF_ADINR, 0x40, 0x214); - - RSND_GEN1_REG_MAP(gen, ADG, BRRA, 0x0, 0x00); - RSND_GEN1_REG_MAP(gen, ADG, BRRB, 0x0, 0x04); - RSND_GEN1_REG_MAP(gen, ADG, SSICKR, 0x0, 0x08); - RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL0, 0x0, 0x0c); - RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL1, 0x0, 0x10); - RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL3, 0x0, 0x18); - RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL4, 0x0, 0x1c); - RSND_GEN1_REG_MAP(gen, ADG, AUDIO_CLK_SEL5, 0x0, 0x20); - - RSND_GEN1_REG_MAP(gen, SSI, SSICR, 0x40, 0x00); - RSND_GEN1_REG_MAP(gen, SSI, SSISR, 0x40, 0x04); - RSND_GEN1_REG_MAP(gen, SSI, SSITDR, 0x40, 0x08); - RSND_GEN1_REG_MAP(gen, SSI, SSIRDR, 0x40, 0x0c); - RSND_GEN1_REG_MAP(gen, SSI, SSIWSR, 0x40, 0x20); + int i; + struct device *dev = rsnd_priv_to_dev(priv); + struct regmap_config regc; + struct reg_field regf[RSND_REG_MAX] = { + RSND_GEN1_S_REG(gen, SRU, SRC_ROUTE_SEL, 0x00), + RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL0, 0x08), + RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL1, 0x0c), + RSND_GEN1_S_REG(gen, SRU, SRC_TMG_SEL2, 0x10), + RSND_GEN1_S_REG(gen, SRU, SRC_CTRL, 0xc0), + RSND_GEN1_S_REG(gen, SRU, SSI_MODE0, 0xD0), + RSND_GEN1_S_REG(gen, SRU, SSI_MODE1, 0xD4), + RSND_GEN1_M_REG(gen, SRU, BUSIF_MODE, 0x20, 0x4), + RSND_GEN1_M_REG(gen, SRU, BUSIF_ADINR, 0x214, 0x40), + + RSND_GEN1_S_REG(gen, ADG, BRRA, 0x00), + RSND_GEN1_S_REG(gen, ADG, BRRB, 0x04), + RSND_GEN1_S_REG(gen, ADG, SSICKR, 0x08), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL0, 0x0c), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL1, 0x10), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL3, 0x18), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL4, 0x1c), + RSND_GEN1_S_REG(gen, ADG, AUDIO_CLK_SEL5, 0x20), + + RSND_GEN1_M_REG(gen, SSI, SSICR, 0x00, 0x40), + RSND_GEN1_M_REG(gen, SSI, SSISR, 0x04, 0x40), + RSND_GEN1_M_REG(gen, SSI, SSITDR, 0x08, 0x40), + RSND_GEN1_M_REG(gen, SSI, SSIRDR, 0x0c, 0x40), + RSND_GEN1_M_REG(gen, SSI, SSIWSR, 0x20, 0x40), + }; + + memset(®c, 0, sizeof(regc)); + regc.reg_bits = 32; + regc.val_bits = 32; + + gen->regmap = devm_regmap_init(dev, &rsnd_regmap_bus, priv, ®c); + if (IS_ERR(gen->regmap)) { + dev_err(dev, "regmap error %ld\n", PTR_ERR(gen->regmap)); + return PTR_ERR(gen->regmap); + } + + for (i = 0; i < RSND_REG_MAX; i++) { + gen->regs[i] = devm_regmap_field_alloc(dev, gen->regmap, regf[i]); + if (IS_ERR(gen->regs[i])) + return PTR_ERR(gen->regs[i]); + + } + + return 0; } static int rsnd_gen1_probe(struct platform_device *pdev, @@ -147,6 +248,7 @@ static int rsnd_gen1_probe(struct platform_device *pdev, struct resource *sru_res; struct resource *adg_res; struct resource *ssi_res; + int ret; /* * map address @@ -163,8 +265,9 @@ static int rsnd_gen1_probe(struct platform_device *pdev, IS_ERR(gen->base[RSND_GEN1_SSI])) return -ENODEV; - gen->ops = &rsnd_gen1_ops; - rsnd_gen1_reg_map_init(gen); + ret = rsnd_gen1_regmap_init(priv, gen); + if (ret < 0) + return ret; dev_dbg(dev, "Gen1 device probed\n"); dev_dbg(dev, "SRU : %08x => %p\n", sru_res->start, @@ -183,6 +286,13 @@ static void rsnd_gen1_remove(struct platform_device *pdev, { } +static struct rsnd_gen_ops rsnd_gen1_ops = { + .probe = rsnd_gen1_probe, + .remove = rsnd_gen1_remove, + .path_init = rsnd_gen1_path_init, + .path_exit = rsnd_gen1_path_exit, +}; + /* * Gen */ @@ -204,46 +314,12 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv, return gen->ops->path_exit(priv, rdai, io); } -void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, - struct rsnd_mod *mod, - enum rsnd_reg reg) -{ - struct rsnd_gen *gen = rsnd_priv_to_gen(priv); - struct device *dev = rsnd_priv_to_dev(priv); - int index; - u32 offset_id, offset_adr; - - if (reg >= RSND_REG_MAX) { - dev_err(dev, "rsnd_reg reg error\n"); - return NULL; - } - - index = gen->reg_map[reg].index; - offset_id = gen->reg_map[reg].offset_id; - offset_adr = gen->reg_map[reg].offset_adr; - - if (index < 0) { - dev_err(dev, "unsupported reg access %d\n", reg); - return NULL; - } - - if (offset_id && mod) - offset_id *= rsnd_mod_id(mod); - - /* - * index/offset were set on gen1/gen2 - */ - - return gen->base[index] + offset_id + offset_adr; -} - int rsnd_gen_probe(struct platform_device *pdev, struct rcar_snd_info *info, struct rsnd_priv *priv) { struct device *dev = rsnd_priv_to_dev(priv); struct rsnd_gen *gen; - int i; gen = devm_kzalloc(dev, sizeof(*gen), GFP_KERNEL); if (!gen) { @@ -251,30 +327,23 @@ int rsnd_gen_probe(struct platform_device *pdev, return -ENOMEM; } - priv->gen = gen; - - /* - * see - * rsnd_reg_get() - * rsnd_gen_probe() - */ - for (i = 0; i < RSND_REG_MAX; i++) - gen->reg_map[i].index = -1; - - /* - * init each module - */ if (rsnd_is_gen1(priv)) - return rsnd_gen1_probe(pdev, info, priv); + gen->ops = &rsnd_gen1_ops; - dev_err(dev, "unknown generation R-Car sound device\n"); + if (!gen->ops) { + dev_err(dev, "unknown generation R-Car sound device\n"); + return -ENODEV; + } - return -ENODEV; + priv->gen = gen; + + return gen->ops->probe(pdev, info, priv); } void rsnd_gen_remove(struct platform_device *pdev, struct rsnd_priv *priv) { - if (rsnd_is_gen1(priv)) - rsnd_gen1_remove(pdev, priv); + struct rsnd_gen *gen = rsnd_priv_to_gen(priv); + + gen->ops->remove(pdev, priv); } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 5dd87f4c919..9e463e50e7e 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -78,10 +78,6 @@ struct rsnd_dai_stream; #define rsnd_mod_bset(m, r, s, d) \ rsnd_bset(rsnd_mod_to_priv(m), m, RSND_REG_##r, s, d) -#define rsnd_priv_read(p, r) rsnd_read(p, NULL, RSND_REG_##r) -#define rsnd_priv_write(p, r, d) rsnd_write(p, NULL, RSND_REG_##r, d) -#define rsnd_priv_bset(p, r, s, d) rsnd_bset(p, NULL, RSND_REG_##r, s, d) - u32 rsnd_read(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); void rsnd_write(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg, u32 data); @@ -285,6 +281,7 @@ int rsnd_scu_probe(struct platform_device *pdev, void rsnd_scu_remove(struct platform_device *pdev, struct rsnd_priv *priv); struct rsnd_mod *rsnd_scu_mod_get(struct rsnd_priv *priv, int id); +bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod); #define rsnd_scu_nr(priv) ((priv)->scu_nr) /* diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 92e3f51c3a4..f4453e33a84 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -146,20 +146,26 @@ static int rsnd_scu_set_hpbif(struct rsnd_priv *priv, return 0; } +bool rsnd_scu_hpbif_is_enable(struct rsnd_mod *mod) +{ + struct rsnd_scu *scu = rsnd_mod_to_scu(mod); + u32 flags = rsnd_scu_mode_flags(scu); + + return !!(flags & RSND_SCU_USE_HPBIF); +} + static int rsnd_scu_start(struct rsnd_mod *mod, struct rsnd_dai *rdai, struct rsnd_dai_stream *io) { struct rsnd_priv *priv = rsnd_mod_to_priv(mod); - struct rsnd_scu *scu = rsnd_mod_to_scu(mod); struct device *dev = rsnd_priv_to_dev(priv); - u32 flags = rsnd_scu_mode_flags(scu); int ret; /* * SCU will be used if it has RSND_SCU_USE_HPBIF flags */ - if (!(flags & RSND_SCU_USE_HPBIF)) { + if (!rsnd_scu_hpbif_is_enable(mod)) { /* it use PIO transter */ dev_dbg(dev, "%s%d is not used\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index fc010d65206..5ac20cd5e00 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -101,29 +101,30 @@ struct rsnd_ssiu { #define rsnd_ssi_to_ssiu(ssi)\ (((struct rsnd_ssiu *)((ssi) - rsnd_mod_id(&(ssi)->mod))) - 1) -static void rsnd_ssi_mode_init(struct rsnd_priv *priv, - struct rsnd_ssiu *ssiu) +static void rsnd_ssi_mode_set(struct rsnd_priv *priv, + struct rsnd_dai *rdai, + struct rsnd_ssi *ssi) { struct device *dev = rsnd_priv_to_dev(priv); - struct rsnd_ssi *ssi; + struct rsnd_mod *scu; + struct rsnd_ssiu *ssiu = rsnd_ssi_to_ssiu(ssi); + int id = rsnd_mod_id(&ssi->mod); u32 flags; u32 val; - int i; + + scu = rsnd_scu_mod_get(priv, rsnd_mod_id(&ssi->mod)); /* * SSI_MODE0 */ - ssiu->ssi_mode0 = 0; - for_each_rsnd_ssi(ssi, priv, i) { - flags = rsnd_ssi_mode_flags(ssi); - - /* see also BUSIF_MODE */ - if (!(flags & RSND_SSI_DEPENDENT)) { - ssiu->ssi_mode0 |= (1 << i); - dev_dbg(dev, "SSI%d uses INDEPENDENT mode\n", i); - } else { - dev_dbg(dev, "SSI%d uses DEPENDENT mode\n", i); - } + + /* see also BUSIF_MODE */ + if (rsnd_scu_hpbif_is_enable(scu)) { + ssiu->ssi_mode0 &= ~(1 << id); + dev_dbg(dev, "SSI%d uses DEPENDENT mode\n", id); + } else { + ssiu->ssi_mode0 |= (1 << id); + dev_dbg(dev, "SSI%d uses INDEPENDENT mode\n", id); } /* @@ -132,7 +133,7 @@ static void rsnd_ssi_mode_init(struct rsnd_priv *priv, #define ssi_parent_set(p, sync, adg, ext) \ do { \ ssi->parent = ssiu->ssi + p; \ - if (flags & RSND_SSI_CLK_FROM_ADG) \ + if (rsnd_rdai_is_clk_master(rdai)) \ val = adg; \ else \ val = ext; \ @@ -140,15 +141,11 @@ static void rsnd_ssi_mode_init(struct rsnd_priv *priv, val |= sync; \ } while (0) - ssiu->ssi_mode1 = 0; - for_each_rsnd_ssi(ssi, priv, i) { - flags = rsnd_ssi_mode_flags(ssi); - - if (!(flags & RSND_SSI_CLK_PIN_SHARE)) - continue; + flags = rsnd_ssi_mode_flags(ssi); + if (flags & RSND_SSI_CLK_PIN_SHARE) { val = 0; - switch (i) { + switch (id) { case 1: ssi_parent_set(0, (1 << 4), (0x2 << 0), (0x1 << 0)); break; @@ -165,11 +162,6 @@ static void rsnd_ssi_mode_init(struct rsnd_priv *priv, ssiu->ssi_mode1 |= val; } -} - -static void rsnd_ssi_mode_set(struct rsnd_ssi *ssi) -{ - struct rsnd_ssiu *ssiu = rsnd_ssi_to_ssiu(ssi); rsnd_mod_write(&ssi->mod, SSI_MODE0, ssiu->ssi_mode0); rsnd_mod_write(&ssi->mod, SSI_MODE1, ssiu->ssi_mode1); @@ -379,7 +371,7 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, ssi->cr_own = cr; ssi->err = -1; /* ignore 1st error */ - rsnd_ssi_mode_set(ssi); + rsnd_ssi_mode_set(priv, rdai, ssi); dev_dbg(dev, "%s.%d init\n", rsnd_mod_name(mod), rsnd_mod_id(mod)); @@ -707,8 +699,6 @@ int rsnd_ssi_probe(struct platform_device *pdev, rsnd_mod_init(priv, &ssi->mod, ops, i); } - rsnd_ssi_mode_init(priv, ssiu); - dev_dbg(dev, "ssi probed\n"); return 0; diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 223dc0636fe..375dc6dfba4 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -11,12 +11,9 @@ * option) any later version. */ -#include <linux/i2c.h> -#include <linux/spi/spi.h> #include <sound/soc.h> -#include <linux/bitmap.h> -#include <linux/rbtree.h> #include <linux/export.h> +#include <linux/slab.h> #include <trace/events/asoc.h> @@ -68,126 +65,42 @@ static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx, return -1; } -static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) +int snd_soc_cache_init(struct snd_soc_codec *codec) { - int i; - int ret; - const struct snd_soc_codec_driver *codec_drv; - unsigned int val; + const struct snd_soc_codec_driver *codec_drv = codec->driver; + size_t reg_size; - codec_drv = codec->driver; - for (i = 0; i < codec_drv->reg_cache_size; ++i) { - ret = snd_soc_cache_read(codec, i, &val); - if (ret) - return ret; - if (codec->reg_def_copy) - if (snd_soc_get_cache_val(codec->reg_def_copy, - i, codec_drv->reg_word_size) == val) - continue; + reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; - WARN_ON(!snd_soc_codec_writable_register(codec, i)); - - ret = snd_soc_write(codec, i, val); - if (ret) - return ret; - dev_dbg(codec->dev, "ASoC: Synced register %#x, value = %#x\n", - i, val); - } - return 0; -} - -static int snd_soc_flat_cache_write(struct snd_soc_codec *codec, - unsigned int reg, unsigned int value) -{ - snd_soc_set_cache_val(codec->reg_cache, reg, value, - codec->driver->reg_word_size); - return 0; -} - -static int snd_soc_flat_cache_read(struct snd_soc_codec *codec, - unsigned int reg, unsigned int *value) -{ - *value = snd_soc_get_cache_val(codec->reg_cache, reg, - codec->driver->reg_word_size); - return 0; -} + mutex_init(&codec->cache_rw_mutex); -static int snd_soc_flat_cache_exit(struct snd_soc_codec *codec) -{ - if (!codec->reg_cache) - return 0; - kfree(codec->reg_cache); - codec->reg_cache = NULL; - return 0; -} + dev_dbg(codec->dev, "ASoC: Initializing cache for %s codec\n", + codec->name); -static int snd_soc_flat_cache_init(struct snd_soc_codec *codec) -{ - if (codec->reg_def_copy) - codec->reg_cache = kmemdup(codec->reg_def_copy, - codec->reg_size, GFP_KERNEL); + if (codec_drv->reg_cache_default) + codec->reg_cache = kmemdup(codec_drv->reg_cache_default, + reg_size, GFP_KERNEL); else - codec->reg_cache = kzalloc(codec->reg_size, GFP_KERNEL); + codec->reg_cache = kzalloc(reg_size, GFP_KERNEL); if (!codec->reg_cache) return -ENOMEM; return 0; } -/* an array of all supported compression types */ -static const struct snd_soc_cache_ops cache_types[] = { - /* Flat *must* be the first entry for fallback */ - { - .id = SND_SOC_FLAT_COMPRESSION, - .name = "flat", - .init = snd_soc_flat_cache_init, - .exit = snd_soc_flat_cache_exit, - .read = snd_soc_flat_cache_read, - .write = snd_soc_flat_cache_write, - .sync = snd_soc_flat_cache_sync - }, -}; - -int snd_soc_cache_init(struct snd_soc_codec *codec) -{ - int i; - - for (i = 0; i < ARRAY_SIZE(cache_types); ++i) - if (cache_types[i].id == codec->compress_type) - break; - - /* Fall back to flat compression */ - if (i == ARRAY_SIZE(cache_types)) { - dev_warn(codec->dev, "ASoC: Could not match compress type: %d\n", - codec->compress_type); - i = 0; - } - - mutex_init(&codec->cache_rw_mutex); - codec->cache_ops = &cache_types[i]; - - if (codec->cache_ops->init) { - if (codec->cache_ops->name) - dev_dbg(codec->dev, "ASoC: Initializing %s cache for %s codec\n", - codec->cache_ops->name, codec->name); - return codec->cache_ops->init(codec); - } - return -ENOSYS; -} - /* * NOTE: keep in mind that this function might be called * multiple times. */ int snd_soc_cache_exit(struct snd_soc_codec *codec) { - if (codec->cache_ops && codec->cache_ops->exit) { - if (codec->cache_ops->name) - dev_dbg(codec->dev, "ASoC: Destroying %s cache for %s codec\n", - codec->cache_ops->name, codec->name); - return codec->cache_ops->exit(codec); - } - return -ENOSYS; + dev_dbg(codec->dev, "ASoC: Destroying cache for %s codec\n", + codec->name); + if (!codec->reg_cache) + return 0; + kfree(codec->reg_cache); + codec->reg_cache = NULL; + return 0; } /** @@ -200,18 +113,15 @@ int snd_soc_cache_exit(struct snd_soc_codec *codec) int snd_soc_cache_read(struct snd_soc_codec *codec, unsigned int reg, unsigned int *value) { - int ret; + if (!value) + return -EINVAL; mutex_lock(&codec->cache_rw_mutex); - - if (value && codec->cache_ops && codec->cache_ops->read) { - ret = codec->cache_ops->read(codec, reg, value); - mutex_unlock(&codec->cache_rw_mutex); - return ret; - } - + *value = snd_soc_get_cache_val(codec->reg_cache, reg, + codec->driver->reg_word_size); mutex_unlock(&codec->cache_rw_mutex); - return -ENOSYS; + + return 0; } EXPORT_SYMBOL_GPL(snd_soc_cache_read); @@ -225,20 +135,42 @@ EXPORT_SYMBOL_GPL(snd_soc_cache_read); int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { + mutex_lock(&codec->cache_rw_mutex); + snd_soc_set_cache_val(codec->reg_cache, reg, value, + codec->driver->reg_word_size); + mutex_unlock(&codec->cache_rw_mutex); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_cache_write); + +static int snd_soc_flat_cache_sync(struct snd_soc_codec *codec) +{ + int i; int ret; + const struct snd_soc_codec_driver *codec_drv; + unsigned int val; - mutex_lock(&codec->cache_rw_mutex); + codec_drv = codec->driver; + for (i = 0; i < codec_drv->reg_cache_size; ++i) { + ret = snd_soc_cache_read(codec, i, &val); + if (ret) + return ret; + if (codec_drv->reg_cache_default) + if (snd_soc_get_cache_val(codec_drv->reg_cache_default, + i, codec_drv->reg_word_size) == val) + continue; - if (codec->cache_ops && codec->cache_ops->write) { - ret = codec->cache_ops->write(codec, reg, value); - mutex_unlock(&codec->cache_rw_mutex); - return ret; - } + WARN_ON(!snd_soc_codec_writable_register(codec, i)); - mutex_unlock(&codec->cache_rw_mutex); - return -ENOSYS; + ret = snd_soc_write(codec, i, val); + if (ret) + return ret; + dev_dbg(codec->dev, "ASoC: Synced register %#x, value = %#x\n", + i, val); + } + return 0; } -EXPORT_SYMBOL_GPL(snd_soc_cache_write); /** * snd_soc_cache_sync: Sync the register cache with the hardware. @@ -251,92 +183,19 @@ EXPORT_SYMBOL_GPL(snd_soc_cache_write); */ int snd_soc_cache_sync(struct snd_soc_codec *codec) { + const char *name = "flat"; int ret; - const char *name; - if (!codec->cache_sync) { + if (!codec->cache_sync) return 0; - } - - if (!codec->cache_ops || !codec->cache_ops->sync) - return -ENOSYS; - if (codec->cache_ops->name) - name = codec->cache_ops->name; - else - name = "unknown"; - - if (codec->cache_ops->name) - dev_dbg(codec->dev, "ASoC: Syncing %s cache for %s codec\n", - codec->cache_ops->name, codec->name); + dev_dbg(codec->dev, "ASoC: Syncing cache for %s codec\n", + codec->name); trace_snd_soc_cache_sync(codec, name, "start"); - ret = codec->cache_ops->sync(codec); + ret = snd_soc_flat_cache_sync(codec); if (!ret) codec->cache_sync = 0; trace_snd_soc_cache_sync(codec, name, "end"); return ret; } EXPORT_SYMBOL_GPL(snd_soc_cache_sync); - -static int snd_soc_get_reg_access_index(struct snd_soc_codec *codec, - unsigned int reg) -{ - const struct snd_soc_codec_driver *codec_drv; - unsigned int min, max, index; - - codec_drv = codec->driver; - min = 0; - max = codec_drv->reg_access_size - 1; - do { - index = (min + max) / 2; - if (codec_drv->reg_access_default[index].reg == reg) - return index; - if (codec_drv->reg_access_default[index].reg < reg) - min = index + 1; - else - max = index; - } while (min <= max); - return -1; -} - -int snd_soc_default_volatile_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].vol; -} -EXPORT_SYMBOL_GPL(snd_soc_default_volatile_register); - -int snd_soc_default_readable_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].read; -} -EXPORT_SYMBOL_GPL(snd_soc_default_readable_register); - -int snd_soc_default_writable_register(struct snd_soc_codec *codec, - unsigned int reg) -{ - int index; - - if (reg >= codec->driver->reg_cache_size) - return 1; - index = snd_soc_get_reg_access_index(codec, reg); - if (index < 0) - return 0; - return codec->driver->reg_access_default[index].write; -} -EXPORT_SYMBOL_GPL(snd_soc_default_writable_register); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1a38be0d0ca..4e53d87e881 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -662,6 +662,8 @@ int snd_soc_suspend(struct device *dev) codec->cache_sync = 1; if (codec->using_regmap) regcache_mark_dirty(codec->control_data); + /* deactivate pins to sleep state */ + pinctrl_pm_select_sleep_state(codec->dev); break; default: dev_dbg(codec->dev, @@ -679,6 +681,9 @@ int snd_soc_suspend(struct device *dev) if (cpu_dai->driver->suspend && cpu_dai->driver->ac97_control) cpu_dai->driver->suspend(cpu_dai); + + /* deactivate pins to sleep state */ + pinctrl_pm_select_sleep_state(cpu_dai->dev); } if (card->suspend_post) @@ -807,6 +812,16 @@ int snd_soc_resume(struct device *dev) if (list_empty(&card->codec_dev_list)) return 0; + /* activate pins from sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; + if (cpu_dai->active) + pinctrl_pm_select_default_state(cpu_dai->dev); + if (codec_dai->active) + pinctrl_pm_select_default_state(codec_dai->dev); + } + /* AC97 devices might have other drivers hanging off them so * need to resume immediately. Other drivers don't have that * problem and may take a substantial amount of time to resume @@ -1589,17 +1604,13 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) soc_remove_codec(codec); } -static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, - enum snd_soc_compress_type compress_type) +static int snd_soc_init_codec_cache(struct snd_soc_codec *codec) { int ret; if (codec->cache_init) return 0; - /* override the compress_type if necessary */ - if (compress_type && codec->compress_type != compress_type) - codec->compress_type = compress_type; ret = snd_soc_cache_init(codec); if (ret < 0) { dev_err(codec->dev, @@ -1614,8 +1625,6 @@ static int snd_soc_init_codec_cache(struct snd_soc_codec *codec, static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; - struct snd_soc_codec_conf *codec_conf; - enum snd_soc_compress_type compress_type; struct snd_soc_dai_link *dai_link; int ret, i, order, dai_fmt; @@ -1639,19 +1648,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) list_for_each_entry(codec, &codec_list, list) { if (codec->cache_init) continue; - /* by default we don't override the compress_type */ - compress_type = 0; - /* check to see if we need to override the compress_type */ - for (i = 0; i < card->num_configs; ++i) { - codec_conf = &card->codec_conf[i]; - if (!strcmp(codec->name, codec_conf->dev_name)) { - compress_type = codec_conf->compress_type; - if (compress_type && compress_type - != codec->compress_type) - break; - } - } - ret = snd_soc_init_codec_cache(codec, compress_type); + ret = snd_soc_init_codec_cache(codec); if (ret < 0) goto base_error; } @@ -1947,6 +1944,14 @@ int snd_soc_poweroff(struct device *dev) snd_soc_dapm_shutdown(card); + /* deactivate pins to sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; + pinctrl_pm_select_sleep_state(codec_dai->dev); + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_poweroff); @@ -2297,13 +2302,6 @@ unsigned int snd_soc_write(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_write); -unsigned int snd_soc_bulk_write_raw(struct snd_soc_codec *codec, - unsigned int reg, const void *data, size_t len) -{ - return codec->bulk_write_raw(codec, reg, data, len); -} -EXPORT_SYMBOL_GPL(snd_soc_bulk_write_raw); - /** * snd_soc_update_bits - update codec register bits * @codec: audio codec @@ -2576,8 +2574,9 @@ int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, if (uinfo->value.enumerated.item > e->max - 1) uinfo->value.enumerated.item = e->max - 1; - strcpy(uinfo->value.enumerated.name, - e->texts[uinfo->value.enumerated.item]); + strlcpy(uinfo->value.enumerated.name, + e->texts[uinfo->value.enumerated.item], + sizeof(uinfo->value.enumerated.name)); return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_enum_double); @@ -3576,6 +3575,22 @@ int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** + * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. + * @dai: DAI + * @ratio Ratio of BCLK to Sample rate. + * + * Configures the DAI for a preset BCLK to sample rate ratio. + */ +int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + if (dai->driver && dai->driver->ops->set_bclk_ratio) + return dai->driver->ops->set_bclk_ratio(dai, ratio); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio); + +/** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI * @fmt: SND_SOC_DAIFMT_ format value. @@ -3775,6 +3790,16 @@ int snd_soc_register_card(struct snd_soc_card *card) if (ret != 0) soc_cleanup_card_debugfs(card); + /* deactivate pins to sleep state */ + for (i = 0; i < card->num_rtd; i++) { + struct snd_soc_dai *cpu_dai = card->rtd[i].cpu_dai; + struct snd_soc_dai *codec_dai = card->rtd[i].codec_dai; + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); + } + return ret; } EXPORT_SYMBOL_GPL(snd_soc_register_card); @@ -4020,6 +4045,113 @@ static void snd_soc_unregister_dais(struct device *dev, size_t count) } /** + * snd_soc_register_component - Register a component with the ASoC core + * + */ +static int +__snd_soc_register_component(struct device *dev, + struct snd_soc_component *cmpnt, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, + int num_dai, bool allow_single_dai) +{ + int ret; + + dev_dbg(dev, "component register %s\n", dev_name(dev)); + + if (!cmpnt) { + dev_err(dev, "ASoC: Failed to connecting component\n"); + return -ENOMEM; + } + + cmpnt->name = fmt_single_name(dev, &cmpnt->id); + if (!cmpnt->name) { + dev_err(dev, "ASoC: Failed to simplifying name\n"); + return -ENOMEM; + } + + cmpnt->dev = dev; + cmpnt->driver = cmpnt_drv; + cmpnt->dai_drv = dai_drv; + cmpnt->num_dai = num_dai; + + /* + * snd_soc_register_dai() uses fmt_single_name(), and + * snd_soc_register_dais() uses fmt_multiple_name() + * for dai->name which is used for name based matching + * + * this function is used from cpu/codec. + * allow_single_dai flag can ignore "codec" driver reworking + * since it had been used snd_soc_register_dais(), + */ + if ((1 == num_dai) && allow_single_dai) + ret = snd_soc_register_dai(dev, dai_drv); + else + ret = snd_soc_register_dais(dev, dai_drv, num_dai); + if (ret < 0) { + dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); + goto error_component_name; + } + + mutex_lock(&client_mutex); + list_add(&cmpnt->list, &component_list); + mutex_unlock(&client_mutex); + + dev_dbg(cmpnt->dev, "ASoC: Registered component '%s'\n", cmpnt->name); + + return ret; + +error_component_name: + kfree(cmpnt->name); + + return ret; +} + +int snd_soc_register_component(struct device *dev, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, + int num_dai) +{ + struct snd_soc_component *cmpnt; + + cmpnt = devm_kzalloc(dev, sizeof(*cmpnt), GFP_KERNEL); + if (!cmpnt) { + dev_err(dev, "ASoC: Failed to allocate memory\n"); + return -ENOMEM; + } + + return __snd_soc_register_component(dev, cmpnt, cmpnt_drv, + dai_drv, num_dai, true); +} +EXPORT_SYMBOL_GPL(snd_soc_register_component); + +/** + * snd_soc_unregister_component - Unregister a component from the ASoC core + * + */ +void snd_soc_unregister_component(struct device *dev) +{ + struct snd_soc_component *cmpnt; + + list_for_each_entry(cmpnt, &component_list, list) { + if (dev == cmpnt->dev) + goto found; + } + return; + +found: + snd_soc_unregister_dais(dev, cmpnt->num_dai); + + mutex_lock(&client_mutex); + list_del(&cmpnt->list); + mutex_unlock(&client_mutex); + + dev_dbg(dev, "ASoC: Unregistered component '%s'\n", cmpnt->name); + kfree(cmpnt->name); +} +EXPORT_SYMBOL_GPL(snd_soc_unregister_component); + +/** * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform * @platform: The platform to add @@ -4165,7 +4297,6 @@ int snd_soc_register_codec(struct device *dev, struct snd_soc_dai_driver *dai_drv, int num_dai) { - size_t reg_size; struct snd_soc_codec *codec; int ret, i; @@ -4182,11 +4313,6 @@ int snd_soc_register_codec(struct device *dev, goto fail_codec; } - if (codec_drv->compress_type) - codec->compress_type = codec_drv->compress_type; - else - codec->compress_type = SND_SOC_FLAT_COMPRESSION; - codec->write = codec_drv->write; codec->read = codec_drv->read; codec->volatile_register = codec_drv->volatile_register; @@ -4203,35 +4329,6 @@ int snd_soc_register_codec(struct device *dev, codec->num_dai = num_dai; mutex_init(&codec->mutex); - /* allocate CODEC register cache */ - if (codec_drv->reg_cache_size && codec_drv->reg_word_size) { - reg_size = codec_drv->reg_cache_size * codec_drv->reg_word_size; - codec->reg_size = reg_size; - /* it is necessary to make a copy of the default register cache - * because in the case of using a compression type that requires - * the default register cache to be marked as the - * kernel might have freed the array by the time we initialize - * the cache. - */ - if (codec_drv->reg_cache_default) { - codec->reg_def_copy = kmemdup(codec_drv->reg_cache_default, - reg_size, GFP_KERNEL); - if (!codec->reg_def_copy) { - ret = -ENOMEM; - goto fail_codec_name; - } - } - } - - if (codec_drv->reg_access_size && codec_drv->reg_access_default) { - if (!codec->volatile_register) - codec->volatile_register = snd_soc_default_volatile_register; - if (!codec->readable_register) - codec->readable_register = snd_soc_default_readable_register; - if (!codec->writable_register) - codec->writable_register = snd_soc_default_writable_register; - } - for (i = 0; i < num_dai; i++) { fixup_codec_formats(&dai_drv[i].playback); fixup_codec_formats(&dai_drv[i].capture); @@ -4241,10 +4338,12 @@ int snd_soc_register_codec(struct device *dev, list_add(&codec->list, &codec_list); mutex_unlock(&client_mutex); - /* register any DAIs */ - ret = snd_soc_register_dais(dev, dai_drv, num_dai); + /* register component */ + ret = __snd_soc_register_component(dev, &codec->component, + &codec_drv->component_driver, + dai_drv, num_dai, false); if (ret < 0) { - dev_err(codec->dev, "ASoC: Failed to regster DAIs: %d\n", ret); + dev_err(codec->dev, "ASoC: Failed to regster component: %d\n", ret); goto fail_codec_name; } @@ -4279,7 +4378,7 @@ void snd_soc_unregister_codec(struct device *dev) return; found: - snd_soc_unregister_dais(dev, codec->num_dai); + snd_soc_unregister_component(dev); mutex_lock(&client_mutex); list_del(&codec->list); @@ -4288,98 +4387,11 @@ found: dev_dbg(codec->dev, "ASoC: Unregistered codec '%s'\n", codec->name); snd_soc_cache_exit(codec); - kfree(codec->reg_def_copy); kfree(codec->name); kfree(codec); } EXPORT_SYMBOL_GPL(snd_soc_unregister_codec); - -/** - * snd_soc_register_component - Register a component with the ASoC core - * - */ -int snd_soc_register_component(struct device *dev, - const struct snd_soc_component_driver *cmpnt_drv, - struct snd_soc_dai_driver *dai_drv, - int num_dai) -{ - struct snd_soc_component *cmpnt; - int ret; - - dev_dbg(dev, "component register %s\n", dev_name(dev)); - - cmpnt = devm_kzalloc(dev, sizeof(*cmpnt), GFP_KERNEL); - if (!cmpnt) { - dev_err(dev, "ASoC: Failed to allocate memory\n"); - return -ENOMEM; - } - - cmpnt->name = fmt_single_name(dev, &cmpnt->id); - if (!cmpnt->name) { - dev_err(dev, "ASoC: Failed to simplifying name\n"); - return -ENOMEM; - } - - cmpnt->dev = dev; - cmpnt->driver = cmpnt_drv; - cmpnt->num_dai = num_dai; - - /* - * snd_soc_register_dai() uses fmt_single_name(), and - * snd_soc_register_dais() uses fmt_multiple_name() - * for dai->name which is used for name based matching - */ - if (1 == num_dai) - ret = snd_soc_register_dai(dev, dai_drv); - else - ret = snd_soc_register_dais(dev, dai_drv, num_dai); - if (ret < 0) { - dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); - goto error_component_name; - } - - mutex_lock(&client_mutex); - list_add(&cmpnt->list, &component_list); - mutex_unlock(&client_mutex); - - dev_dbg(cmpnt->dev, "ASoC: Registered component '%s'\n", cmpnt->name); - - return ret; - -error_component_name: - kfree(cmpnt->name); - - return ret; -} -EXPORT_SYMBOL_GPL(snd_soc_register_component); - -/** - * snd_soc_unregister_component - Unregister a component from the ASoC core - * - */ -void snd_soc_unregister_component(struct device *dev) -{ - struct snd_soc_component *cmpnt; - - list_for_each_entry(cmpnt, &component_list, list) { - if (dev == cmpnt->dev) - goto found; - } - return; - -found: - snd_soc_unregister_dais(dev, cmpnt->num_dai); - - mutex_lock(&client_mutex); - list_del(&cmpnt->list); - mutex_unlock(&client_mutex); - - dev_dbg(dev, "ASoC: Unregistered component '%s'\n", cmpnt->name); - kfree(cmpnt->name); -} -EXPORT_SYMBOL_GPL(snd_soc_unregister_component); - /* Retrieve a card's name from device tree */ int snd_soc_of_parse_card_name(struct snd_soc_card *card, const char *propname) @@ -4567,6 +4579,60 @@ unsigned int snd_soc_of_parse_daifmt(struct device_node *np, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_daifmt); +int snd_soc_of_get_dai_name(struct device_node *of_node, + const char **dai_name) +{ + struct snd_soc_component *pos; + struct of_phandle_args args; + int ret; + + ret = of_parse_phandle_with_args(of_node, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) + return ret; + + ret = -EPROBE_DEFER; + + mutex_lock(&client_mutex); + list_for_each_entry(pos, &component_list, list) { + if (pos->dev->of_node != args.np) + continue; + + if (pos->driver->of_xlate_dai_name) { + ret = pos->driver->of_xlate_dai_name(pos, &args, dai_name); + } else { + int id = -1; + + switch (args.args_count) { + case 0: + id = 0; /* same as dai_drv[0] */ + break; + case 1: + id = args.args[0]; + break; + default: + /* not supported */ + break; + } + + if (id < 0 || id >= pos->num_dai) { + ret = -EINVAL; + } else { + *dai_name = pos->dai_drv[id].name; + ret = 0; + } + } + + break; + } + mutex_unlock(&client_mutex); + + of_node_put(args.np); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name); + static int __init snd_soc_init(void) { #ifdef CONFIG_DEBUG_FS diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 47dfe17ed4e..5738c19ef14 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -499,18 +499,22 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, int val; struct soc_mixer_control *mc = (struct soc_mixer_control *) w->kcontrol_news[i].private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - val = soc_widget_read(w, reg); - val = (val >> shift) & mask; - if (invert) - val = max - val; + if (reg != SND_SOC_NOPM) { + val = soc_widget_read(w, reg); + val = (val >> shift) & mask; + if (invert) + val = max - val; + p->connect = !!val; + } else { + p->connect = 0; + } - p->connect = !!val; } break; case snd_soc_dapm_mux: { @@ -1840,6 +1844,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) */ switch (w->id) { case snd_soc_dapm_siggen: + case snd_soc_dapm_vmid: break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: @@ -2791,7 +2796,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; @@ -2804,7 +2809,7 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, kcontrol->id.name); mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (dapm_kcontrol_is_powered(kcontrol)) + if (dapm_kcontrol_is_powered(kcontrol) && reg != SND_SOC_NOPM) val = (snd_soc_read(codec, reg) >> shift) & mask; else val = dapm_kcontrol_get_value(kcontrol); @@ -2835,7 +2840,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = codec->card; struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; - unsigned int reg = mc->reg; + int reg = mc->reg; unsigned int shift = mc->shift; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; @@ -2857,19 +2862,24 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - dapm_kcontrol_set_value(kcontrol, val); + change = dapm_kcontrol_set_value(kcontrol, val); - mask = mask << shift; - val = val << shift; + if (reg != SND_SOC_NOPM) { + mask = mask << shift; + val = val << shift; + + change = snd_soc_test_bits(codec, reg, mask, val); + } - change = snd_soc_test_bits(codec, reg, mask, val); if (change) { - update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; + if (reg != SND_SOC_NOPM) { + update.kcontrol = kcontrol; + update.reg = reg; + update.mask = mask; + update.val = val; - card->update = &update; + card->update = &update; + } soc_dapm_mixer_update_power(card, kcontrol, connect); diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c new file mode 100644 index 00000000000..b1d732255c0 --- /dev/null +++ b/sound/soc/soc-devres.c @@ -0,0 +1,86 @@ +/* + * soc-devres.c -- ALSA SoC Audio Layer devres functions + * + * Copyright (C) 2013 Linaro Ltd + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/soc.h> + +static void devm_component_release(struct device *dev, void *res) +{ + snd_soc_unregister_component(*(struct device **)res); +} + +/** + * devm_snd_soc_register_component - resource managed component registration + * @dev: Device used to manage component + * @cmpnt_drv: Component driver + * @dai_drv: DAI driver + * @num_dai: Number of DAIs to register + * + * Register a component with automatic unregistration when the device is + * unregistered. + */ +int devm_snd_soc_register_component(struct device *dev, + const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_dai_driver *dai_drv, int num_dai) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_component_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_soc_register_component(dev, cmpnt_drv, dai_drv, num_dai); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_soc_register_component); + +static void devm_card_release(struct device *dev, void *res) +{ + snd_soc_unregister_card(*(struct snd_soc_card **)res); +} + +/** + * devm_snd_soc_register_card - resource managed card registration + * @dev: Device used to manage card + * @card: Card to register + * + * Register a card with automatic unregistration when the device is + * unregistered. + */ +int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) +{ + struct device **ptr; + int ret; + + ptr = devres_alloc(devm_card_release, sizeof(*ptr), GFP_KERNEL); + if (!ptr) + return -ENOMEM; + + ret = snd_soc_register_card(card); + if (ret == 0) { + *ptr = dev; + devres_add(dev, ptr); + } else { + devres_free(ptr); + } + + return ret; +} +EXPORT_SYMBOL_GPL(devm_snd_soc_register_card); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index e29ec3cd84b..6ad4c7a47f5 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -25,7 +25,7 @@ #include <sound/dmaengine_pcm.h> struct dmaengine_pcm { - struct dma_chan *chan[SNDRV_PCM_STREAM_CAPTURE + 1]; + struct dma_chan *chan[SNDRV_PCM_STREAM_LAST + 1]; const struct snd_dmaengine_pcm_config *config; struct snd_soc_platform platform; unsigned int flags; @@ -36,6 +36,15 @@ static struct dmaengine_pcm *soc_platform_to_pcm(struct snd_soc_platform *p) return container_of(p, struct dmaengine_pcm, platform); } +static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm, + struct snd_pcm_substream *substream) +{ + if (!pcm->chan[substream->stream]) + return NULL; + + return pcm->chan[substream->stream]->device->dev; +} + /** * snd_dmaengine_pcm_prepare_slave_config() - Generic prepare_slave_config callback * @substream: PCM substream @@ -75,12 +84,21 @@ static int dmaengine_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + int (*prepare_slave_config)(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct dma_slave_config *slave_config); struct dma_slave_config slave_config; int ret; - if (pcm->config->prepare_slave_config) { - ret = pcm->config->prepare_slave_config(substream, params, - &slave_config); + memset(&slave_config, 0, sizeof(slave_config)); + + if (!pcm->config) + prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config; + else + prepare_slave_config = pcm->config->prepare_slave_config; + + if (prepare_slave_config) { + ret = prepare_slave_config(substream, params, &slave_config); if (ret) return ret; @@ -92,28 +110,54 @@ static int dmaengine_pcm_hw_params(struct snd_pcm_substream *substream, return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); } -static int dmaengine_pcm_open(struct snd_pcm_substream *substream) +static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct device *dma_dev = dmaengine_dma_dev(pcm, substream); struct dma_chan *chan = pcm->chan[substream->stream]; + struct snd_dmaengine_dai_dma_data *dma_data; + struct dma_slave_caps dma_caps; + struct snd_pcm_hardware hw; int ret; - ret = snd_soc_set_runtime_hwparams(substream, + if (pcm->config && pcm->config->pcm_hardware) + return snd_soc_set_runtime_hwparams(substream, pcm->config->pcm_hardware); - if (ret) - return ret; - return snd_dmaengine_pcm_open(substream, chan); + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + memset(&hw, 0, sizeof(hw)); + hw.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED; + hw.periods_min = 2; + hw.periods_max = UINT_MAX; + hw.period_bytes_min = 256; + hw.period_bytes_max = dma_get_max_seg_size(dma_dev); + hw.buffer_bytes_max = SIZE_MAX; + hw.fifo_size = dma_data->fifo_size; + + ret = dma_get_slave_caps(chan, &dma_caps); + if (ret == 0) { + if (dma_caps.cmd_pause) + hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; + } + + return snd_soc_set_runtime_hwparams(substream, &hw); } -static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm, - struct snd_pcm_substream *substream) +static int dmaengine_pcm_open(struct snd_pcm_substream *substream) { - if (!pcm->chan[substream->stream]) - return NULL; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct dma_chan *chan = pcm->chan[substream->stream]; + int ret; - return pcm->chan[substream->stream]->device->dev; + ret = dmaengine_pcm_set_runtime_hwparams(substream); + if (ret) + return ret; + + return snd_dmaengine_pcm_open(substream, chan); } static void dmaengine_pcm_free(struct snd_pcm *pcm) @@ -126,6 +170,9 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( struct snd_pcm_substream *substream) { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct snd_dmaengine_dai_dma_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) return pcm->chan[0]; @@ -134,22 +181,42 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( return pcm->config->compat_request_channel(rtd, substream); return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn, - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream)); + dma_data->filter_data); } static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); const struct snd_dmaengine_pcm_config *config = pcm->config; + struct device *dev = rtd->platform->dev; + struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_substream *substream; + size_t prealloc_buffer_size; + size_t max_buffer_size; unsigned int i; int ret; + if (config && config->prealloc_buffer_size) { + prealloc_buffer_size = config->prealloc_buffer_size; + max_buffer_size = config->pcm_hardware->buffer_bytes_max; + } else { + prealloc_buffer_size = 512 * 1024; + max_buffer_size = SIZE_MAX; + } + + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { substream = rtd->pcm->streams[i].substream; if (!substream) continue; + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + if (!pcm->chan[i] && + (pcm->flags & SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) + pcm->chan[i] = dma_request_slave_channel(dev, + dma_data->chan_name); + if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, substream); @@ -165,8 +232,8 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) ret = snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV, dmaengine_dma_dev(pcm, substream), - config->prealloc_buffer_size, - config->pcm_hardware->buffer_bytes_max); + prealloc_buffer_size, + max_buffer_size); if (ret) goto err_free; } @@ -222,7 +289,9 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, { unsigned int i; - if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node) + if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || + !dev->of_node) return; if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 122c0c18b9d..4f11d23f206 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -65,31 +65,6 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) return val; } -/* Primitive bulk write support for soc-cache. The data pointed to by - * `data' needs to already be in the form the hardware expects. Any - * data written through this function will not go through the cache as - * it only handles writing to volatile or out of bounds registers. - * - * This is currently only supported for devices using the regmap API - * wrappers. - */ -static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, - unsigned int reg, - const void *data, size_t len) -{ - /* To ensure that we don't get out of sync with the cache, check - * whether the base register is volatile or if we've directly asked - * to bypass the cache. Out of bounds registers are considered - * volatile. - */ - if (!codec->cache_bypass - && !snd_soc_codec_volatile_register(codec, reg) - && reg < codec->driver->reg_cache_size) - return -EINVAL; - - return regmap_raw_write(codec->control_data, reg, data, len); -} - /** * snd_soc_codec_set_cache_io: Set up standard I/O functions. * @@ -119,7 +94,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, memset(&config, 0, sizeof(config)); codec->write = hw_write; codec->read = hw_read; - codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; config.reg_bits = addr_bits; config.val_bits = data_bits; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 330c9a6b5cb..42782c01e41 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -19,6 +19,7 @@ #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> +#include <linux/pinctrl/consumer.h> #include <linux/pm_runtime.h> #include <linux/slab.h> #include <linux/workqueue.h> @@ -183,6 +184,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) struct snd_soc_dai_driver *codec_dai_drv = codec_dai->driver; int ret = 0; + pinctrl_pm_select_default_state(cpu_dai->dev); + pinctrl_pm_select_default_state(codec_dai->dev); pm_runtime_get_sync(cpu_dai->dev); pm_runtime_get_sync(codec_dai->dev); pm_runtime_get_sync(platform->dev); @@ -190,7 +193,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); /* startup the audio subsystem */ - if (cpu_dai->driver->ops->startup) { + if (cpu_dai->driver->ops && cpu_dai->driver->ops->startup) { ret = cpu_dai->driver->ops->startup(substream, cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: can't open interface" @@ -208,7 +211,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } } - if (codec_dai->driver->ops->startup) { + if (codec_dai->driver->ops && codec_dai->driver->ops->startup) { ret = codec_dai->driver->ops->startup(substream, codec_dai); if (ret < 0) { dev_err(codec_dai->dev, "ASoC: can't open codec" @@ -317,6 +320,10 @@ out: pm_runtime_put(platform->dev); pm_runtime_put(codec_dai->dev); pm_runtime_put(cpu_dai->dev); + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); return ret; } @@ -426,6 +433,10 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) pm_runtime_put(platform->dev); pm_runtime_put(codec_dai->dev); pm_runtime_put(cpu_dai->dev); + if (!codec_dai->active) + pinctrl_pm_select_sleep_state(codec_dai->dev); + if (!cpu_dai->active) + pinctrl_pm_select_sleep_state(cpu_dai->dev); return 0; } @@ -463,7 +474,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (codec_dai->driver->ops->prepare) { + if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) { ret = codec_dai->driver->ops->prepare(substream, codec_dai); if (ret < 0) { dev_err(codec_dai->dev, "ASoC: DAI prepare error: %d\n", @@ -472,7 +483,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (cpu_dai->driver->ops->prepare) { + if (cpu_dai->driver->ops && cpu_dai->driver->ops->prepare) { ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: DAI prepare error: %d\n", @@ -523,7 +534,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (codec_dai->driver->ops->hw_params) { + if (codec_dai->driver->ops && codec_dai->driver->ops->hw_params) { ret = codec_dai->driver->ops->hw_params(substream, params, codec_dai); if (ret < 0) { dev_err(codec_dai->dev, "ASoC: can't set %s hw params:" @@ -532,7 +543,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - if (cpu_dai->driver->ops->hw_params) { + if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_params) { ret = cpu_dai->driver->ops->hw_params(substream, params, cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: %s hw params failed: %d\n", @@ -559,11 +570,11 @@ out: return ret; platform_err: - if (cpu_dai->driver->ops->hw_free) + if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); interface_err: - if (codec_dai->driver->ops->hw_free) + if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); codec_err: @@ -600,10 +611,10 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) platform->driver->ops->hw_free(substream); /* now free hw params for the DAIs */ - if (codec_dai->driver->ops->hw_free) + if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); - if (cpu_dai->driver->ops->hw_free) + if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); mutex_unlock(&rtd->pcm_mutex); @@ -618,7 +629,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (codec_dai->driver->ops->trigger) { + if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) { ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); if (ret < 0) return ret; @@ -630,7 +641,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - if (cpu_dai->driver->ops->trigger) { + if (cpu_dai->driver->ops && cpu_dai->driver->ops->trigger) { ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; @@ -647,19 +658,20 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (codec_dai->driver->ops->bespoke_trigger) { + if (codec_dai->driver->ops && + codec_dai->driver->ops->bespoke_trigger) { ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai); if (ret < 0) return ret; } - if (platform->driver->bespoke_trigger) { + if (platform->driver->ops && platform->driver->bespoke_trigger) { ret = platform->driver->bespoke_trigger(substream, cmd); if (ret < 0) return ret; } - if (cpu_dai->driver->ops->bespoke_trigger) { + if (cpu_dai->driver->ops && cpu_dai->driver->ops->bespoke_trigger) { ret = cpu_dai->driver->ops->bespoke_trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; @@ -684,10 +696,10 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) if (platform->driver->ops && platform->driver->ops->pointer) offset = platform->driver->ops->pointer(substream); - if (cpu_dai->driver->ops->delay) + if (cpu_dai->driver->ops && cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); - if (codec_dai->driver->ops->delay) + if (codec_dai->driver->ops && codec_dai->driver->ops->delay) delay += codec_dai->driver->ops->delay(substream, codec_dai); if (platform->driver->delay) @@ -721,7 +733,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); - dev_dbg(fe->dev, " connected new DPCM %s path %s %s %s\n", + dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, stream ? "<-" : "->", be->dai_link->name); @@ -749,7 +761,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, if (dpcm->fe == fe) continue; - dev_dbg(fe->dev, " reparent %s path %s %s %s\n", + dev_dbg(fe->dev, "reparent %s path %s %s %s\n", stream ? "capture" : "playback", dpcm->fe->dai_link->name, stream ? "<-" : "->", dpcm->be->dai_link->name); @@ -773,7 +785,7 @@ static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) if (dpcm->state != SND_SOC_DPCM_LINK_STATE_FREE) continue; - dev_dbg(fe->dev, " freed DSP %s path %s %s %s\n", + dev_dbg(fe->dev, "freed DSP %s path %s %s %s\n", stream ? "capture" : "playback", fe->dai_link->name, stream ? "<-" : "->", dpcm->be->dai_link->name); @@ -1037,6 +1049,12 @@ static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); + if (!be_substream) { + dev_err(be->dev, "ASoC: no backend %s stream\n", + stream ? "capture" : "playback"); + continue; + } + /* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue; @@ -1054,7 +1072,8 @@ static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) (be->dpcm[stream].state != SND_SOC_DPCM_STATE_CLOSE)) continue; - dev_dbg(be->dev, "ASoC: open BE %s\n", be->dai_link->name); + dev_dbg(be->dev, "ASoC: open %s BE %s\n", + stream ? "capture" : "playback", be->dai_link->name); be_substream->runtime = be->dpcm[stream].runtime; err = soc_pcm_open(be_substream); @@ -1673,7 +1692,7 @@ static int soc_pcm_ioctl(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_platform *platform = rtd->platform; - if (platform->driver->ops->ioctl) + if (platform->driver->ops && platform->driver->ops->ioctl) return platform->driver->ops->ioctl(substream, cmd, arg); return snd_pcm_lib_ioctl(substream, cmd, arg); } @@ -1934,8 +1953,8 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute) dev_dbg(be->dev, "ASoC: BE digital mute %s\n", be->dai_link->name); - if (drv->ops->digital_mute && dai->playback_active) - drv->ops->digital_mute(dai, mute); + if (drv->ops && drv->ops->digital_mute && dai->playback_active) + drv->ops->digital_mute(dai, mute); } return 0; @@ -2116,7 +2135,7 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) pcm->private_free = platform->driver->pcm_free; out: - dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name, + dev_info(rtd->card->dev, "%s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; } @@ -2224,7 +2243,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); int snd_soc_platform_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_platform *platform) { - if (platform->driver->ops->trigger) + if (platform->driver->ops && platform->driver->ops->trigger) return platform->driver->ops->trigger(substream, cmd); return 0; } diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 29b211e9c06..5e633659c1b 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -75,7 +75,11 @@ static const struct snd_pcm_hardware dummy_dma_hardware = { static int dummy_dma_open(struct snd_pcm_substream *substream) { - snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* BE's dont need dummy params */ + if (!rtd->dai_link->no_pcm) + snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware); return 0; } diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index 63acfeb4b69..21a8c954af1 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -257,20 +257,12 @@ static int spdif_in_probe(struct platform_device *pdev) return ret; } - return snd_soc_register_component(&pdev->dev, &spdif_in_component, - &spdif_in_dai, 1); -} - -static int spdif_in_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - - return 0; + return devm_snd_soc_register_component(&pdev->dev, &spdif_in_component, + &spdif_in_dai, 1); } static struct platform_driver spdif_in_driver = { .probe = spdif_in_probe, - .remove = spdif_in_remove, .driver = { .name = "spdif-in", .owner = THIS_MODULE, diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 2fdf68c98d2..b6ef6f78dc7 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -280,7 +280,6 @@ static int spdif_out_probe(struct platform_device *pdev) struct spdif_out_dev *host; struct spear_spdif_platform_data *pdata; struct resource *res; - int ret; host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); if (!host) { @@ -307,16 +306,8 @@ static int spdif_out_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, host); - ret = snd_soc_register_component(&pdev->dev, &spdif_out_component, - &spdif_out_dai, 1); - return ret; -} - -static int spdif_out_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - - return 0; + return devm_snd_soc_register_component(&pdev->dev, &spdif_out_component, + &spdif_out_dai, 1); } #ifdef CONFIG_PM @@ -357,7 +348,6 @@ static SIMPLE_DEV_PM_OPS(spdif_out_dev_pm_ops, spdif_out_suspend, \ static struct platform_driver spdif_out_driver = { .probe = spdif_out_probe, - .remove = spdif_out_remove, .driver = { .name = "spdif-out", .owner = THIS_MODULE, diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d554d46d08b..bdd19db4a08 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -100,6 +100,7 @@ int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, { int channel; u32 reg, val; + struct tegra30_ahub_cif_conf cif_conf; channel = find_first_zero_bit(ahub->rx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT); @@ -123,15 +124,21 @@ int tegra30_ahub_allocate_rx_fifo(enum tegra30_ahub_rxcif *rxcif, TEGRA30_AHUB_CHANNEL_CTRL_RX_PACK_16; tegra30_apbif_write(reg, val); + cif_conf.threshold = 0; + cif_conf.audio_channels = 2; + cif_conf.client_channels = 2; + cif_conf.audio_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.client_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.expand = 0; + cif_conf.stereo_conv = 0; + cif_conf.replicate = 0; + cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_RX; + cif_conf.truncate = 0; + cif_conf.mono_conv = 0; + reg = TEGRA30_AHUB_CIF_RX_CTRL + (channel * TEGRA30_AHUB_CIF_RX_CTRL_STRIDE); - val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | - TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; - tegra30_apbif_write(reg, val); + ahub->soc_data->set_audio_cif(ahub->regmap_apbif, reg, &cif_conf); return 0; } @@ -183,6 +190,7 @@ int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, { int channel; u32 reg, val; + struct tegra30_ahub_cif_conf cif_conf; channel = find_first_zero_bit(ahub->tx_usage, TEGRA30_AHUB_CHANNEL_CTRL_COUNT); @@ -206,15 +214,21 @@ int tegra30_ahub_allocate_tx_fifo(enum tegra30_ahub_txcif *txcif, TEGRA30_AHUB_CHANNEL_CTRL_TX_PACK_16; tegra30_apbif_write(reg, val); + cif_conf.threshold = 0; + cif_conf.audio_channels = 2; + cif_conf.client_channels = 2; + cif_conf.audio_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.client_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.expand = 0; + cif_conf.stereo_conv = 0; + cif_conf.replicate = 0; + cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_TX; + cif_conf.truncate = 0; + cif_conf.mono_conv = 0; + reg = TEGRA30_AHUB_CIF_TX_CTRL + (channel * TEGRA30_AHUB_CIF_TX_CTRL_STRIDE); - val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | - TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; - tegra30_apbif_write(reg, val); + ahub->soc_data->set_audio_cif(ahub->regmap_apbif, reg, &cif_conf); return 0; } @@ -437,13 +451,21 @@ static const struct regmap_config tegra30_ahub_ahub_regmap_config = { static struct tegra30_ahub_soc_data soc_data_tegra30 = { .clk_list_mask = CLK_LIST_MASK_TEGRA30, + .set_audio_cif = tegra30_ahub_set_cif, }; static struct tegra30_ahub_soc_data soc_data_tegra114 = { .clk_list_mask = CLK_LIST_MASK_TEGRA114, + .set_audio_cif = tegra30_ahub_set_cif, +}; + +static struct tegra30_ahub_soc_data soc_data_tegra124 = { + .clk_list_mask = CLK_LIST_MASK_TEGRA114, + .set_audio_cif = tegra124_ahub_set_cif, }; static const struct of_device_id tegra30_ahub_of_match[] = { + { .compatible = "nvidia,tegra124-ahub", .data = &soc_data_tegra124 }, { .compatible = "nvidia,tegra114-ahub", .data = &soc_data_tegra114 }, { .compatible = "nvidia,tegra30-ahub", .data = &soc_data_tegra30 }, {}, @@ -497,6 +519,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, ahub); + ahub->soc_data = soc_data; ahub->dev = &pdev->dev; ahub->clk_d_audio = clk_get(&pdev->dev, "d_audio"); @@ -669,6 +692,70 @@ static struct platform_driver tegra30_ahub_driver = { }; module_platform_driver(tegra30_ahub_driver); +void tegra30_ahub_set_cif(struct regmap *regmap, unsigned int reg, + struct tegra30_ahub_cif_conf *conf) +{ + unsigned int value; + + value = (conf->threshold << + TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + ((conf->audio_channels - 1) << + TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + ((conf->client_channels - 1) << + TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + (conf->audio_bits << + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) | + (conf->client_bits << + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) | + (conf->expand << + TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) | + (conf->stereo_conv << + TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) | + (conf->replicate << + TEGRA30_AUDIOCIF_CTRL_REPLICATE_SHIFT) | + (conf->direction << + TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) | + (conf->truncate << + TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) | + (conf->mono_conv << + TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT); + + regmap_write(regmap, reg, value); +} +EXPORT_SYMBOL_GPL(tegra30_ahub_set_cif); + +void tegra124_ahub_set_cif(struct regmap *regmap, unsigned int reg, + struct tegra30_ahub_cif_conf *conf) +{ + unsigned int value; + + value = (conf->threshold << + TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | + ((conf->audio_channels - 1) << + TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | + ((conf->client_channels - 1) << + TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | + (conf->audio_bits << + TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_SHIFT) | + (conf->client_bits << + TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_SHIFT) | + (conf->expand << + TEGRA30_AUDIOCIF_CTRL_EXPAND_SHIFT) | + (conf->stereo_conv << + TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) | + (conf->replicate << + TEGRA30_AUDIOCIF_CTRL_REPLICATE_SHIFT) | + (conf->direction << + TEGRA30_AUDIOCIF_CTRL_DIRECTION_SHIFT) | + (conf->truncate << + TEGRA30_AUDIOCIF_CTRL_TRUNCATE_SHIFT) | + (conf->mono_conv << + TEGRA30_AUDIOCIF_CTRL_MONO_CONV_SHIFT); + + regmap_write(regmap, reg, value); +} +EXPORT_SYMBOL_GPL(tegra124_ahub_set_cif); + MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>"); MODULE_DESCRIPTION("Tegra30 AHUB driver"); MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/tegra/tegra30_ahub.h b/sound/soc/tegra/tegra30_ahub.h index 09766cdc45c..d67321d90fa 100644 --- a/sound/soc/tegra/tegra30_ahub.h +++ b/sound/soc/tegra/tegra30_ahub.h @@ -25,16 +25,30 @@ #define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US 0xf #define TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK (TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) +#define TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT 24 +#define TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US 0x3f +#define TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK (TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_MASK_US << TEGRA124_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) + /* Channel count minus 1 */ #define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT 24 #define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US 7 #define TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) /* Channel count minus 1 */ +#define TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT 20 +#define TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US 0xf +#define TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK (TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_MASK_US << TEGRA124_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) + +/* Channel count minus 1 */ #define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT 16 #define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US 7 #define TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) +/* Channel count minus 1 */ +#define TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT 16 +#define TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US 0xf +#define TEGRA124_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK (TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_MASK_US << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) + #define TEGRA30_AUDIOCIF_BITS_4 0 #define TEGRA30_AUDIOCIF_BITS_8 1 #define TEGRA30_AUDIOCIF_BITS_12 2 @@ -86,7 +100,7 @@ #define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_CH1 (TEGRA30_AUDIOCIF_STEREO_CONV_CH1 << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) #define TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_AVG (TEGRA30_AUDIOCIF_STEREO_CONV_AVG << TEGRA30_AUDIOCIF_CTRL_STEREO_CONV_SHIFT) -#define TEGRA30_AUDIOCIF_CTRL_REPLICATE 3 +#define TEGRA30_AUDIOCIF_CTRL_REPLICATE_SHIFT 3 #define TEGRA30_AUDIOCIF_DIRECTION_TX 0 #define TEGRA30_AUDIOCIF_DIRECTION_RX 1 @@ -468,8 +482,30 @@ extern int tegra30_ahub_set_rx_cif_source(enum tegra30_ahub_rxcif rxcif, enum tegra30_ahub_txcif txcif); extern int tegra30_ahub_unset_rx_cif_source(enum tegra30_ahub_rxcif rxcif); +struct tegra30_ahub_cif_conf { + unsigned int threshold; + unsigned int audio_channels; + unsigned int client_channels; + unsigned int audio_bits; + unsigned int client_bits; + unsigned int expand; + unsigned int stereo_conv; + unsigned int replicate; + unsigned int direction; + unsigned int truncate; + unsigned int mono_conv; +}; + +void tegra30_ahub_set_cif(struct regmap *regmap, unsigned int reg, + struct tegra30_ahub_cif_conf *conf); +void tegra124_ahub_set_cif(struct regmap *regmap, unsigned int reg, + struct tegra30_ahub_cif_conf *conf); + struct tegra30_ahub_soc_data { u32 clk_list_mask; + void (*set_audio_cif)(struct regmap *regmap, + unsigned int reg, + struct tegra30_ahub_cif_conf *conf); /* * FIXME: There are many more differences in HW, such as: * - More APBIF channels. diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 47565fd0450..5f20b695eba 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -30,6 +30,7 @@ #include <linux/io.h> #include <linux/module.h> #include <linux/of.h> +#include <linux/of_device.h> #include <linux/platform_device.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> @@ -179,6 +180,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); unsigned int mask, val, reg; int ret, sample_size, srate, i2sclock, bitcnt; + struct tegra30_ahub_cif_conf cif_conf; if (params_channels(params) != 2) return -EINVAL; @@ -217,21 +219,26 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, regmap_write(i2s->regmap, TEGRA30_I2S_TIMING, val); - val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | - (1 << TEGRA30_AUDIOCIF_CTRL_CLIENT_CHANNELS_SHIFT) | - TEGRA30_AUDIOCIF_CTRL_AUDIO_BITS_16 | - TEGRA30_AUDIOCIF_CTRL_CLIENT_BITS_16; + cif_conf.threshold = 0; + cif_conf.audio_channels = 2; + cif_conf.client_channels = 2; + cif_conf.audio_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.client_bits = TEGRA30_AUDIOCIF_BITS_16; + cif_conf.expand = 0; + cif_conf.stereo_conv = 0; + cif_conf.replicate = 0; + cif_conf.truncate = 0; + cif_conf.mono_conv = 0; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; + cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_RX; reg = TEGRA30_I2S_CIF_RX_CTRL; } else { - val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; + cif_conf.direction = TEGRA30_AUDIOCIF_DIRECTION_TX; reg = TEGRA30_I2S_CIF_TX_CTRL; } - regmap_write(i2s->regmap, reg, val); + i2s->soc_data->set_audio_cif(i2s->regmap, reg, &cif_conf); val = (1 << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) | (1 << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT); @@ -396,9 +403,24 @@ static const struct regmap_config tegra30_i2s_regmap_config = { .cache_type = REGCACHE_RBTREE, }; +static const struct tegra30_i2s_soc_data tegra30_i2s_config = { + .set_audio_cif = tegra30_ahub_set_cif, +}; + +static const struct tegra30_i2s_soc_data tegra124_i2s_config = { + .set_audio_cif = tegra124_ahub_set_cif, +}; + +static const struct of_device_id tegra30_i2s_of_match[] = { + { .compatible = "nvidia,tegra124-i2s", .data = &tegra124_i2s_config }, + { .compatible = "nvidia,tegra30-i2s", .data = &tegra30_i2s_config }, + {}, +}; + static int tegra30_i2s_platform_probe(struct platform_device *pdev) { struct tegra30_i2s *i2s; + const struct of_device_id *match; u32 cif_ids[2]; struct resource *mem, *memregion; void __iomem *regs; @@ -412,6 +434,14 @@ static int tegra30_i2s_platform_probe(struct platform_device *pdev) } dev_set_drvdata(&pdev->dev, i2s); + match = of_match_device(tegra30_i2s_of_match, &pdev->dev); + if (!match) { + dev_err(&pdev->dev, "Error: No device match found\n"); + ret = -ENODEV; + goto err; + } + i2s->soc_data = (struct tegra30_i2s_soc_data *)match->data; + i2s->dai = tegra30_i2s_dai_template; i2s->dai.name = dev_name(&pdev->dev); @@ -539,11 +569,6 @@ static int tegra30_i2s_resume(struct device *dev) } #endif -static const struct of_device_id tegra30_i2s_of_match[] = { - { .compatible = "nvidia,tegra30-i2s", }, - {}, -}; - static const struct dev_pm_ops tegra30_i2s_pm_ops = { SET_RUNTIME_PM_OPS(tegra30_i2s_runtime_suspend, tegra30_i2s_runtime_resume, NULL) diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index bea23afe3b9..4d0b0a30dbf 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -225,7 +225,14 @@ #define TEGRA30_I2S_LCOEF_COEF_MASK_US 0xffff #define TEGRA30_I2S_LCOEF_COEF_MASK (TEGRA30_I2S_LCOEF_COEF_MASK_US << TEGRA30_I2S_LCOEF_COEF_SHIFT) +struct tegra30_i2s_soc_data { + void (*set_audio_cif)(struct regmap *regmap, + unsigned int reg, + struct tegra30_ahub_cif_conf *conf); +}; + struct tegra30_i2s { + const struct tegra30_i2s_soc_data *soc_data; struct snd_soc_dai_driver dai; int cif_id; struct clk *clk_i2s; diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c index d173880f290..1be311c51a1 100644 --- a/sound/soc/tegra/tegra_asoc_utils.c +++ b/sound/soc/tegra/tegra_asoc_utils.c @@ -182,6 +182,8 @@ int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data, data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA30; else if (of_machine_is_compatible("nvidia,tegra114")) data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA114; + else if (of_machine_is_compatible("nvidia,tegra124")) + data->soc = TEGRA_ASOC_UTILS_SOC_TEGRA124; else { dev_err(data->dev, "SoC unknown to Tegra ASoC utils\n"); return -EINVAL; diff --git a/sound/soc/tegra/tegra_asoc_utils.h b/sound/soc/tegra/tegra_asoc_utils.h index 19fdcafed32..9577121ce97 100644 --- a/sound/soc/tegra/tegra_asoc_utils.h +++ b/sound/soc/tegra/tegra_asoc_utils.h @@ -30,6 +30,7 @@ enum tegra_asoc_utils_soc { TEGRA_ASOC_UTILS_SOC_TEGRA20, TEGRA_ASOC_UTILS_SOC_TEGRA30, TEGRA_ASOC_UTILS_SOC_TEGRA114, + TEGRA_ASOC_UTILS_SOC_TEGRA124, }; struct tegra_asoc_utils_data { diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index f056f632557..7b2d23ba69b 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -56,7 +56,6 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = { static const struct snd_dmaengine_pcm_config tegra_dmaengine_pcm_config = { .pcm_hardware = &tegra_pcm_hardware, .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .compat_filter_fn = NULL, .prealloc_buffer_size = PAGE_SIZE * 8, }; |