August 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: multi-component: TWL4030: Restore registers on removal
by Peter Ujfalusi 12 Aug '10

12 Aug '10

Add back the register restore call, when the codec driver is removed. This does not affect normal operation, but it is usefull when debugging audio through the twl4030 class codecs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- Hello Liam, Mark I did missed the removal of the register restore from the module unload path in multi-comp. This patchs adds it back. Peter sound/soc/codecs/twl4030.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 898430f..c7ee1a4 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2258,6 +2258,8 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec) static int twl4030_soc_remove(struct snd_soc_codec *codec) { + /* Reset registers to their chip default before leaving */ + twl4030_reset_registers(codec); twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } -- 1.7.2

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[alsa-devel] route plugin issue
by Tim Blechmann 12 Aug '10

12 Aug '10

hi all, when trying to use the route plugin with an lx6464es soundcard, it seems that only gain values of 1 work correctly. if i apply a different gain, the specific channel is silent. when i was trying this on another computer (with an hdsp and internal hda chip), it was working fine, though ... the only difference that i found is, that i am aware of is, that the lx6464es was used with 24bit samples, while the hdsp/hda were using 32bit samples ... is it possible that the route plugin has an issue with 24bit samples? or may there be another reason, explaining this behavior? thanks, tim -- tim(a)klingt.org http://tim.klingt.org Just what the hell is the experimental tradition? Morton Feldman

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[alsa-devel] Beginner to Alsa Development
by Tejas 12 Aug '10

12 Aug '10

Hello All, I am new to alsa interface development. I would like to write one application which accept *.pcm file as input and give playback of it on sound device on linux system. My requirement is to set sample rate 44100, channel 2 and access method is SND_PCM_ACCESS_RW_INTERLEAVED and data length is 16 bit little endian. Can any oen guide me how can I start writing application in c file and which are those APIs which I need to take care and handling of different error codes. It is very urgent. Please correct me if I am wrong at anywhere. -Tejas.

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[alsa-devel] Alsamixer-Qt 0.4.1 available
by Sebastian H. 12 Aug '10

12 Aug '10

Hello again This is a bugfix release. Not all issues have been addressed, but hopefully the most annoying ones. 1.) https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3678 The "EQ Peaks" mixer element gets skipped since there's not way to display it correctly at the moment. Also the mixer now looks for all element channels from 0 up to SND_MIXER_SCHN_LAST (= 31). Formerly it was only for the first 9 channels. 2.) https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4204 Raymond brought this up but I'm not entirely sure what's the problem here. Are there actually 5 controls named "Master" on this card? If so alsamixer-qt4 now appends the element index to the element name for elements whose name appears more that once. This should happen for all elements but I could only test it for "Capture" and "Input Source". Both appear twice on my HDA-Intel onboard card. Download at: http://xwmw.org/alsamixer-qt4/ http://sourceforge.net/projects/alsamixer-qt4/files/ Comments are welcome! Best, Sebastian H.

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[alsa-devel] [PATCH 1/2] ASoC: Automatically determine control_data for soc-cache users
by Mark Brown 12 Aug '10

12 Aug '10

Since the provision of a struct device for the CODEC is now mandatory we can use container_of() to locate the struct i2c_client and struct spi_device for relevant devices, removing the need to manually set it in each driver. A further patch will automate selection of the control type based on the bus_type of the struct device, further reducing the amount of driver code required. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 83cd8ed..b856177 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -584,11 +584,19 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, #endif if (io_types[i].i2c_read) codec->hw_read = io_types[i].i2c_read; + + codec->control_data = container_of(codec->dev, + struct i2c_client, + dev); break; case SND_SOC_SPI: if (io_types[i].spi_write) codec->hw_write = io_types[i].spi_write; + + codec->control_data = container_of(codec->dev, + struct spi_device, + dev); break; } -- 1.7.1

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[alsa-devel] [PATCH] ASoC: Fix I2C device names
by Mark Brown 12 Aug '10

12 Aug '10

Some of the I2C device names got mangled into the form 0-0x1a rather than 0-001a as they should be. Also, the WM8580 is at I2C address 0x1b not 0x1a on SMDK64xx. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/davinci/davinci-evm.c | 8 ++++---- sound/soc/ep93xx/snappercl15.c | 2 +- sound/soc/imx/eukrea-tlv320.c | 2 +- sound/soc/kirkwood/kirkwood-openrd.c | 2 +- sound/soc/s3c24xx/smdk64xx_wm8580.c | 4 ++-- sound/soc/s6000/s6105-ipcam.c | 2 +- sound/soc/sh/fsi-ak4642.c | 2 +- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/migor.c | 2 +- 10 files changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index cf9f6c9..66a6f18 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -184,7 +184,7 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { .codec_dai_name = "wm8731-hifi", .init = at91sam9g20ek_wm8731_init, .platform_name = "atmel_pcm-audio", - .codec_name = "wm8731-codec.0-0x1a", + .codec_name = "wm8731-codec.0-001a", .ops = &at91sam9g20ek_ops, }; diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 3d6b5b6..097bbbe 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -162,7 +162,7 @@ static struct snd_soc_dai_link evm_dai = { .stream_name = "AIC3X", .cpu_dai_name = "davinci-mcasp-dai.0", .codec_dai_name = "tlv320aic3x-hifi", - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init, .ops = &evm_ops, @@ -175,7 +175,7 @@ static struct snd_soc_dai_link dm365_evm_dai = { .cpu_dai_name = "davinci-i2s-dai", .codec_dai_name = "tlv320aic3x-hifi", .init = evm_aic3x_init, - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .ops = &evm_ops, #elif defined(CONFIG_SND_DM365_VOICE_CODEC) .name = "Voice Codec - CQ93VC", @@ -194,7 +194,7 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .cpu_dai_name= "davinci-mcasp-dai.0", .codec_dai_name = "tlv320aic3x-hifi", .platform_name ="davinci-pcm-audio", - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .init = evm_aic3x_init, .ops = &evm_ops, }, @@ -213,7 +213,7 @@ static struct snd_soc_dai_link da8xx_evm_dai = { .stream_name = "AIC3X", .cpu_dai_name= "davinci-mcasp-dai.0", .codec_dai_name = "tlv320aic3x-hifi", - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init, .ops = &evm_ops, diff --git a/sound/soc/ep93xx/snappercl15.c b/sound/soc/ep93xx/snappercl15.c index 539e839..402d299 100644 --- a/sound/soc/ep93xx/snappercl15.c +++ b/sound/soc/ep93xx/snappercl15.c @@ -92,7 +92,7 @@ static struct snd_soc_dai_link snappercl15_dai = { .stream_name = "AIC23", .cpu_dai_name = "ep93xx-i2s-dai", .codec_dai_name = "tlv320aic23-hifi", - .codec_name = "tlv320aic23-codec.0-0x1a", + .codec_name = "tlv320aic23-codec.0-001a", .platform_name = "ep93xx-pcm-audio", .init = snappercl15_tlv320aic23_init, .ops = &snappercl15_ops, diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c index ab76cae..807f736 100644 --- a/sound/soc/imx/eukrea-tlv320.c +++ b/sound/soc/imx/eukrea-tlv320.c @@ -81,7 +81,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .stream_name = "TLV320AIC23", .codec_dai = "tlv320aic23-hifi", .platform_name = "imx-pcm-audio.0", - .codec_name = "tlv320aic23-codec.0-0x1a", + .codec_name = "tlv320aic23-codec.0-001a", .cpu_dai = "imx-ssi-dai.0", .ops = &eukrea_tlv320_snd_ops, }; diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c index 6f9ac42..b375021 100644 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ b/sound/soc/kirkwood/kirkwood-openrd.c @@ -67,7 +67,7 @@ static struct snd_soc_dai_link openrd_client_dai[] = { .cpu_dai_name = "kirkwood-i2s-dai", .platform_name = "kirkwood-pcm-audio", .codec_dai_name = "cs42l51_hifi", - .codec_name = "cs42l51-codec.0-0x4a", + .codec_name = "cs42l51-codec.0-004a", .ops = &openrd_client_ops, }, }; diff --git a/sound/soc/s3c24xx/smdk64xx_wm8580.c b/sound/soc/s3c24xx/smdk64xx_wm8580.c index 4ccfe84..9d959a0 100644 --- a/sound/soc/s3c24xx/smdk64xx_wm8580.c +++ b/sound/soc/s3c24xx/smdk64xx_wm8580.c @@ -226,7 +226,7 @@ static struct snd_soc_dai_link smdk64xx_dai[] = { .cpu_dai_name = "s3c64xx-i2s-v4-dai", .codec_dai_name = "wm8580-hifi-playback", .platform_name = "s3c24xx-pcm-audio", - .codec_name = "wm8580-codec.0-0x1a", + .codec_name = "wm8580-codec.0-001b", .init = smdk64xx_wm8580_init_paifrx, .ops = &smdk64xx_ops, }, @@ -236,7 +236,7 @@ static struct snd_soc_dai_link smdk64xx_dai[] = { .cpu_dai_name = "s3c64xx-i2s-v4-dai", .codec_dai_name = "wm8580-hifi-capture", .platform_name = "s3c24xx-pcm-audio", - .codec_name = "wm8580-codec.0-0x1a", + .codec_name = "wm8580-codec.0-001b", .init = smdk64xx_wm8580_init_paiftx, .ops = &smdk64xx_ops, }, diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index dd39467..dff9d39 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -179,7 +179,7 @@ static struct snd_soc_dai_link s6105_dai = { .cpu_dai_name = "s6000-i2s-dai", .codec_dai_name = "tlv320aic3x-hifi", .platform_name = "s6000-pcm-audio", - .codec_name = "tlv320aic3x-codec.0-0x1a", + .codec_name = "tlv320aic3x-codec.0-001a", .init = s6105_aic3x_init, .ops = &s6105_ops, }; diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index ff17ff3..9e107a9 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -32,7 +32,7 @@ static struct snd_soc_dai_link fsi_dai_link = { .cpu_dai_name = "fsia-dai", /* fsi A */ .codec_dai_name = "ak4642-hifi", .platform_name = "fsi-pcm-audio", - .codec_name = "ak4642-codec.0-0x12", + .codec_name = "ak4642-codec.0-0012", .init = fsi_ak4642_dai_init, .ops = NULL, }; diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c index e2f6fe5..4f9298f 100644 --- a/sound/soc/sh/fsi-da7210.c +++ b/sound/soc/sh/fsi-da7210.c @@ -28,7 +28,7 @@ static struct snd_soc_dai_link fsi_da7210_dai = { .cpu_dai_name = "fsib-dai", /* FSI B */ .codec_dai_name = "da7210-hifi", .platform_name = "fsi-pcm-audio", - .codec_name = "da7210-codec.0-0x1a", + .codec_name = "da7210-codec.0-001a", .init = fsi_da7210_init, }; diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 0dc3a0e..866d78f 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -155,7 +155,7 @@ static struct snd_soc_dai_link migor_dai = { .cpu_dai_name = "siu-i2s-dai", .codec_dai_name = "wm8978-hifi", .platform_name = "siu-pcm-audio", - .codec_name = "wm8978-codec.0-0x1a", + .codec_name = "wm8978-codec.0-001a", .ops = &migor_dai_ops, .init = migor_dai_init, }; -- 1.7.1

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[alsa-devel] ALSA: How to create multiple cards?
by Truong Hoang 12 Aug '10

12 Aug '10

Hi all, I have a problem that I need help from the ALSA experts. I want to write an alsa shim .ko which creates multiple cards. Let's assume 2 cards. My program calls snd_card_create() and snd_card_register() two times. I also has 1 PCM for each of these card. After install the .ko to the kernal, I see the /proc/asound has card0 and card1. But the "aplay -l" sees only card0. The snd_card_next() only returns 0 (i.e. 1 card). What did I miss in my program? The /dev/snd/ has only hwC0D0 and pcmD0C0p. I am a beginner in ALSA, so I appreciate if you can give the explanation in detail. Thank you,

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[alsa-devel] [PATCH 1/4] sound: oss: sh_dac_audio.c removed duplicated #include
by Andrea Gelmini 12 Aug '10

12 Aug '10

Signed-off-by: Andrea Gelmini <andrea.gelmini(a)gelma.net> --- sound/oss/sh_dac_audio.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c index fdb58eb..479e302 100644 --- a/sound/oss/sh_dac_audio.c +++ b/sound/oss/sh_dac_audio.c @@ -15,7 +15,6 @@ #include <linux/linkage.h> #include <linux/slab.h> #include <linux/fs.h> -#include <linux/smp_lock.h> #include <linux/sound.h> #include <linux/smp_lock.h> #include <linux/soundcard.h> -- 1.7.2.1.85.g2d089

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[PATCH 5/5] ASoc: add machine driver for 88pm860x
by Haojian Zhuang 12 Aug '10

12 Aug '10

88PM860x codec is used in Marvell development board (Saarb and tavorevb3). Codec is used as master mode in both of these two boards. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/pxa/Kconfig | 18 ++++ sound/soc/pxa/Makefile | 4 + sound/soc/pxa/saarb.c | 208 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/pxa/tavorevb3.c | 209 +++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 439 insertions(+), 0 deletions(-) create mode 100644 sound/soc/pxa/saarb.c create mode 100644 sound/soc/pxa/tavorevb3.c diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index e30c832..37f191b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,24 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. +config SND_SOC_SAARB + tristate "SoC Audio support for Marvell Saarb" + depends on SND_PXA2XX_SOC && MACH_SAARB + select SND_PXA_SOC_SSP + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on the + Marvell Saarb reference platform. + +config SND_SOC_TAVOREVB3 + tristate "SoC Audio support for Marvell Tavor EVB3" + depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 + select SND_PXA_SOC_SSP + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on the + Marvell Saarb reference platform. + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index caa03d8..0766016 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -19,6 +19,8 @@ snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o +snd-soc-saarb-objs := saarb.o +snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o @@ -38,6 +40,8 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o +obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o +obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c new file mode 100644 index 0000000..4220c28 --- /dev/null +++ b/sound/soc/pxa/saarb.c @@ -0,0 +1,208 @@ +/* + * saarb.c -- SoC audio for saarb + * + * Copyright (C) 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> + +#include <asm/mach-types.h> + +#include "../codecs/88pm860x-codec.h" +#include "../../arm/pxa2xx-pcm.h" +#include "pxa-ssp.h" + +static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd); + +static struct platform_device *saarb_snd_device; +static struct snd_soc_jack hs_jack; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +/* saarb machine dapm widgets */ +static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_LINE("Lineout Out 1", NULL), + SND_SOC_DAPM_LINE("Lineout Out 2", NULL), + SND_SOC_DAPM_SPK("Board Speaker", NULL), + SND_SOC_DAPM_MIC("Board Mic 1", NULL), + SND_SOC_DAPM_MIC("Board Mic 3", NULL), +}; + +/* saarb machine audio map */ +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "HS1"}, + {"Headphone Jack", NULL, "HS2"}, + + {"Board Speaker", NULL, "LSP"}, + {"Board Speaker", NULL, "LSN"}, + + {"Lineout Out 1", NULL, "LINEOUT1"}, + {"Lineout Out 2", NULL, "LINEOUT2"}, + + {"MIC1P", NULL, "Mic1 Bias"}, + {"MIC1N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Board Mic 1"}, + + {"MIC2P", NULL, "Mic1 Bias"}, + {"MIC2N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Mic Jack"}, + + {"MIC3P", NULL, "Mic3 Bias"}, + {"MIC3N", NULL, "Mic3 Bias"}, + {"Mic3 Bias", NULL, "Board Mic 3"}, +}; + +static const struct snd_kcontrol_new pm860x_saarb_controls[] = { + SOC_DAPM_PIN_SWITCH("Board Speaker"), + SOC_DAPM_PIN_SWITCH("Board Mic 1"), + SOC_DAPM_PIN_SWITCH("Board Mic 3"), +}; + +static int saarb_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + 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; + int width = snd_pcm_format_physical_width(params_format(params)); + int ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, + PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); + + return ret; +} + +static struct snd_soc_ops saarb_i2s_ops = { + .hw_params = saarb_i2s_hw_params, +}; + +static struct snd_soc_dai_link saarb_dai[] = { + { + .name = "88PM860x I2S", + .stream_name = "I2S Audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .platform_name = "pxa-pcm-audio", + .codec_name = "88pm860x-codec", + .init = saarb_pm860x_init, + .ops = &saarb_i2s_ops, + }, +}; + +static struct snd_soc_card snd_soc_card_saarb = { + .name = "Saarb", + .dai_link = saarb_dai, + .num_links = ARRAY_SIZE(saarb_dai), +}; + +static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + ret = snd_soc_add_controls(codec, pm860x_saarb_controls, + ARRAY_SIZE(pm860x_saarb_controls)); + if (ret < 0) + return ret; + + snd_soc_dapm_new_controls(codec, saarb_dapm_widgets, + ARRAY_SIZE(saarb_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_disable_pin(codec, "Board Speaker"); + snd_soc_dapm_disable_pin(codec, "Board Mic 1"); + snd_soc_dapm_disable_pin(codec, "Board Mic 3"); + + ret = snd_soc_dapm_sync(codec); + if (ret < 0) + return ret; + + /* Jack detection */ + ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET + | SND_JACK_LINEOUT | SND_JACK_BTN_0, &hs_jack); + if (ret < 0) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret < 0) + return ret; + + ret = pm860x_hs_detect(codec, &hs_jack, PM860X_DET_MASK); + return ret; +} + +static int __init saarb_init(void) +{ + int ret; + + if (!machine_is_saarb()) + return -ENODEV; + saarb_snd_device = platform_device_alloc("soc-audio", -1); + if (!saarb_snd_device) + return -ENOMEM; + + platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb); + + ret = platform_device_add(saarb_snd_device); + if (ret) + platform_device_put(saarb_snd_device); + + return ret; +} + +static void __exit saarb_exit(void) +{ + platform_device_unregister(saarb_snd_device); +} + +module_init(saarb_init); +module_exit(saarb_exit); + +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c new file mode 100644 index 0000000..d5f8ac9 --- /dev/null +++ b/sound/soc/pxa/tavorevb3.c @@ -0,0 +1,209 @@ +/* + * tavorevb3.c -- SoC audio for Tavor EVB3 + * + * Copyright (C) 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/clk.h> +#include <linux/i2c.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> + +#include <asm/mach-types.h> + +#include "../codecs/88pm860x-codec.h" +#include "../../arm/pxa2xx-pcm.h" +#include "pxa-ssp.h" + +static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd); + +static struct platform_device *evb3_snd_device; +static struct snd_soc_jack hs_jack; + +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin hs_jack_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +/* tavorevb3 machine dapm widgets */ +static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_LINE("Lineout Out 1", NULL), + SND_SOC_DAPM_LINE("Lineout Out 2", NULL), + SND_SOC_DAPM_SPK("Board Speaker", NULL), + SND_SOC_DAPM_MIC("Board Mic 1", NULL), + SND_SOC_DAPM_MIC("Board Mic 3", NULL), +}; + +/* tavorevb3 machine audio map */ +static const struct snd_soc_dapm_route audio_map[] = { + {"Headphone Jack", NULL, "HS1"}, + {"Headphone Jack", NULL, "HS2"}, + + {"Board Speaker", NULL, "LSP"}, + {"Board Speaker", NULL, "LSN"}, + + {"Lineout Out 1", NULL, "LINEOUT1"}, + {"Lineout Out 2", NULL, "LINEOUT2"}, + + {"MIC1P", NULL, "Mic1 Bias"}, + {"MIC1N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Board Mic 1"}, + + {"MIC2P", NULL, "Mic1 Bias"}, + {"MIC2N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Mic Jack"}, + + {"MIC3P", NULL, "Mic3 Bias"}, + {"MIC3N", NULL, "Mic3 Bias"}, + {"Mic3 Bias", NULL, "Board Mic 3"}, +}; + +static const struct snd_kcontrol_new pm860x_evb3_controls[] = { + SOC_DAPM_PIN_SWITCH("Board Speaker"), + SOC_DAPM_PIN_SWITCH("Board Mic 1"), + SOC_DAPM_PIN_SWITCH("Board Mic 3"), +}; + +static int evb3_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + 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; + int width = snd_pcm_format_physical_width(params_format(params)); + int ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, + PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); + return ret; +} + +static struct snd_soc_ops evb3_i2s_ops = { + .hw_params = evb3_i2s_hw_params, +}; + +static struct snd_soc_dai_link evb3_dai[] = { + { + .name = "88PM860x I2S", + .stream_name = "I2S Audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .platform_name = "pxa-pcm-audio", + .codec_name = "88pm860x-codec", + .init = evb3_pm860x_init, + .ops = &evb3_i2s_ops, + }, +}; + +static struct snd_soc_card snd_soc_card_evb3 = { + .name = "Tavor EVB3", + .dai_link = evb3_dai, + .num_links = ARRAY_SIZE(evb3_dai), +}; + +static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + ret = snd_soc_add_controls(codec, pm860x_evb3_controls, + ARRAY_SIZE(pm860x_evb3_controls)); + if (ret < 0) + return ret; + + snd_soc_dapm_new_controls(codec, evb3_dapm_widgets, + ARRAY_SIZE(evb3_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + snd_soc_dapm_disable_pin(codec, "Board Speaker"); + snd_soc_dapm_disable_pin(codec, "Board Mic 1"); + snd_soc_dapm_disable_pin(codec, "Board Mic 3"); + + ret = snd_soc_dapm_sync(codec); + if (ret < 0) + return ret; + + /* Jack detection */ + ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET + | SND_JACK_LINEOUT | SND_JACK_BTN_0, &hs_jack); + if (ret < 0) + return ret; + + ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + if (ret < 0) + return ret; + + ret = pm860x_hs_detect(codec, &hs_jack, PM860X_DET_MASK); + return ret; +} + +static int __init tavorevb3_init(void) +{ + int ret; + + if (!machine_is_tavorevb3()) + return -ENODEV; + evb3_snd_device = platform_device_alloc("soc-audio", -1); + if (!evb3_snd_device) + return -ENOMEM; + + platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3); + + ret = platform_device_add(evb3_snd_device); + if (ret) + platform_device_put(evb3_snd_device); + + return ret; +} + +static void __exit tavorevb3_exit(void) +{ + platform_device_unregister(evb3_snd_device); +} + +module_init(tavorevb3_init); +module_exit(tavorevb3_exit); + +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb"); +MODULE_LICENSE("GPL"); + -- 1.5.6.5

1 0

[PATCH 1/3] ASoC: add 88pm860x codec driver
by Haojian Zhuang 12 Aug '10

12 Aug '10

Add 88PM860x codec driver. 88PM860x codec supports two interfaces. And it also supports headset/mic/hook/short detection. Signed-off-by: Haojian Zhuang <haojian.zhuang(a)marvell.com> --- sound/soc/codecs/88pm860x-codec.c | 1410 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/88pm860x-codec.h | 92 +++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + 4 files changed, 1508 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/88pm860x-codec.c create mode 100644 sound/soc/codecs/88pm860x-codec.h diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c new file mode 100644 index 0000000..1acb1f1 --- /dev/null +++ b/sound/soc/codecs/88pm860x-codec.c @@ -0,0 +1,1410 @@ +/* + * 88pm860x-codec.c -- 88PM860x ALSA SoC Audio Driver + * + * Copyright 2010 Marvell International Ltd. + * Author: Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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. + */ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/mfd/88pm860x.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/initval.h> + +#include "88pm860x-codec.h" + +#define MAX_NAME_LEN 20 +#define REG_CACHE_SIZE 0x40 +#define REG_CACHE_BASE 0xb0 + +/* Status Register 1 (0x01) */ +#define REG_STATUS_1 0x01 +#define MIC_STATUS (1 << 7) +#define HOOK_STATUS (1 << 6) +#define HEADSET_STATUS (1 << 5) + +/* Mic Detection Register (0x37) */ +#define REG_MIC_DET 0x37 +#define CONTINUOUS_POLLING (3 << 1) +#define EN_MIC_DET (1 << 0) +#define MICDET_MASK 0x07 + +/* Headset Detection Register (0x38) */ +#define REG_HS_DET 0x38 +#define EN_HS_DET (1 << 0) + +/* Misc2 Register (0x42) */ +#define REG_MISC2 0x42 +#define AUDIO_PLL (1 << 5) +#define AUDIO_SECTION_RESET (1 << 4) +#define AUDIO_SECTION_ON (1 << 3) + +/* PCM Interface Register 2 (0xb1) */ +#define PCM_INF2_BCLK (1 << 6) /* Bit clock polarity */ +#define PCM_INF2_FS (1 << 5) /* Frame Sync polarity */ +#define PCM_INF2_MASTER (1 << 4) /* Master / Slave */ +#define PCM_INF2_18WL (1 << 3) /* 18 / 16 bits */ +#define PCM_GENERAL_I2S 0 +#define PCM_EXACT_I2S 1 +#define PCM_LEFT_I2S 2 +#define PCM_RIGHT_I2S 3 +#define PCM_SHORT_FS 4 +#define PCM_LONG_FS 5 +#define PCM_MODE_MASK 7 + +/* I2S Interface Register 4 (0xbe) */ +#define I2S_EQU_BYP (1 << 6) + +/* DAC Offset Register (0xcb) */ +#define DAC_MUTE (1 << 7) + +/* ADC Analog Register 1 (0xd0) */ +#define REG_ADC_ANA_1 0xd0 +#define MIC1BIAS_MASK 0x60 + +/* Earpiece/Speaker Control Register 2 (0xda) */ +#define REG_EAR2 0xda +#define RSYNC_CHANGE (1 << 2) + +/* Audio Supplies Register 2 (0xdc) */ +#define REG_SUPPLIES2 0xdc +#define LDO15_READY (1 << 4) +#define LDO15_EN (1 << 3) +#define CPUMP_READY (1 << 2) +#define CPUMP_EN (1 << 1) +#define AUDIO_EN (1 << 0) +#define SUPPLY_MASK (LDO15_EN | CPUMP_EN | AUDIO_EN) + +/* Audio Enable Register 1 (0xdd) */ +#define ADC_MOD_RIGHT (1 << 1) +#define ADC_MOD_LEFT (1 << 0) + +/* Audio Enable Register 2 (0xde) */ +#define ADC_LEFT (1 << 5) +#define ADC_RIGHT (1 << 4) + +/* DAC Enable Register 2 (0xe1) */ +#define DAC_LEFT (1 << 5) +#define DAC_RIGHT (1 << 4) +#define MODULATOR (1 << 3) + +/* Shorts Register (0xeb) */ +#define REG_SHORTS 0xeb +#define SHORT_LO2 (1 << 6) +#define SHORT_LO1 (1 << 4) +#define SHORT_HS2 (1 << 2) +#define SHORT_HS1 (1 << 0) + +enum { + FILTER_BYPASS = 0, + FILTER_LOW_PASS_1, + FILTER_LOW_PASS_2, + FILTER_HIGH_PASS_3, + FILTER_MAX, +}; + +struct pm860x_priv { + unsigned int sysclk; + unsigned int pcmclk; + unsigned int dir; + unsigned int filter; + struct snd_soc_codec *codec; + struct i2c_client *i2c; + struct pm860x_chip *chip; + + int irq[4]; + unsigned char name[4][MAX_NAME_LEN]; + unsigned char reg_cache[REG_CACHE_SIZE]; +}; + +/* -9450dB to 0dB in 150dB steps ( mute instead of -9450dB) */ +static const DECLARE_TLV_DB_SCALE(dpga_tlv, -9450, 150, 1); + +/* -9dB to 0db in 3dB steps */ +static const DECLARE_TLV_DB_SCALE(adc_tlv, -900, 300, 0); + +/* {-23, -17, -13.5, -11, -9, -6, -3, 0}dB */ +static const unsigned int mic_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(-2300, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-1700, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-1350, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(-1100, 0, 0), + 4, 7, TLV_DB_SCALE_ITEM(-900, 300, 0), +}; + +/* {0, 0, 0, -6, 0, 6, 12, 18}dB */ +static const unsigned int aux_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(0, 0, 0), + 3, 7, TLV_DB_SCALE_ITEM(-600, 600, 0), +}; + +/* {-16, -13, -10, -7, -5.2, -3,3, -2.2, 0}dB, mute instead of -16dB */ +static const unsigned int out_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 3, TLV_DB_SCALE_ITEM(-1600, 300, 1), + 4, 4, TLV_DB_SCALE_ITEM(-520, 0, 0), + 5, 5, TLV_DB_SCALE_ITEM(-330, 0, 0), + 6, 7, TLV_DB_SCALE_ITEM(-220, 220, 0), +}; + +static const unsigned int st_tlv[] = { + TLV_DB_RANGE_HEAD(8), + 0, 1, TLV_DB_SCALE_ITEM(-12041, 602, 0), + 2, 3, TLV_DB_SCALE_ITEM(-11087, 250, 0), + 4, 5, TLV_DB_SCALE_ITEM(-10643, 158, 0), + 6, 7, TLV_DB_SCALE_ITEM(-10351, 116, 0), + 8, 9, TLV_DB_SCALE_ITEM(-10133, 92, 0), + 10, 13, TLV_DB_SCALE_ITEM(-9958, 70, 0), + 14, 17, TLV_DB_SCALE_ITEM(-9689, 53, 0), + 18, 271, TLV_DB_SCALE_ITEM(-9484, 37, 0), +}; + +/* Sidetone Gain = M * 2^(-5-N) */ +struct st_gain { + unsigned int db; + unsigned int m; + unsigned int n; +}; + +static struct st_gain st_table[] = { + {-12041, 1, 15}, {-11439, 1, 14}, {-11087, 3, 15}, {-10837, 1, 13}, + {-10643, 5, 15}, {-10485, 3, 14}, {-10351, 7, 15}, {-10235, 1, 12}, + {-10133, 9, 15}, {-10041, 5, 14}, { -9958, 11, 15}, { -9883, 3, 13}, + { -9813, 13, 15}, { -9749, 7, 14}, { -9689, 15, 15}, { -9633, 1, 11}, + { -9580, 17, 15}, { -9531, 9, 14}, { -9484, 19, 15}, { -9439, 5, 13}, + { -9397, 21, 15}, { -9356, 11, 14}, { -9318, 23, 15}, { -9281, 3, 12}, + { -9245, 25, 15}, { -9211, 13, 14}, { -9178, 27, 15}, { -9147, 7, 13}, + { -9116, 29, 15}, { -9087, 15, 14}, { -9058, 31, 15}, { -9031, 1, 10}, + { -8978, 17, 14}, { -8929, 9, 13}, { -8882, 19, 14}, { -8837, 5, 12}, + { -8795, 21, 14}, { -8754, 11, 13}, { -8716, 23, 14}, { -8679, 3, 11}, + { -8643, 25, 14}, { -8609, 13, 13}, { -8576, 27, 14}, { -8545, 7, 12}, + { -8514, 29, 14}, { -8485, 15, 13}, { -8456, 31, 14}, { -8429, 1, 9}, + { -8376, 17, 13}, { -8327, 9, 12}, { -8280, 19, 13}, { -8235, 5, 11}, + { -8193, 21, 13}, { -8152, 11, 12}, { -8114, 23, 13}, { -8077, 3, 10}, + { -8041, 25, 13}, { -8007, 13, 12}, { -7974, 27, 13}, { -7943, 7, 11}, + { -7912, 29, 13}, { -7883, 15, 12}, { -7854, 31, 13}, { -7827, 1, 8}, + { -7774, 17, 12}, { -7724, 9, 11}, { -7678, 19, 12}, { -7633, 5, 10}, + { -7591, 21, 12}, { -7550, 11, 11}, { -7512, 23, 12}, { -7475, 3, 9}, + { -7439, 25, 12}, { -7405, 13, 11}, { -7372, 27, 12}, { -7341, 7, 10}, + { -7310, 29, 12}, { -7281, 15, 11}, { -7252, 31, 12}, { -7225, 1, 7}, + { -7172, 17, 11}, { -7122, 9, 10}, { -7075, 19, 11}, { -7031, 5, 9}, + { -6989, 21, 11}, { -6948, 11, 10}, { -6910, 23, 11}, { -6873, 3, 8}, + { -6837, 25, 11}, { -6803, 13, 10}, { -6770, 27, 11}, { -6739, 7, 9}, + { -6708, 29, 11}, { -6679, 15, 10}, { -6650, 31, 11}, { -6623, 1, 6}, + { -6570, 17, 10}, { -6520, 9, 9}, { -6473, 19, 10}, { -6429, 5, 8}, + { -6386, 21, 10}, { -6346, 11, 9}, { -6307, 23, 10}, { -6270, 3, 7}, + { -6235, 25, 10}, { -6201, 13, 9}, { -6168, 27, 10}, { -6137, 7, 8}, + { -6106, 29, 10}, { -6077, 15, 9}, { -6048, 31, 10}, { -6021, 1, 5}, + { -5968, 17, 9}, { -5918, 9, 8}, { -5871, 19, 9}, { -5827, 5, 7}, + { -5784, 21, 9}, { -5744, 11, 8}, { -5705, 23, 9}, { -5668, 3, 6}, + { -5633, 25, 9}, { -5599, 13, 8}, { -5566, 27, 9}, { -5535, 7, 7}, + { -5504, 29, 9}, { -5475, 15, 8}, { -5446, 31, 9}, { -5419, 1, 4}, + { -5366, 17, 8}, { -5316, 9, 7}, { -5269, 19, 8}, { -5225, 5, 6}, + { -5182, 21, 8}, { -5142, 11, 7}, { -5103, 23, 8}, { -5066, 3, 5}, + { -5031, 25, 8}, { -4997, 13, 7}, { -4964, 27, 8}, { -4932, 7, 6}, + { -4902, 29, 8}, { -4873, 15, 7}, { -4844, 31, 8}, { -4816, 1, 3}, + { -4764, 17, 7}, { -4714, 9, 6}, { -4667, 19, 7}, { -4623, 5, 5}, + { -4580, 21, 7}, { -4540, 11, 6}, { -4501, 23, 7}, { -4464, 3, 4}, + { -4429, 25, 7}, { -4395, 13, 6}, { -4362, 27, 7}, { -4330, 7, 5}, + { -4300, 29, 7}, { -4270, 15, 6}, { -4242, 31, 7}, { -4214, 1, 2}, + { -4162, 17, 6}, { -4112, 9, 5}, { -4065, 19, 6}, { -4021, 5, 4}, + { -3978, 21, 6}, { -3938, 11, 5}, { -3899, 23, 6}, { -3862, 3, 3}, + { -3827, 25, 6}, { -3793, 13, 5}, { -3760, 27, 6}, { -3728, 7, 4}, + { -3698, 29, 6}, { -3668, 15, 5}, { -3640, 31, 6}, { -3612, 1, 1}, + { -3560, 17, 5}, { -3510, 9, 4}, { -3463, 19, 5}, { -3419, 5, 3}, + { -3376, 21, 5}, { -3336, 11, 4}, { -3297, 23, 5}, { -3260, 3, 2}, + { -3225, 25, 5}, { -3191, 13, 4}, { -3158, 27, 5}, { -3126, 7, 3}, + { -3096, 29, 5}, { -3066, 15, 4}, { -3038, 31, 5}, { -3010, 1, 0}, + { -2958, 17, 4}, { -2908, 9, 3}, { -2861, 19, 4}, { -2816, 5, 2}, + { -2774, 21, 4}, { -2734, 11, 3}, { -2695, 23, 4}, { -2658, 3, 1}, + { -2623, 25, 4}, { -2589, 13, 3}, { -2556, 27, 4}, { -2524, 7, 2}, + { -2494, 29, 4}, { -2464, 15, 3}, { -2436, 31, 4}, { -2408, 2, 0}, + { -2356, 17, 3}, { -2306, 9, 2}, { -2259, 19, 3}, { -2214, 5, 1}, + { -2172, 21, 3}, { -2132, 11, 2}, { -2093, 23, 3}, { -2056, 3, 0}, + { -2021, 25, 3}, { -1987, 13, 2}, { -1954, 27, 3}, { -1922, 7, 1}, + { -1892, 29, 3}, { -1862, 15, 2}, { -1834, 31, 3}, { -1806, 4, 0}, + { -1754, 17, 2}, { -1704, 9, 1}, { -1657, 19, 2}, { -1612, 5, 0}, + { -1570, 21, 2}, { -1530, 11, 1}, { -1491, 23, 2}, { -1454, 6, 0}, + { -1419, 25, 2}, { -1384, 13, 1}, { -1352, 27, 2}, { -1320, 7, 0}, + { -1290, 29, 2}, { -1260, 15, 1}, { -1232, 31, 2}, { -1204, 8, 0}, + { -1151, 17, 1}, { -1102, 9, 0}, { -1055, 19, 1}, { -1010, 10, 0}, + { -968, 21, 1}, { -928, 11, 0}, { -889, 23, 1}, { -852, 12, 0}, + { -816, 25, 1}, { -782, 13, 0}, { -750, 27, 1}, { -718, 14, 0}, + { -688, 29, 1}, { -658, 15, 0}, { -630, 31, 1}, { -602, 16, 0}, + { -549, 17, 0}, { -500, 18, 0}, { -453, 19, 0}, { -408, 20, 0}, + { -366, 21, 0}, { -325, 22, 0}, { -287, 23, 0}, { -250, 24, 0}, + { -214, 25, 0}, { -180, 26, 0}, { -148, 27, 0}, { -116, 28, 0}, + { -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) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + int val[2], val2[2], i; + + val[0] = snd_soc_read(codec, reg) & 0x3f; + val[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT) >> 4) & 0xf; + val2[0] = snd_soc_read(codec, reg2) & 0x3f; + val2[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT)) & 0xf; + + for (i = 0; i < ARRAY_SIZE(st_table); i++) { + if ((st_table[i].m == val[0]) && (st_table[i].n == val[1])) + ucontrol->value.integer.value[0] = i; + if ((st_table[i].m == val2[0]) && (st_table[i].n == val2[1])) + ucontrol->value.integer.value[1] = i; + } + return 0; +} + +static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + int err; + unsigned int val, val2; + + val = ucontrol->value.integer.value[0]; + val2 = ucontrol->value.integer.value[1]; + + err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m); + if (err < 0) + return err; + err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0xf0, + st_table[val].n << 4); + if (err < 0) + return err; + + err = snd_soc_update_bits(codec, reg2, 0x3f, st_table[val2].m); + if (err < 0) + return err; + err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0x0f, + st_table[val2].n); + return err; +} + +static int snd_soc_get_volsw_2r_out(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + int max = mc->max, val, val2; + unsigned int mask = (1 << fls(max)) - 1; + + val = snd_soc_read(codec, reg) >> shift; + val2 = snd_soc_read(codec, reg2) >> shift; + ucontrol->value.integer.value[0] = (max - val) & mask; + ucontrol->value.integer.value[1] = (max - val2) & mask; + + return 0; +} + +static int snd_soc_put_volsw_2r_out(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + int err; + unsigned int val, val2, val_mask; + + val_mask = mask << shift; + val = ((max - ucontrol->value.integer.value[0]) & mask); + val2 = ((max - ucontrol->value.integer.value[1]) & mask); + + val = val << shift; + val2 = val2 << shift; + + err = snd_soc_update_bits(codec, reg, val_mask, val); + if (err < 0) + return err; + + err = snd_soc_update_bits(codec, reg2, val_mask, val2); + return err; +} + +/* DAPM Widget Events */ +static int pm860x_rsync_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + /* unmute DAC */ + snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, 0); + snd_soc_update_bits(codec, PM860X_EAR_CTRL_2, + RSYNC_CHANGE, RSYNC_CHANGE); + return 0; +} + +static int pm860x_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int en1 = 0, en2 = 0; + + if (!strcmp(w->name, "Left ADC")) { + en1 = ADC_MOD_LEFT; + en2 = ADC_LEFT; + } + if (!strcmp(w->name, "Right ADC")) { + en1 = ADC_MOD_RIGHT; + en2 = ADC_RIGHT; + } + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, PM860X_ADC_EN_1, en1, en1); + snd_soc_update_bits(codec, PM860X_ADC_EN_2, en2, en2); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, PM860X_ADC_EN_1, en1, 0); + snd_soc_update_bits(codec, PM860X_ADC_EN_2, en2, 0); + break; + } + return 0; +} + +static int pm860x_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int dac = 0; + int data; + + if (!strcmp(w->name, "Left DAC")) + dac = DAC_LEFT; + if (!strcmp(w->name, "Right DAC")) + dac = DAC_RIGHT; + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (dac) { + dac |= MODULATOR; + /* mute */ + snd_soc_update_bits(codec, PM860X_DAC_OFFSET, + DAC_MUTE, DAC_MUTE); + snd_soc_update_bits(codec, PM860X_EAR_CTRL_2, + RSYNC_CHANGE, RSYNC_CHANGE); + /* update dac */ + snd_soc_update_bits(codec, PM860X_DAC_EN_2, + dac, dac); + } + break; + case SND_SOC_DAPM_PRE_PMD: + if (dac) { + /* mute */ + snd_soc_update_bits(codec, PM860X_DAC_OFFSET, + DAC_MUTE, DAC_MUTE); + snd_soc_update_bits(codec, PM860X_EAR_CTRL_2, + RSYNC_CHANGE, RSYNC_CHANGE); + /* update dac */ + data = snd_soc_read(codec, PM860X_DAC_EN_2); + data &= ~dac; + if (!(data & (DAC_LEFT | DAC_RIGHT))) + data &= ~MODULATOR; + snd_soc_write(codec, PM860X_DAC_EN_2, data); + } + break; + } + return 0; +} + +static int pm860x_mic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Enable Mic1 Bias & MICDET, HSDET */ + snd_soc_update_bits(codec, PM860X_ADC_ANA_1, MIC1BIAS_MASK, + MIC1BIAS_MASK); + pm860x_set_bits(codec->control_data, REG_MIC_DET, + MICDET_MASK, MICDET_MASK); + pm860x_set_bits(codec->control_data, REG_HS_DET, + EN_HS_DET, EN_HS_DET); + break; + case SND_SOC_DAPM_PRE_PMD: + /* disable Mic1 Bias & MICDET, HSDET */ + pm860x_set_bits(codec->control_data, REG_MIC_DET, + MICDET_MASK, 0); + pm860x_set_bits(codec->control_data, REG_HS_DET, + EN_HS_DET, 0); + snd_soc_update_bits(codec, PM860X_ADC_ANA_1, MIC1BIAS_MASK, 0); + break; + } + return 0; +} + +static const char *pm860x_opamp_texts[] = {"-50%", "-25%", "0%", "75%"}; + +static const char *pm860x_pa_texts[] = {"-33%", "0%", "33%", "66%"}; + +static const struct soc_enum pm860x_hs1_opamp_enum = + SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts); + +static const struct soc_enum pm860x_hs2_opamp_enum = + SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 5, 4, pm860x_opamp_texts); + +static const struct soc_enum pm860x_hs1_pa_enum = + SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 3, 4, pm860x_pa_texts); + +static const struct soc_enum pm860x_hs2_pa_enum = + SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 3, 4, pm860x_pa_texts); + +static const struct soc_enum pm860x_lo1_opamp_enum = + SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 5, 4, pm860x_opamp_texts); + +static const struct soc_enum pm860x_lo2_opamp_enum = + SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 5, 4, pm860x_opamp_texts); + +static const struct soc_enum pm860x_lo1_pa_enum = + SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 3, 4, pm860x_pa_texts); + +static const struct soc_enum pm860x_lo2_pa_enum = + SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 3, 4, pm860x_pa_texts); + +static const struct soc_enum pm860x_spk_pa_enum = + SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 5, 4, pm860x_pa_texts); + +static const struct soc_enum pm860x_ear_pa_enum = + SOC_ENUM_SINGLE(PM860X_EAR_CTRL_2, 0, 4, pm860x_pa_texts); + +static const struct soc_enum pm860x_spk_ear_opamp_enum = + SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 3, 4, pm860x_opamp_texts); + +static const struct snd_kcontrol_new pm860x_snd_controls[] = { + SOC_DOUBLE_R_TLV("ADC Capture Volume", PM860X_ADC_ANA_2, + PM860X_ADC_ANA_3, 6, 3, 0, adc_tlv), + SOC_DOUBLE_TLV("AUX Capture Volume", PM860X_ADC_ANA_3, 0, 3, 7, 0, + aux_tlv), + SOC_SINGLE_TLV("MIC1 Capture Volume", PM860X_ADC_ANA_2, 0, 7, 0, + mic_tlv), + SOC_SINGLE_TLV("MIC3 Capture Volume", PM860X_ADC_ANA_2, 3, 7, 0, + mic_tlv), + SOC_DOUBLE_R_EXT_TLV("Sidetone Volume", PM860X_SIDETONE_L_GAIN, + PM860X_SIDETONE_R_GAIN, 0, ARRAY_SIZE(st_table)-1, + 0, snd_soc_get_volsw_2r_st, + snd_soc_put_volsw_2r_st, st_tlv), + SOC_SINGLE_TLV("Speaker Playback Volume", PM860X_EAR_CTRL_1, + 0, 7, 0, out_tlv), + SOC_DOUBLE_R_TLV("Line Playback Volume", PM860X_LO1_CTRL, + PM860X_LO2_CTRL, 0, 7, 0, out_tlv), + SOC_DOUBLE_R_TLV("Headset Playback Volume", PM860X_HS1_CTRL, + PM860X_HS2_CTRL, 0, 7, 0, out_tlv), + SOC_DOUBLE_R_EXT_TLV("Hifi Left Playback Volume", + PM860X_HIFIL_GAIN_LEFT, + PM860X_HIFIL_GAIN_RIGHT, 0, 63, 0, + snd_soc_get_volsw_2r_out, + snd_soc_put_volsw_2r_out, dpga_tlv), + SOC_DOUBLE_R_EXT_TLV("Hifi Right Playback Volume", + PM860X_HIFIR_GAIN_LEFT, + PM860X_HIFIR_GAIN_RIGHT, 0, 63, 0, + snd_soc_get_volsw_2r_out, + snd_soc_put_volsw_2r_out, dpga_tlv), + SOC_DOUBLE_R_EXT_TLV("Lofi Playback Volume", PM860X_LOFI_GAIN_LEFT, + PM860X_LOFI_GAIN_RIGHT, 0, 63, 0, + snd_soc_get_volsw_2r_out, + snd_soc_put_volsw_2r_out, dpga_tlv), + SOC_ENUM("Headset1 Operational Amplifier Current", + pm860x_hs1_opamp_enum), + SOC_ENUM("Headset2 Operational Amplifier Current", + pm860x_hs2_opamp_enum), + SOC_ENUM("Headset1 Amplifier Current", pm860x_hs1_pa_enum), + SOC_ENUM("Headset2 Amplifier Current", pm860x_hs2_pa_enum), + SOC_ENUM("Lineout1 Operational Amplifier Current", + pm860x_lo1_opamp_enum), + SOC_ENUM("Lineout2 Operational Amplifier Current", + pm860x_lo2_opamp_enum), + SOC_ENUM("Lineout1 Amplifier Current", pm860x_lo1_pa_enum), + SOC_ENUM("Lineout2 Amplifier Current", pm860x_lo2_pa_enum), + SOC_ENUM("Speaker Operational Amplifier Current", + pm860x_spk_ear_opamp_enum), + SOC_ENUM("Speaker Amplifier Current", pm860x_spk_pa_enum), + SOC_ENUM("Earpiece Amplifier Current", pm860x_ear_pa_enum), +}; + +/* + * DAPM Controls + */ + +/* PCM Switch / PCM Interface */ +static const struct snd_kcontrol_new pcm_switch_controls = + SOC_DAPM_SINGLE("Switch", PM860X_ADC_EN_2, 0, 1, 0); + +/* AUX1 Switch */ +static const struct snd_kcontrol_new aux1_switch_controls = + SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 4, 1, 0); + +/* AUX2 Switch */ +static const struct snd_kcontrol_new aux2_switch_controls = + SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 5, 1, 0); + +/* Left Ex. PA Switch */ +static const struct snd_kcontrol_new lepa_switch_controls = + SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 2, 1, 0); + +/* Right Ex. PA Switch */ +static const struct snd_kcontrol_new repa_switch_controls = + SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 1, 1, 0); + +/* PCM Mux / Mux7 */ +static const char *aif1_text[] = { + "PCM L", "PCM R", +}; + +static const struct soc_enum aif1_enum = + SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 6, 2, aif1_text); + +static const struct snd_kcontrol_new aif1_mux = + SOC_DAPM_ENUM("PCM Mux", aif1_enum); + +/* I2S Mux / Mux9 */ +static const char *i2s_din_text[] = { + "DIN", "DIN1", +}; + +static const struct soc_enum i2s_din_enum = + SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 1, 2, i2s_din_text); + +static const struct snd_kcontrol_new i2s_din_mux = + SOC_DAPM_ENUM("I2S DIN Mux", i2s_din_enum); + +/* I2S Mic Mux / Mux8 */ +static const char *i2s_mic_text[] = { + "Ex PA", "ADC", +}; + +static const struct soc_enum i2s_mic_enum = + SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 4, 2, i2s_mic_text); + +static const struct snd_kcontrol_new i2s_mic_mux = + SOC_DAPM_ENUM("I2S Mic Mux", i2s_mic_enum); + +/* ADCL Mux / Mux2 */ +static const char *adcl_text[] = { + "ADCR", "ADCL", +}; + +static const struct soc_enum adcl_enum = + SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 4, 2, adcl_text); + +static const struct snd_kcontrol_new adcl_mux = + SOC_DAPM_ENUM("ADC Left Mux", adcl_enum); + +/* ADCR Mux / Mux3 */ +static const char *adcr_text[] = { + "ADCL", "ADCR", +}; + +static const struct soc_enum adcr_enum = + SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 2, 2, adcr_text); + +static const struct snd_kcontrol_new adcr_mux = + SOC_DAPM_ENUM("ADC Right Mux", adcr_enum); + +/* ADCR EC Mux / Mux6 */ +static const char *adcr_ec_text[] = { + "ADCR", "EC", +}; + +static const struct soc_enum adcr_ec_enum = + SOC_ENUM_SINGLE(PM860X_ADC_EN_2, 3, 2, adcr_ec_text); + +static const struct snd_kcontrol_new adcr_ec_mux = + SOC_DAPM_ENUM("ADCR EC Mux", adcr_ec_enum); + +/* EC Mux / Mux4 */ +static const char *ec_text[] = { + "Left", "Right", "Left + Right", +}; + +static const struct soc_enum ec_enum = + SOC_ENUM_SINGLE(PM860X_EC_PATH, 1, 3, ec_text); + +static const struct snd_kcontrol_new ec_mux = + SOC_DAPM_ENUM("EC Mux", ec_enum); + +static const char *dac_text[] = { + "No input", "Right", "Left", "No input", +}; + +/* DAC Headset 1 Mux / Mux10 */ +static const struct soc_enum dac_hs1_enum = + SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 0, 4, dac_text); + +static const struct snd_kcontrol_new dac_hs1_mux = + SOC_DAPM_ENUM("DAC HS1 Mux", dac_hs1_enum); + +/* DAC Headset 2 Mux / Mux11 */ +static const struct soc_enum dac_hs2_enum = + SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 2, 4, dac_text); + +static const struct snd_kcontrol_new dac_hs2_mux = + SOC_DAPM_ENUM("DAC HS2 Mux", dac_hs2_enum); + +/* DAC Lineout 1 Mux / Mux12 */ +static const struct soc_enum dac_lo1_enum = + SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 4, 4, dac_text); + +static const struct snd_kcontrol_new dac_lo1_mux = + SOC_DAPM_ENUM("DAC LO1 Mux", dac_lo1_enum); + +/* DAC Lineout 2 Mux / Mux13 */ +static const struct soc_enum dac_lo2_enum = + SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 6, 4, dac_text); + +static const struct snd_kcontrol_new dac_lo2_mux = + SOC_DAPM_ENUM("DAC LO2 Mux", dac_lo2_enum); + +/* DAC Spearker Earphone Mux / Mux14 */ +static const struct soc_enum dac_spk_ear_enum = + SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_2, 0, 4, dac_text); + +static const struct snd_kcontrol_new dac_spk_ear_mux = + SOC_DAPM_ENUM("DAC SP Mux", dac_spk_ear_enum); + +/* Headset 1 Mux / Mux15 */ +static const char *in_text[] = { + "Digital", "Analog", +}; + +static const struct soc_enum hs1_enum = + SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 0, 2, in_text); + +static const struct snd_kcontrol_new hs1_mux = + SOC_DAPM_ENUM("Headset1 Mux", hs1_enum); + +/* Headset 2 Mux / Mux16 */ +static const struct soc_enum hs2_enum = + SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 1, 2, in_text); + +static const struct snd_kcontrol_new hs2_mux = + SOC_DAPM_ENUM("Headset2 Mux", hs2_enum); + +/* Lineout 1 Mux / Mux17 */ +static const struct soc_enum lo1_enum = + SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 2, 2, in_text); + +static const struct snd_kcontrol_new lo1_mux = + SOC_DAPM_ENUM("Lineout1 Mux", lo1_enum); + +/* Lineout 2 Mux / Mux18 */ +static const struct soc_enum lo2_enum = + SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 3, 2, in_text); + +static const struct snd_kcontrol_new lo2_mux = + SOC_DAPM_ENUM("Lineout2 Mux", lo2_enum); + +/* Speaker Earpiece Demux */ +static const char *spk_text[] = { + "Earpiece", "Speaker", +}; + +static const struct soc_enum spk_enum = + SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 6, 2, spk_text); + +static const struct snd_kcontrol_new spk_demux = + SOC_DAPM_ENUM("Speaker Earpiece Demux", spk_enum); + +/* MIC Mux / Mux1 */ +static const char *mic_text[] = { + "Mic 1", "Mic 2", +}; + +static const struct soc_enum mic_enum = + SOC_ENUM_SINGLE(PM860X_ADC_ANA_4, 4, 2, mic_text); + +static const struct snd_kcontrol_new mic_mux = + SOC_DAPM_ENUM("MIC Mux", mic_enum); + +static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("PCM SDI", "PCM Playback", 0, + PM860X_ADC_EN_2, 0, 0), + SND_SOC_DAPM_AIF_OUT("PCM SDO", "PCM Capture", 0, + PM860X_PCM_IFACE_3, 1, 1), + + + SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0, + PM860X_DAC_EN_2, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0, + PM860X_DAC_EN_2, 0, 0), + SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0, + PM860X_I2S_IFACE_3, 5, 1), + SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux), + SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux), + SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux), + SND_SOC_DAPM_MUX("EC Mux", SND_SOC_NOPM, 0, 0, &ec_mux), + SND_SOC_DAPM_MUX("ADCR EC Mux", SND_SOC_NOPM, 0, 0, &adcr_ec_mux), + SND_SOC_DAPM_SWITCH("Left EPA", SND_SOC_NOPM, 0, 0, + &lepa_switch_controls), + SND_SOC_DAPM_SWITCH("Right EPA", SND_SOC_NOPM, 0, 0, + &repa_switch_controls), + + SND_SOC_DAPM_ADC_E("Left ADC", NULL, SND_SOC_NOPM, 0, 0, + pm860x_adc_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("Right ADC", NULL, SND_SOC_NOPM, 0, 0, + pm860x_adc_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_SWITCH("AUX1 Switch", SND_SOC_NOPM, 0, 0, + &aux1_switch_controls), + SND_SOC_DAPM_SWITCH("AUX2 Switch", SND_SOC_NOPM, 0, 0, + &aux2_switch_controls), + + SND_SOC_DAPM_MUX("MIC Mux", SND_SOC_NOPM, 0, 0, &mic_mux), + SND_SOC_DAPM_MICBIAS_E("Mic1 Bias", SND_SOC_NOPM, 0, 0, + &pm860x_mic1_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MICBIAS("Mic3 Bias", PM860X_ADC_ANA_1, 7, 0), + SND_SOC_DAPM_PGA("MIC1 Volume", PM860X_ADC_EN_1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC3 Volume", PM860X_ADC_EN_1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX1 Volume", PM860X_ADC_EN_1, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX2 Volume", PM860X_ADC_EN_1, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Sidetone PGA", PM860X_ADC_EN_2, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Lofi PGA", PM860X_ADC_EN_2, 2, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("AUX1"), + SND_SOC_DAPM_INPUT("AUX2"), + SND_SOC_DAPM_INPUT("MIC1P"), + SND_SOC_DAPM_INPUT("MIC1N"), + SND_SOC_DAPM_INPUT("MIC2P"), + SND_SOC_DAPM_INPUT("MIC2N"), + SND_SOC_DAPM_INPUT("MIC3P"), + SND_SOC_DAPM_INPUT("MIC3N"), + + SND_SOC_DAPM_DAC_E("Left DAC", NULL, SND_SOC_NOPM, 0, 0, + pm860x_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("Right DAC", NULL, SND_SOC_NOPM, 0, 0, + pm860x_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_MUX("I2S DIN Mux", SND_SOC_NOPM, 0, 0, &i2s_din_mux), + SND_SOC_DAPM_MUX("DAC HS1 Mux", SND_SOC_NOPM, 0, 0, &dac_hs1_mux), + SND_SOC_DAPM_MUX("DAC HS2 Mux", SND_SOC_NOPM, 0, 0, &dac_hs2_mux), + SND_SOC_DAPM_MUX("DAC LO1 Mux", SND_SOC_NOPM, 0, 0, &dac_lo1_mux), + SND_SOC_DAPM_MUX("DAC LO2 Mux", SND_SOC_NOPM, 0, 0, &dac_lo2_mux), + SND_SOC_DAPM_MUX("DAC SP Mux", SND_SOC_NOPM, 0, 0, &dac_spk_ear_mux), + SND_SOC_DAPM_MUX("Headset1 Mux", SND_SOC_NOPM, 0, 0, &hs1_mux), + SND_SOC_DAPM_MUX("Headset2 Mux", SND_SOC_NOPM, 0, 0, &hs2_mux), + SND_SOC_DAPM_MUX("Lineout1 Mux", SND_SOC_NOPM, 0, 0, &lo1_mux), + SND_SOC_DAPM_MUX("Lineout2 Mux", SND_SOC_NOPM, 0, 0, &lo2_mux), + SND_SOC_DAPM_MUX("Speaker Earpiece Demux", SND_SOC_NOPM, 0, 0, + &spk_demux), + + + SND_SOC_DAPM_PGA("Headset1 PGA", PM860X_DAC_EN_1, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headset2 PGA", PM860X_DAC_EN_1, 1, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("HS1"), + SND_SOC_DAPM_OUTPUT("HS2"), + SND_SOC_DAPM_PGA("Lineout1 PGA", PM860X_DAC_EN_1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Lineout2 PGA", PM860X_DAC_EN_1, 3, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("LINEOUT1"), + SND_SOC_DAPM_OUTPUT("LINEOUT2"), + SND_SOC_DAPM_PGA("Earpiece PGA", PM860X_DAC_EN_1, 4, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("EARP"), + SND_SOC_DAPM_OUTPUT("EARN"), + SND_SOC_DAPM_PGA("Speaker PGA", PM860X_DAC_EN_1, 5, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("LSP"), + SND_SOC_DAPM_OUTPUT("LSN"), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "VCODEC", PM860X_AUDIO_SUPPLIES_2, + 0, SUPPLY_MASK, SUPPLY_MASK, 0), + + SND_SOC_DAPM_POST("RSYNC", pm860x_rsync_event), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* supply */ + {"Left DAC", NULL, "VCODEC"}, + {"Right DAC", NULL, "VCODEC"}, + {"Left ADC", NULL, "VCODEC"}, + {"Right ADC", NULL, "VCODEC"}, + + /* PCM/AIF1 Inputs */ + {"PCM SDO", NULL, "ADC Left Mux"}, + {"PCM SDO", NULL, "ADCR EC Mux"}, + + /* PCM/AFI2 Outputs */ + {"Lofi PGA", NULL, "PCM SDI"}, + {"Lofi PGA", NULL, "Sidetone PGA"}, + {"Left DAC", NULL, "Lofi PGA"}, + {"Right DAC", NULL, "Lofi PGA"}, + + /* I2S/AIF2 Inputs */ + {"MIC Mux", "Mic 1", "MIC1P"}, + {"MIC Mux", "Mic 1", "MIC1N"}, + {"MIC Mux", "Mic 2", "MIC2P"}, + {"MIC Mux", "Mic 2", "MIC2N"}, + {"MIC1 Volume", NULL, "MIC Mux"}, + {"MIC3 Volume", NULL, "MIC3P"}, + {"MIC3 Volume", NULL, "MIC3N"}, + {"Left ADC", NULL, "MIC1 Volume"}, + {"Right ADC", NULL, "MIC3 Volume"}, + {"ADC Left Mux", "ADCR", "Right ADC"}, + {"ADC Left Mux", "ADCL", "Left ADC"}, + {"ADC Right Mux", "ADCL", "Left ADC"}, + {"ADC Right Mux", "ADCR", "Right ADC"}, + {"Left EPA", "Switch", "Left DAC"}, + {"Right EPA", "Switch", "Right DAC"}, + {"EC Mux", "Left", "Left DAC"}, + {"EC Mux", "Right", "Right DAC"}, + {"EC Mux", "Left + Right", "Left DAC"}, + {"EC Mux", "Left + Right", "Right DAC"}, + {"ADCR EC Mux", "ADCR", "ADC Right Mux"}, + {"ADCR EC Mux", "EC", "EC Mux"}, + {"I2S Mic Mux", "Ex PA", "Left EPA"}, + {"I2S Mic Mux", "Ex PA", "Right EPA"}, + {"I2S Mic Mux", "ADC", "ADC Left Mux"}, + {"I2S Mic Mux", "ADC", "ADCR EC Mux"}, + {"I2S DOUT", NULL, "I2S Mic Mux"}, + + /* I2S/AIF2 Outputs */ + {"I2S DIN Mux", "DIN", "I2S DIN"}, + {"I2S DIN Mux", "DIN1", "I2S DIN1"}, + {"Left DAC", NULL, "I2S DIN Mux"}, + {"Right DAC", NULL, "I2S DIN Mux"}, + {"DAC HS1 Mux", "Left", "Left DAC"}, + {"DAC HS1 Mux", "Right", "Right DAC"}, + {"DAC HS2 Mux", "Left", "Left DAC"}, + {"DAC HS2 Mux", "Right", "Right DAC"}, + {"DAC LO1 Mux", "Left", "Left DAC"}, + {"DAC LO1 Mux", "Right", "Right DAC"}, + {"DAC LO2 Mux", "Left", "Left DAC"}, + {"DAC LO2 Mux", "Right", "Right DAC"}, + {"Headset1 Mux", "Digital", "DAC HS1 Mux"}, + {"Headset2 Mux", "Digital", "DAC HS2 Mux"}, + {"Lineout1 Mux", "Digital", "DAC LO1 Mux"}, + {"Lineout2 Mux", "Digital", "DAC LO2 Mux"}, + {"Headset1 PGA", NULL, "Headset1 Mux"}, + {"Headset2 PGA", NULL, "Headset2 Mux"}, + {"Lineout1 PGA", NULL, "Lineout1 Mux"}, + {"Lineout2 PGA", NULL, "Lineout2 Mux"}, + {"DAC SP Mux", "Left", "Left DAC"}, + {"DAC SP Mux", "Right", "Right DAC"}, + {"Speaker Earpiece Demux", "Speaker", "DAC SP Mux"}, + {"Speaker PGA", NULL, "Speaker Earpiece Demux"}, + {"Earpiece PGA", NULL, "Speaker Earpiece Demux"}, + + {"HS1", NULL, "Headset1 PGA"}, + {"HS2", NULL, "Headset2 PGA"}, + {"LINEOUT1", NULL, "Lineout1 PGA"}, + {"LINEOUT2", NULL, "Lineout2 PGA"}, + {"LSP", NULL, "Speaker PGA"}, + {"LSN", NULL, "Speaker PGA"}, + {"EARP", NULL, "Earpiece PGA"}, + {"EARN", NULL, "Earpiece PGA"}, +}; + +static int pm860x_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int data = 0; + + if (mute) + data = DAC_MUTE; + snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, data); + snd_soc_update_bits(codec, PM860X_EAR_CTRL_2, + RSYNC_CHANGE, RSYNC_CHANGE); + return 0; +} + +static int set_dai_fmt(struct pm860x_priv *pm860x, unsigned int fmt, + unsigned char *inf, unsigned char *mask) +{ + *mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + if (pm860x->dir == PM860X_CLK_DIR_OUT) + *inf |= PCM_INF2_MASTER; + else + return -EINVAL; + break; + case SND_SOC_DAIFMT_CBS_CFS: + if (pm860x->dir == PM860X_CLK_DIR_IN) + *inf &= ~PCM_INF2_MASTER; + else + return -EINVAL; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + *inf |= PCM_EXACT_I2S; + break; + default: + return -EINVAL; + } + *mask |= PCM_MODE_MASK; + return 0; +} + +static int pm860x_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned char inf = 0, mask = 0; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + inf &= ~PCM_INF2_18WL; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + inf |= PCM_INF2_18WL; + break; + default: + return -EINVAL; + } + mask |= PCM_INF2_18WL; + snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf); + + /* sample rate */ + switch (params_rate(params)) { + case 8000: + inf = 0; + break; + case 16000: + inf = 3; + break; + case 32000: + inf = 6; + break; + case 48000: + inf = 8; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, PM860X_PCM_RATE, 0x0f, inf); + + return 0; +} + +static int pm860x_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); + unsigned char inf = 0, mask = 0; + int ret; + + ret = set_dai_fmt(pm860x, fmt, &inf, &mask); + if (!ret) + snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf); + return ret; +} + +static int pm860x_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 pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); + + if (dir == PM860X_CLK_DIR_OUT) + pm860x->dir = PM860X_CLK_DIR_OUT; + else + pm860x->dir = PM860X_CLK_DIR_IN; + + return 0; +} + +static int pm860x_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned char inf; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + inf = 0; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + inf = PCM_INF2_18WL; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, PCM_INF2_18WL, inf); + + /* sample rate */ + switch (params_rate(params)) { + case 8000: + inf = 0; + break; + case 11025: + inf = 1; + break; + case 16000: + inf = 3; + break; + case 22050: + inf = 4; + break; + case 32000: + inf = 6; + break; + case 44100: + inf = 7; + break; + case 48000: + inf = 8; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, PM860X_I2S_IFACE_4, 0xf, inf); + + return 0; +} + +static int pm860x_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); + unsigned char inf = 0, mask = 0; + int ret; + + ret = set_dai_fmt(pm860x, fmt, &inf, &mask); + if (!ret) + snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, mask, inf); + return ret; +} + +static int pm860x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int data; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Enable Audio PLL & Audio section */ + data = AUDIO_PLL | AUDIO_SECTION_RESET + | AUDIO_SECTION_ON; + pm860x_reg_write(codec->control_data, 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); + break; + } + codec->bias_level = level; + return 0; +} + +static struct snd_soc_dai_ops pm860x_pcm_dai_ops = { + .digital_mute = pm860x_digital_mute, + .hw_params = pm860x_pcm_hw_params, + .set_fmt = pm860x_pcm_set_dai_fmt, + .set_sysclk = pm860x_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops pm860x_i2s_dai_ops = { + .digital_mute = pm860x_digital_mute, + .hw_params = pm860x_i2s_hw_params, + .set_fmt = pm860x_i2s_set_dai_fmt, + .set_sysclk = pm860x_set_dai_sysclk, +}; + +#define PM860X_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000) + +static struct snd_soc_dai_driver pm860x_dai[] = { + { + /* DAI PCM */ + .name = "88pm860x-pcm", + .id = 1, + .playback = { + .stream_name = "PCM Playback", + .channels_min = 2, + .channels_max = 2, + .rates = PM860X_RATES, + .formats = SNDRV_PCM_FORMAT_S16_LE | \ + SNDRV_PCM_FORMAT_S18_3LE, + }, + .capture = { + .stream_name = "PCM Capture", + .channels_min = 2, + .channels_max = 2, + .rates = PM860X_RATES, + .formats = SNDRV_PCM_FORMAT_S16_LE | \ + SNDRV_PCM_FORMAT_S18_3LE, + }, + .ops = &pm860x_pcm_dai_ops, + }, { + /* DAI I2S */ + .name = "88pm860x-i2s", + .id = 2, + .playback = { + .stream_name = "I2S Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FORMAT_S16_LE | \ + SNDRV_PCM_FORMAT_S18_3LE, + }, + .capture = { + .stream_name = "I2S Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FORMAT_S16_LE | \ + SNDRV_PCM_FORMAT_S18_3LE, + }, + .ops = &pm860x_i2s_dai_ops, + }, +}; + +static irqreturn_t pm860x_codec_handler(int irq, void *data) +{ + struct pm860x_priv *pm860x = data; + int status, shrt, report = 0; + + status = pm860x_reg_read(pm860x->i2c, REG_STATUS_1); + shrt = pm860x_reg_read(pm860x->i2c, REG_SHORTS); + + if (status & HEADSET_STATUS) + report |= PM860X_DET_HEADSET; + if (status & MIC_STATUS) + report |= PM860X_DET_MIC; + if (status & HOOK_STATUS) + report |= PM860X_DET_HOOK; + if (shrt & (SHORT_LO1 | SHORT_LO2)) + report |= PM860X_SHORT_LINEOUT; + if (shrt & (SHORT_HS1 | SHORT_HS2)) + report |= PM860X_SHORT_HEADSET; + dev_dbg(pm860x->codec->dev, "report:0x%x\n", report); + return IRQ_HANDLED; +} + +static int pm860x_probe(struct snd_soc_codec *codec) +{ + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); + int i, ret; + + pm860x->codec = codec; + + codec->control_data = pm860x->i2c; + + for (i = 0; i < 4; i++) { + ret = request_threaded_irq(pm860x->irq[i], NULL, + pm860x_codec_handler, IRQF_ONESHOT, + pm860x->name[i], pm860x); + if (ret < 0) { + dev_err(codec->dev, "Failed to request IRQ!\n"); + goto out_irq; + } + } + + 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_codec; + } + + snd_soc_add_controls(codec, pm860x_snd_controls, + ARRAY_SIZE(pm860x_snd_controls)); + snd_soc_dapm_new_controls(codec, pm860x_dapm_widgets, + ARRAY_SIZE(pm860x_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + return 0; + +out_codec: + i = 3; +out_irq: + for (; i >= 0; i--) + free_irq(pm860x->irq[i], pm860x); + return -EINVAL; +} + +static int pm860x_remove(struct snd_soc_codec *codec) +{ + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 3; i >= 0; i--) + free_irq(pm860x->irq[i], pm860x); + pm860x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +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, +}; + +static int __devinit pm860x_codec_probe(struct platform_device *pdev) +{ + struct pm860x_chip *chip = dev_get_drvdata(pdev->dev.parent); + struct pm860x_priv *pm860x; + struct resource *res; + int i, ret; + + pm860x = kzalloc(sizeof(struct pm860x_priv), GFP_KERNEL); + if (pm860x == NULL) + return -ENOMEM; + + pm860x->chip = chip; + pm860x->i2c = (chip->id == CHIP_PM8607) ? chip->client + : chip->companion; + platform_set_drvdata(pdev, pm860x); + + for (i = 0; i < 4; i++) { + res = platform_get_resource(pdev, IORESOURCE_IRQ, i); + if (!res) { + dev_err(&pdev->dev, "Failed to get IRQ resources\n"); + goto out; + } + pm860x->irq[i] = res->start + chip->irq_base; + strncpy(pm860x->name[i], res->name, MAX_NAME_LEN); + } + + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pm860x, + pm860x_dai, ARRAY_SIZE(pm860x_dai)); + if (ret) { + dev_err(&pdev->dev, "Failed to register codec\n"); + goto out; + } + return ret; + +out: + platform_set_drvdata(pdev, NULL); + kfree(pm860x); + return -EINVAL; +} + +static int __devexit pm860x_codec_remove(struct platform_device *pdev) +{ + struct pm860x_priv *pm860x = platform_get_drvdata(pdev); + + snd_soc_unregister_codec(&pdev->dev); + platform_set_drvdata(pdev, NULL); + kfree(pm860x); + return 0; +} + +static struct platform_driver pm860x_codec_driver = { + .driver = { + .name = "88pm860x-codec", + .owner = THIS_MODULE, + }, + .probe = pm860x_codec_probe, + .remove = __devexit_p(pm860x_codec_remove), +}; + +static __init int pm860x_init(void) +{ + return platform_driver_register(&pm860x_codec_driver); +} +module_init(pm860x_init); + +static __exit void pm860x_exit(void) +{ + platform_driver_unregister(&pm860x_codec_driver); +} +module_exit(pm860x_exit); + +MODULE_DESCRIPTION("ASoC 88PM860x driver"); +MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang(a)marvell.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:88pm860x-codec"); + diff --git a/sound/soc/codecs/88pm860x-codec.h b/sound/soc/codecs/88pm860x-codec.h new file mode 100644 index 0000000..b034de2 --- /dev/null +++ b/sound/soc/codecs/88pm860x-codec.h @@ -0,0 +1,92 @@ +/* + * 88pm860x-codec.h -- 88PM860x ALSA SoC Audio Driver + * + * Copyright 2010 Marvell International Ltd. + * Haojian Zhuang <haojian.zhuang(a)marvell.com> + * + * 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 __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_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_PCM_IFACE_4 0x39 +#define PM860X_I2S_IFACE_5 0x3a + +#define PM860X_SHORTS 0x3b +#define PM860X_PLL_ADJ_1 0x3c +#define PM860X_PLL_ADJ_2 0x3d + +/* bits definition */ +#define PM860X_CLK_DIR_IN 0 +#define PM860X_CLK_DIR_OUT 1 + +#define PM860X_DET_HEADSET (1 << 0) +#define PM860X_DET_MIC (1 << 1) +#define PM860X_DET_HOOK (1 << 2) +#define PM860X_SHORT_HEADSET (1 << 3) +#define PM860X_SHORT_LINEOUT (1 << 4) +#define PM860X_DET_MASK 0x1F + +#endif /* __88PM860X_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bfdd92b..a3cfc18 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" + select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS select SND_SOC_AD1836 if SPI_MASTER @@ -85,6 +86,9 @@ config SND_SOC_ALL_CODECS If unsure select "N". +config SND_SOC_88PM860X + tristate + config SND_SOC_WM_HUBS tristate default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9c3c39f..b9c4358 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,3 +1,4 @@ +snd-soc-88pm860x-objs := 88pm860x-codec.o snd-soc-ac97-objs := ac97.o snd-soc-ad1836-objs := ad1836.o snd-soc-ad193x-objs := ad193x.o @@ -67,6 +68,7 @@ snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-wm2000-objs := wm2000.o snd-soc-wm9090-objs := wm9090.o +obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o -- 1.5.6.5 --0016e6d63f44021a1f048e002b69 Content-Type: text/x-patch; charset=US-ASCII; name="0001-ASoC-add-88pm860x-codec-driver.patch" Content-Disposition: attachment; filename="0001-ASoC-add-88pm860x-codec-driver.patch" Content-Transfer-Encoding: base64 X-Attachment-Id: f_gcyfxo2w0 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