diff options
31 files changed, 2695 insertions, 447 deletions
diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index a5e63fa47dc5..c454e67f54bb 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -4,7 +4,7 @@ This device supports I2C only. Required properties: -- compatible : "maxim,max98090". +- compatible : "maxim,max98090" or "maxim,max98091". - reg : The I2C address of the device. diff --git a/arch/x86/include/asm/platform_sst_audio.h b/arch/x86/include/asm/platform_sst_audio.h new file mode 100644 index 000000000000..0a4e140315b6 --- /dev/null +++ b/arch/x86/include/asm/platform_sst_audio.h @@ -0,0 +1,78 @@ +/* + * platform_sst_audio.h: sst audio platform data header file + * + * Copyright (C) 2012-14 Intel Corporation + * Author: Jeeja KP <jeeja.kp@intel.com> + * Omair Mohammed Abdullah <omair.m.abdullah@intel.com> + * Vinod Koul ,vinod.koul@intel.com> + * + * 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; version 2 + * of the License. + */ +#ifndef _PLATFORM_SST_AUDIO_H_ +#define _PLATFORM_SST_AUDIO_H_ + +#include <linux/sfi.h> + +enum sst_audio_task_id_mrfld { + SST_TASK_ID_NONE = 0, + SST_TASK_ID_SBA = 1, + SST_TASK_ID_MEDIA = 3, + SST_TASK_ID_MAX = SST_TASK_ID_MEDIA, +}; + +/* Device IDs for Merrifield are Pipe IDs, + * ref: DSP spec v0.75 */ +enum sst_audio_device_id_mrfld { + /* Output pipeline IDs */ + PIPE_ID_OUT_START = 0x0, + PIPE_CODEC_OUT0 = 0x2, + PIPE_CODEC_OUT1 = 0x3, + PIPE_SPROT_LOOP_OUT = 0x4, + PIPE_MEDIA_LOOP1_OUT = 0x5, + PIPE_MEDIA_LOOP2_OUT = 0x6, + PIPE_VOIP_OUT = 0xC, + PIPE_PCM0_OUT = 0xD, + PIPE_PCM1_OUT = 0xE, + PIPE_PCM2_OUT = 0xF, + PIPE_MEDIA0_OUT = 0x12, + PIPE_MEDIA1_OUT = 0x13, +/* Input Pipeline IDs */ + PIPE_ID_IN_START = 0x80, + PIPE_CODEC_IN0 = 0x82, + PIPE_CODEC_IN1 = 0x83, + PIPE_SPROT_LOOP_IN = 0x84, + PIPE_MEDIA_LOOP1_IN = 0x85, + PIPE_MEDIA_LOOP2_IN = 0x86, + PIPE_VOIP_IN = 0x8C, + PIPE_PCM0_IN = 0x8D, + PIPE_PCM1_IN = 0x8E, + PIPE_MEDIA0_IN = 0x8F, + PIPE_MEDIA1_IN = 0x90, + PIPE_MEDIA2_IN = 0x91, + PIPE_RSVD = 0xFF, +}; + +/* The stream map for each platform consists of an array of the below + * stream map structure. + */ +struct sst_dev_stream_map { + u8 dev_num; /* device id */ + u8 subdev_num; /* substream */ + u8 direction; + u8 device_id; /* fw id */ + u8 task_id; /* fw task */ + u8 status; +}; + +struct sst_platform_data { + /* Intel software platform id*/ + struct sst_dev_stream_map *pdev_strm_map; + unsigned int strm_map_size; +}; + +int add_sst_platform_device(void); +#endif + diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 000000000000..eb773d1485f2 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * 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 __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool gpio2_en; /*GPIO2 enable*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e4a1d2aece36..4c7542571484 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER + select SND_SOC_RT286 if I2C select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_RT5645 if I2C @@ -455,6 +456,9 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m +config SND_SOC_RT286 + tristate + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 97b80a1e03af..ade412e49bd0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -69,6 +69,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rl6231-objs := rl6231.o +snd-soc-rt286-objs := rt286.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o @@ -237,6 +238,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index d97f1ce7ff7d..4a063fa88526 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -26,10 +26,6 @@ #include <sound/max98090.h> #include "max98090.h" -#define DEBUG -#define EXTMIC_METHOD -#define EXTMIC_METHOD_TEST - /* Allows for sparsely populated register maps */ static struct reg_default max98090_reg[] = { { 0x00, 0x00 }, /* 00 Software Reset */ @@ -820,7 +816,6 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w, else val = (val & M98090_MIC_PA2EN_MASK) >> M98090_MIC_PA2EN_SHIFT; - if (val >= 1) { if (w->reg == M98090_REG_MIC1_INPUT_LEVEL) { max98090->pa1en = val - 1; /* Update for volatile */ @@ -1140,7 +1135,6 @@ static const struct snd_kcontrol_new max98090_mixhprsel_mux = SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_INPUT("DMICL"), @@ -1304,7 +1298,6 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { }; static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { - SND_SOC_DAPM_INPUT("DMIC3"), SND_SOC_DAPM_INPUT("DMIC4"), @@ -1315,7 +1308,6 @@ static const struct snd_soc_dapm_widget max98091_dapm_widgets[] = { }; static const struct snd_soc_dapm_route max98090_dapm_routes[] = { - {"MIC1 Input", NULL, "MIC1"}, {"MIC2 Input", NULL, "MIC2"}, @@ -1493,17 +1485,14 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { {"SPKR", NULL, "SPK Right Out"}, {"RCVL", NULL, "RCV Left Out"}, {"RCVR", NULL, "RCV Right Out"}, - }; static const struct snd_soc_dapm_route max98091_dapm_routes[] = { - /* DMIC inputs */ {"DMIC3", NULL, "DMIC3_ENA"}, {"DMIC4", NULL, "DMIC4_ENA"}, {"DMIC3", NULL, "AHPF"}, {"DMIC4", NULL, "AHPF"}, - }; static int max98090_add_widgets(struct snd_soc_codec *codec) @@ -1531,7 +1520,6 @@ static int max98090_add_widgets(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, max98091_dapm_routes, ARRAY_SIZE(max98091_dapm_routes)); - } return 0; @@ -2212,22 +2200,11 @@ static struct snd_soc_dai_driver max98090_dai[] = { } }; -static void max98090_handle_pdata(struct snd_soc_codec *codec) -{ - struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); - struct max98090_pdata *pdata = max98090->pdata; - - if (!pdata) { - dev_err(codec->dev, "No platform data\n"); - return; - } - -} - static int max98090_probe(struct snd_soc_codec *codec) { struct max98090_priv *max98090 = snd_soc_codec_get_drvdata(codec); struct max98090_cdata *cdata; + enum max98090_type devtype; int ret = 0; dev_dbg(codec->dev, "max98090_probe\n"); @@ -2263,16 +2240,21 @@ static int max98090_probe(struct snd_soc_codec *codec) } if ((ret >= M98090_REVA) && (ret <= M98090_REVA + 0x0f)) { - max98090->devtype = MAX98090; + devtype = MAX98090; dev_info(codec->dev, "MAX98090 REVID=0x%02x\n", ret); } else if ((ret >= M98091_REVA) && (ret <= M98091_REVA + 0x0f)) { - max98090->devtype = MAX98091; + devtype = MAX98091; dev_info(codec->dev, "MAX98091 REVID=0x%02x\n", ret); } else { - max98090->devtype = MAX98090; + devtype = MAX98090; dev_err(codec->dev, "Unrecognized revision 0x%02x\n", ret); } + if (max98090->devtype != devtype) { + dev_warn(codec->dev, "Mismatch in DT specified CODEC type.\n"); + max98090->devtype = devtype; + } + max98090->jack_state = M98090_JACK_STATE_NO_HEADSET; INIT_DELAYED_WORK(&max98090->jack_work, max98090_jack_work); @@ -2317,8 +2299,6 @@ static int max98090_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, M98090_REG_MIC_BIAS_VOLTAGE, M98090_MBVSEL_MASK, M98090_MBVSEL_2V8); - max98090_handle_pdata(codec); - max98090_add_widgets(codec); err_access: @@ -2428,7 +2408,7 @@ static int max98090_runtime_suspend(struct device *dev) } #endif -#ifdef CONFIG_PM +#ifdef CONFIG_PM_SLEEP static int max98090_resume(struct device *dev) { struct max98090_priv *max98090 = dev_get_drvdata(dev); @@ -2460,12 +2440,14 @@ static const struct dev_pm_ops max98090_pm = { static const struct i2c_device_id max98090_i2c_id[] = { { "max98090", MAX98090 }, + { "max98091", MAX98091 }, { } }; MODULE_DEVICE_TABLE(i2c, max98090_i2c_id); static const struct of_device_id max98090_of_match[] = { { .compatible = "maxim,max98090", }, + { .compatible = "maxim,max98091", }, { } }; MODULE_DEVICE_TABLE(of, max98090_of_match); diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 9965277b595a..388f90a597fa 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -766,11 +766,11 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = of_property_read_u32(np, "adc-port", &priv->adc_ssi_port); if (ret) - return ret; + goto out; ret = of_property_read_u32(np, "dac-port", &priv->dac_ssi_port); if (ret) - return ret; + goto out; } dev_set_drvdata(&pdev->dev, priv); @@ -783,6 +783,8 @@ static int __init mc13783_codec_probe(struct platform_device *pdev) ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, mc13783_dai_async, ARRAY_SIZE(mc13783_dai_async)); +out: + of_node_put(np); return ret; } diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 000000000000..218f86efd196 --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1224 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao@realtek.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/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.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> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/jack.h> +#include <linux/workqueue.h> +#include <sound/rt286.h> +#include <sound/hda_verbs.h> + +#include "rt286.h" + +#define RT286_VENDOR_ID 0x10ec0286 + +struct rt286_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + struct rt286_platform_data pdata; + struct i2c_client *i2c; + struct snd_soc_jack *jack; + struct delayed_work jack_detect_work; + int sys_clk; + struct reg_default *index_cache; +}; + +static struct reg_default rt286_index_def[] = { + { 0x01, 0xaaaa }, + { 0x02, 0x8aaa }, + { 0x03, 0x0002 }, + { 0x04, 0xaf01 }, + { 0x08, 0x000d }, + { 0x09, 0xd810 }, + { 0x0a, 0x0060 }, + { 0x0b, 0x0000 }, + { 0x0d, 0x2800 }, + { 0x0f, 0x0000 }, + { 0x19, 0x0a17 }, + { 0x20, 0x0020 }, + { 0x33, 0x0208 }, + { 0x49, 0x0004 }, + { 0x4f, 0x50e9 }, + { 0x50, 0x2c00 }, + { 0x63, 0x2902 }, + { 0x67, 0x1111 }, + { 0x68, 0x1016 }, + { 0x69, 0x273f }, +}; +#define INDEX_CACHE_SIZE ARRAY_SIZE(rt286_index_def) + +static const struct reg_default rt286_reg[] = { + { 0x00170500, 0x00000400 }, + { 0x00220000, 0x00000031 }, + { 0x00239000, 0x0000007f }, + { 0x0023a000, 0x0000007f }, + { 0x00270500, 0x00000400 }, + { 0x00370500, 0x00000400 }, + { 0x00870500, 0x00000400 }, + { 0x00920000, 0x00000031 }, + { 0x00935000, 0x000000c3 }, + { 0x00936000, 0x000000c3 }, + { 0x00970500, 0x00000400 }, + { 0x00b37000, 0x00000097 }, + { 0x00b37200, 0x00000097 }, + { 0x00b37300, 0x00000097 }, + { 0x00c37000, 0x00000000 }, + { 0x00c37100, 0x00000080 }, + { 0x01270500, 0x00000400 }, + { 0x01370500, 0x00000400 }, + { 0x01371f00, 0x411111f0 }, + { 0x01439000, 0x00000080 }, + { 0x0143a000, 0x00000080 }, + { 0x01470700, 0x00000000 }, + { 0x01470500, 0x00000400 }, + { 0x01470c00, 0x00000000 }, + { 0x01470100, 0x00000000 }, + { 0x01837000, 0x00000000 }, + { 0x01870500, 0x00000400 }, + { 0x02050000, 0x00000000 }, + { 0x02139000, 0x00000080 }, + { 0x0213a000, 0x00000080 }, + { 0x02170100, 0x00000000 }, + { 0x02170500, 0x00000400 }, + { 0x02170700, 0x00000000 }, + { 0x02270100, 0x00000000 }, + { 0x02370100, 0x00000000 }, + { 0x02040000, 0x00004002 }, + { 0x01870700, 0x00000020 }, + { 0x00830000, 0x000000c3 }, + { 0x00930000, 0x000000c3 }, + { 0x01270700, 0x00000000 }, +}; + +static bool rt286_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT286_GET_HP_SENSE: + case RT286_GET_MIC1_SENSE: + case RT286_PROC_COEF: + return true; + default: + return false; + } + + +} + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0 ... 0xff: + case RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID): + case RT286_GET_HP_SENSE: + case RT286_GET_MIC1_SENSE: + case RT286_SET_AUDIO_POWER: + case RT286_SET_HPO_POWER: + case RT286_SET_SPK_POWER: + case RT286_SET_DMIC1_POWER: + case RT286_SPK_MUX: + case RT286_HPO_MUX: + case RT286_ADC0_MUX: + case RT286_ADC1_MUX: + case RT286_SET_MIC1: + case RT286_SET_PIN_HPO: + case RT286_SET_PIN_SPK: + case RT286_SET_PIN_DMIC1: + case RT286_SPK_EAPD: + case RT286_SET_AMP_GAIN_HPO: + case RT286_SET_DMIC2_DEFAULT: + case RT286_DACL_GAIN: + case RT286_DACR_GAIN: + case RT286_ADCL_GAIN: + case RT286_ADCR_GAIN: + case RT286_MIC_GAIN: + case RT286_SPOL_GAIN: + case RT286_SPOR_GAIN: + case RT286_HPOL_GAIN: + case RT286_HPOR_GAIN: + case RT286_F_DAC_SWITCH: + case RT286_F_RECMIX_SWITCH: + case RT286_REC_MIC_SWITCH: + case RT286_REC_I2S_SWITCH: + case RT286_REC_LINE_SWITCH: + case RT286_REC_BEEP_SWITCH: + case RT286_DAC_FORMAT: + case RT286_ADC_FORMAT: + case RT286_COEF_INDEX: + case RT286_PROC_COEF: + case RT286_SET_AMP_GAIN_ADC_IN1: + case RT286_SET_AMP_GAIN_ADC_IN2: + case RT286_SET_POWER(RT286_DAC_OUT1): + case RT286_SET_POWER(RT286_DAC_OUT2): + case RT286_SET_POWER(RT286_ADC_IN1): + case RT286_SET_POWER(RT286_ADC_IN2): + case RT286_SET_POWER(RT286_DMIC2): + case RT286_SET_POWER(RT286_MIC1): + return true; + default: + return false; + } +} + +static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + struct rt286_priv *rt286 = i2c_get_clientdata(client); + u8 data[4]; + int ret, i; + + /*handle index registers*/ + if (reg <= 0xff) { + rt286_hw_write(client, RT286_COEF_INDEX, reg); + reg = RT286_PROC_COEF; + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + if (reg == rt286->index_cache[i].reg) { + rt286->index_cache[i].def = value; + break; + } + + } + } + + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + /* + * 4 bit VID: reg should be 0 + * 12 bit VID: value should be 0 + * So we use an OR operator to handle it rather than use if condition. + */ + data[2] = ((reg >> 8) & 0xff) | ((value >> 8) & 0xff); + data[3] = value & 0xff; + + ret = i2c_master_send(client, data, 4); + + if (ret == 4) + return 0; + else + pr_err("ret=%d\n", ret); + if (ret < 0) + return ret; + else + return -EIO; +} + +static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + __be32 be_reg; + unsigned int index, vid, buf = 0x0; + + /*handle index registers*/ + if (reg <= 0xff) { + rt286_hw_write(client, RT286_COEF_INDEX, reg); + reg = RT286_PROC_COEF; + } + + reg = reg | 0x80000; + vid = (reg >> 8) & 0xfff; + + if (AC_VERB_GET_AMP_GAIN_MUTE == (vid & 0xf00)) { + index = (reg >> 8) & 0xf; + reg = (reg & ~0xf0f) | index; + } + be_reg = cpu_to_be32(reg); + + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)&be_reg; + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} + +static void rt286_index_sync(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 0; i < INDEX_CACHE_SIZE; i++) { + snd_soc_write(codec, rt286->index_cache[i].reg, + rt286->index_cache[i].def); + } +} + +static int rt286_support_power_controls[] = { + RT286_DAC_OUT1, + RT286_DAC_OUT2, + RT286_ADC_IN1, + RT286_ADC_IN2, + RT286_MIC1, + RT286_DMIC1, + RT286_DMIC2, + RT286_SPK_OUT, + RT286_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val, buf; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + *hp = buf & 0x80000000; + if (*hp) { + /* power on HV,VERF */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL1, 0x1001, 0x0); + /* power LDO1 */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL2, 0x4, 0x4); + snd_soc_write(codec, RT286_SET_MIC1, 0x24); + val = snd_soc_read(codec, RT286_CBJ_CTRL2); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + val = snd_soc_read(codec, + RT286_CBJ_CTRL2); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + snd_soc_write(codec, + RT286_SET_MIC1, 0x20); + /* power off HV,VERF */ + snd_soc_update_bits(codec, + RT286_POWER_CTRL1, 0x1001, 0x1001); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x0000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0030, 0x0000); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, 0x0030, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, 0xc000, 0x8000); + } else { + *mic = false; + } + + snd_soc_update_bits(codec, + RT286_MISC_CTRL1, + 0x0060, 0x0000); + } else { + snd_soc_update_bits(codec, + RT286_MISC_CTRL1, + 0x0060, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL3, + 0xc000, 0x8000); + snd_soc_update_bits(codec, + RT286_CBJ_CTRL1, + 0x0030, 0x0020); + snd_soc_update_bits(codec, + RT286_A_BIAS_CTRL2, + 0xc000, 0x8000); + + *mic = false; + } + } else { + buf = snd_soc_read(codec, RT286_GET_HP_SENSE); + *hp = buf & 0x80000000; + buf = snd_soc_read(codec, RT286_GET_MIC1_SENSE); + *mic = buf & 0x80000000; + } + + return 0; +} + +static void rt286_jack_detect_work(struct work_struct *work) +{ + struct rt286_priv *rt286 = + container_of(work, struct rt286_priv, jack_detect_work.work); + int status = 0; + bool hp = false; + bool mic = false; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); +} + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + rt286->jack = jack; + + /* Send an initial empty report */ + snd_soc_jack_report(rt286->jack, 0, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + return 0; +} +EXPORT_SYMBOL_GPL(rt286_mic_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC0 Playback Volume", RT286_DACL_GAIN, + RT286_DACR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_DOUBLE_R_TLV("ADC0 Capture Volume", RT286_ADCL_GAIN, + RT286_ADCR_GAIN, 0, 0x7f, 0, out_vol_tlv), + SOC_SINGLE_TLV("AMIC Volume", RT286_MIC_GAIN, + 0, 0x3, 0, mic_vol_tlv), + SOC_DOUBLE_R("Speaker Playback Switch", RT286_SPOL_GAIN, + RT286_SPOR_GAIN, RT286_MUTE_SFT, 1, 1), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", RT286_F_DAC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", RT286_F_RECMIX_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Mic1 Switch", RT286_REC_MIC_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", RT286_REC_I2S_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", RT286_REC_LINE_SWITCH, + RT286_MUTE_SFT, 1, 1), + SOC_DAPM_SINGLE("Beep Switch", RT286_REC_BEEP_SWITCH, + RT286_MUTE_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new spo_enable_control = + SOC_DAPM_SINGLE("Switch", RT286_SET_PIN_SPK, + RT286_SET_PIN_SFT, 1, 0); + +static const struct snd_kcontrol_new hpol_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOL_GAIN, + RT286_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpor_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT286_HPOR_GAIN, + RT286_MUTE_SFT, 1, 1); + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "RECMIX", "Dmic" +}; + +static const int rt286_adc_values[] = { + 0, 4, 5, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, RT286_ADC0_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_ENUM("ADC 0 source", rt286_adc0_enum); + +static SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, RT286_ADC1_MUX, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_ENUM("ADC 1 source", rt286_adc1_enum); + +static const char * const rt286_dac_src[] = { + "Front", "Surround" +}; +/* HP-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, RT286_HPO_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static SOC_ENUM_SINGLE_DECL(rt286_spo_enum, RT286_SPK_MUX, + 0, rt286_dac_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +static int rt286_spk_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: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_HIGH); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, + RT286_SPK_EAPD, RT286_SET_EAPD_LOW); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_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: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0x20); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT286_SET_PIN_DMIC1, 0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int nid; + + nid = (w->reg >> 20) & 0xff; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, nid, 0), + 0x7080, 0x7080); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", RT286_SET_POWER(RT286_DMIC1), 0, 1, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA("DMIC2", RT286_SET_POWER(RT286_DMIC2), 0, 1, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", RT286_SET_POWER(RT286_ADC_IN1), 0, 1, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", RT286_SET_POWER(RT286_ADC_IN2), 0, 1, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* Audio Interface */ + 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), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DACs */ + SND_SOC_DAPM_DAC("DAC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* Output Mux */ + SND_SOC_DAPM_MUX("SPK Mux", SND_SOC_NOPM, 0, 0, &rt286_spo_mux), + SND_SOC_DAPM_MUX("HPO Mux", SND_SOC_NOPM, 0, 0, &rt286_hpo_mux), + + SND_SOC_DAPM_SUPPLY("HP Power", RT286_SET_PIN_HPO, + RT286_SET_PIN_SFT, 0, NULL, 0), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", RT286_SET_POWER(RT286_DAC_OUT1), 0, 1, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", RT286_SET_POWER(RT286_DAC_OUT2), 0, 1, + NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_SWITCH_E("SPO", SND_SOC_NOPM, 0, 0, + &spo_enable_control, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("HPO L", SND_SOC_NOPM, 0, 0, + &hpol_enable_control), + SND_SOC_DAPM_SWITCH("HPO R", SND_SOC_NOPM, 0, 0, + &hpor_enable_control), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "RECMIX", "RECMIX"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "RECMIX", "RECMIX"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", "Switch", "SPK Mux"}, + {"HPO L", "Switch", "HPO Mux"}, + {"HPO R", "Switch", "HPO Mux"}, + {"HPO L", NULL, "HP Power"}, + {"HPO R", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO L"}, + {"HPO Pin", NULL, "HPO R"}, +}; + +static int rt286_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; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_width(params)) { + /* bit 6:4 Bits per Sample */ + case 16: + d_len_code = 0; + val |= (0x1 << 4); + break; + case 32: + d_len_code = 2; + val |= (0x4 << 4); + break; + case 20: + d_len_code = 1; + val |= (0x2 << 4); + break; + case 24: + d_len_code = 2; + val |= (0x3 << 4); + break; + case 8: + d_len_code = 3; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x0018, d_len_code << 3); + dev_dbg(codec->dev, "format val = 0x%x\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x407f, val); + else + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + snd_soc_update_bits(codec, RT286_DAC_FORMAT, 0x8000, 0); + snd_soc_update_bits(codec, RT286_ADC_FORMAT, 0x8000, 0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x0100, 0x0); + snd_soc_update_bits(codec, + RT286_PLL_CTRL1, 0x20, 0x20); + } else { + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x0100, 0x0100); + snd_soc_update_bits(codec, + RT286_PLL_CTRL, 0x4, 0x4); + snd_soc_update_bits(codec, + RT286_PLL_CTRL1, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x40, 0x40); + break; + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x40, 0x0); + break; + case 12288000: + case 11289600: + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x8, 0x0); + snd_soc_update_bits(codec, + RT286_CLK_DIV, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + snd_soc_update_bits(codec, + RT286_I2S_CTRL2, 0x8, 0x8); + snd_soc_update_bits(codec, + RT286_CLK_DIV, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (50 == ratio) + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x1000, 0x1000); + else + snd_soc_update_bits(codec, + RT286_I2S_CTRL1, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D0); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x200); + } + break; + + case SND_SOC_BIAS_ON: + mdelay(10); + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, + RT286_SET_AUDIO_POWER, AC_PWRST_D3); + snd_soc_update_bits(codec, + RT286_DC_GAIN, 0x200, 0x0); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static irqreturn_t rt286_irq(int irq, void *data) +{ + struct rt286_priv *rt286 = data; + bool hp = false; + bool mic = false; + int status = 0; + + rt286_jack_detect(rt286->codec, &hp, &mic); + + /* Clear IRQ */ + snd_soc_update_bits(rt286->codec, + RT286_IRQ_CTRL, 0x1, 0x1); + + if (hp == true) + status |= SND_JACK_HEADPHONE; + + if (mic == true) + status |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(rt286->jack, status, + SND_JACK_MICROPHONE | SND_JACK_HEADPHONE); + + pm_wakeup_event(&rt286->i2c->dev, 300); + + return IRQ_HANDLED; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + rt286->codec = codec; + + return 0; +} + +static int rt286_remove(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + cancel_delayed_work_sync(&rt286->jack_detect_work); + + return 0; +} + +#ifdef CONFIG_PM +static int rt286_suspend(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt286->regmap, true); + regcache_mark_dirty(rt286->regmap); + + return 0; +} + +static int rt286_resume(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt286->regmap, false); + rt286_index_sync(codec); + regcache_sync(rt286->regmap); + + return 0; +} +#else +#define rt286_suspend NULL +#define rt286_resume NULL +#endif + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_bclk_ratio = rt286_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "rt286-aif2", + .id = RT286_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + .symmetric_rates = 1, + }, + +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .remove = rt286_remove, + .suspend = rt286_suspend, + .resume = rt286_resume, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 32, + .val_bits = 32, + .max_register = 0x02370100, + .volatile_reg = rt286_volatile_register, + .readable_reg = rt286_readable_register, + .reg_write = rt286_hw_write, + .reg_read = rt286_hw_read, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt286_reg, + .num_reg_defaults = ARRAY_SIZE(rt286_reg), +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt286", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static const struct acpi_device_id rt286_acpi_match[] = { + { "INT343A", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + int i, ret; + + rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt286->regmap, + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &ret); + if (ret != RT286_VENDOR_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt286\n", ret); + return -ENODEV; + } + + rt286->index_cache = rt286_index_def; + rt286->i2c = i2c; + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); + + for (i = 0; i < RT286_POWER_REG_LEN; i++) + regmap_write(rt286->regmap, + RT286_SET_POWER(rt286_support_power_controls[i]), + AC_PWRST_D1); + + if (!rt286->pdata.cbj_en) { + regmap_write(rt286->regmap, RT286_CBJ_CTRL2, 0x0000); + regmap_write(rt286->regmap, RT286_MIC1_DET_CTRL, 0x0816); + regmap_write(rt286->regmap, RT286_MISC_CTRL1, 0x0000); + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xf000, 0xb000); + } else { + regmap_update_bits(rt286->regmap, + RT286_CBJ_CTRL1, 0xf000, 0x5000); + } + + mdelay(10); + + if (!rt286->pdata.gpio2_en) + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0x4000); + else + regmap_write(rt286->regmap, RT286_SET_DMIC2_DEFAULT, 0); + + mdelay(10); + + /*Power down LDO2*/ + regmap_update_bits(rt286->regmap, RT286_POWER_CTRL2, 0x8, 0x0); + + /*Set depop parameter*/ + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL2, 0x403a, 0x401a); + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); + regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); + + if (rt286->i2c->irq) { + regmap_update_bits(rt286->regmap, + RT286_IRQ_CTRL, 0x2, 0x2); + + INIT_DELAYED_WORK(&rt286->jack_detect_work, + rt286_jack_detect_work); + schedule_delayed_work(&rt286->jack_detect_work, + msecs_to_jiffies(1250)); + + ret = request_threaded_irq(rt286->i2c->irq, NULL, rt286_irq, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT, "rt286", rt286); + if (ret != 0) { + dev_err(&i2c->dev, + "Failed to reguest IRQ: %d\n", ret); + return ret; + } + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + + return ret; +} + +static int rt286_i2c_remove(struct i2c_client *i2c) +{ + struct rt286_priv *rt286 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt286); + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + + +static struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt286", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .remove = rt286_i2c_remove, + .id_table = rt286_i2c_id, +}; + +module_i2c_driver(rt286_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao@realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 000000000000..b539b7320a79 --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,198 @@ +/* + * rt286.h -- RT286 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu@realtek.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 __RT286_H__ +#define __RT286_H__ + +#define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) + +#define RT286_AUDIO_FUNCTION_GROUP 0x01 +#define RT286_DAC_OUT1 0x02 +#define RT286_DAC_OUT2 0x03 +#define RT286_ADC_IN1 0x09 +#define RT286_ADC_IN2 0x08 +#define RT286_MIXER_IN 0x0b +#define RT286_MIXER_OUT1 0x0c +#define RT286_MIXER_OUT2 0x0d +#define RT286_DMIC1 0x12 +#define RT286_DMIC2 0x13 +#define RT286_SPK_OUT 0x14 +#define RT286_MIC1 0x18 +#define RT286_LINE1 0x1a +#define RT286_BEEP 0x1d +#define RT286_SPDIF 0x1e +#define RT286_VENDOR_REGISTERS 0x20 +#define RT286_HP_OUT 0x21 +#define RT286_MIXER_IN1 0x22 +#define RT286_MIXER_IN2 0x23 + +#define RT286_SET_PIN_SFT 6 +#define RT286_SET_PIN_ENABLE 0x40 +#define RT286_SET_PIN_DISABLE 0 +#define RT286_SET_EAPD_HIGH 0x2 +#define RT286_SET_EAPD_LOW 0 + +#define RT286_MUTE_SFT 7 + +/* Verb commands */ +#define RT286_GET_PARAM(NID, PARAM) VERB_CMD(AC_VERB_PARAMETERS, NID, PARAM) +#define RT286_SET_POWER(NID) VERB_CMD(AC_VERB_SET_POWER_STATE, NID, 0) +#define RT286_SET_AUDIO_POWER RT286_SET_POWER(RT286_AUDIO_FUNCTION_GROUP) +#define RT286_SET_HPO_POWER RT286_SET_POWER(RT286_HP_OUT) +#define RT286_SET_SPK_POWER RT286_SET_POWER(RT286_SPK_OUT) +#define RT286_SET_DMIC1_POWER RT286_SET_POWER(RT286_DMIC1) +#define RT286_SPK_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_SPK_OUT, 0) +#define RT286_HPO_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_HP_OUT, 0) +#define RT286_ADC0_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN1, 0) +#define RT286_ADC1_MUX\ + VERB_CMD(AC_VERB_SET_CONNECT_SEL, RT286_MIXER_IN2, 0) +#define RT286_SET_MIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_MIC1, 0) +#define RT286_SET_PIN_HPO\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_HP_OUT, 0) +#define RT286_SET_PIN_SPK\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_SPK_OUT, 0) +#define RT286_SET_PIN_DMIC1\ + VERB_CMD(AC_VERB_SET_PIN_WIDGET_CONTROL, RT286_DMIC1, 0) +#define RT286_SPK_EAPD\ + VERB_CMD(AC_VERB_SET_EAPD_BTLENABLE, RT286_SPK_OUT, 0) +#define RT286_SET_AMP_GAIN_HPO\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0) +#define RT286_SET_AMP_GAIN_ADC_IN1\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0) +#define RT286_SET_AMP_GAIN_ADC_IN2\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN2, 0) +#define RT286_GET_HP_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_HP_OUT, 0) +#define RT286_GET_MIC1_SENSE\ + VERB_CMD(AC_VERB_GET_PIN_SENSE, RT286_MIC1, 0) +#define RT286_SET_DMIC2_DEFAULT\ + VERB_CMD(AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, RT286_DMIC2, 0) +#define RT286_DACL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0xa000) +#define RT286_DACR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_DAC_OUT1, 0x9000) +#define RT286_ADCL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x6000) +#define RT286_ADCR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_ADC_IN1, 0x5000) +#define RT286_MIC_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIC1, 0x7000) +#define RT286_SPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0xa000) +#define RT286_SPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_SPK_OUT, 0x9000) +#define RT286_HPOL_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0xa000) +#define RT286_HPOR_GAIN\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_HP_OUT, 0x9000) +#define RT286_F_DAC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7000) +#define RT286_F_RECMIX_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_OUT1, 0x7100) +#define RT286_REC_MIC_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7000) +#define RT286_REC_I2S_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7100) +#define RT286_REC_LINE_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7200) +#define RT286_REC_BEEP_SWITCH\ + VERB_CMD(AC_VERB_SET_AMP_GAIN_MUTE, RT286_MIXER_IN, 0x7300) +#define RT286_DAC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_DAC_OUT1, 0) +#define RT286_ADC_FORMAT\ + VERB_CMD(AC_VERB_SET_STREAM_FORMAT, RT286_ADC_IN1, 0) +#define RT286_COEF_INDEX\ + VERB_CMD(AC_VERB_SET_COEF_INDEX, RT286_VENDOR_REGISTERS, 0) +#define RT286_PROC_COEF\ + VERB_CMD(AC_VERB_SET_PROC_COEF, RT286_VENDOR_REGISTERS, 0) + +/* Index registers */ +#define RT286_A_BIAS_CTRL1 0x01 +#define RT286_A_BIAS_CTRL2 0x02 +#define RT286_POWER_CTRL1 0x03 +#define RT286_A_BIAS_CTRL3 0x04 +#define RT286_POWER_CTRL2 0x08 +#define RT286_I2S_CTRL1 0x09 +#define RT286_I2S_CTRL2 0x0a +#define RT286_CLK_DIV 0x0b +#define RT286_DC_GAIN 0x0d +#define RT286_POWER_CTRL3 0x0f +#define RT286_MIC1_DET_CTRL 0x19 +#define RT286_MISC_CTRL1 0x20 +#define RT286_IRQ_CTRL 0x33 +#define RT286_PLL_CTRL1 0x49 +#define RT286_CBJ_CTRL1 0x4f +#define RT286_CBJ_CTRL2 0x50 +#define RT286_PLL_CTRL 0x63 +#define RT286_DEPOP_CTRL1 0x66 +#define RT286_DEPOP_CTRL2 0x67 +#define RT286_DEPOP_CTRL3 0x68 +#define RT286_DEPOP_CTRL4 0x69 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_M_SPK_MUX_SFT 14 +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_M_HP_MUX_SFT 14 +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIF2, + RT286_AIFS, +}; + +int rt286_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); + +#endif /* __RT286_H__ */ + diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index c30fedb3e149..f5b4a9c79cdf 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -58,3 +58,15 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH help This adds audio driver for Intel Baytrail platform based boards with the MAX98090 audio codec. + +config SND_SOC_INTEL_BROADWELL_MACH + tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC + select SND_SOC_INTEL_HASWELL + select SND_COMPRESS_OFFLOAD + select SND_SOC_RT286 + help + This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell + Ultrabook platforms. + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 4bfca79a42ba..7acbfc43a0c6 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -24,7 +24,9 @@ obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-soc-sst-broadwell-objs := broadwell.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o +obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o diff --git a/sound/soc/intel/broadwell.c b/sound/soc/intel/broadwell.c new file mode 100644 index 000000000000..0e550f14028f --- /dev/null +++ b/sound/soc/intel/broadwell.c @@ -0,0 +1,251 @@ +/* + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include "sst-dsp.h" +#include "sst-haswell-ipc.h" + +#include "../codecs/rt286.h" + +static const struct snd_soc_dapm_widget broadwell_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC1", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_LINE("Line Jack", NULL), +}; + +static const struct snd_soc_dapm_route broadwell_rt286_map[] = { + + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphones", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + {"LINE1", NULL, "Line Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC1"}, + {"DMIC2 Pin", NULL, "DMIC2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static int broadwell_ssp0_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 ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 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; +} + +static int broadwell_rt286_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; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops broadwell_rt286_ops = { + .hw_params = broadwell_rt286_hw_params, +}; + +static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); + struct sst_hsw *broadwell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings */ + ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_CLOCK_MASTER, 9); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set device config\n"); + return ret; + } + + /* always connected - check HP for jack detect */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin(dapm, "Line Jack"); + snd_soc_dapm_enable_pin(dapm, "DMIC1"); + snd_soc_dapm_enable_pin(dapm, "DMIC2"); + + return 0; +} + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link broadwell_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback", + .cpu_dai_name = "System Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = broadwell_rtd_init, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload0", + .stream_name = "Offload0 Playback", + .cpu_dai_name = "Offload0 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload1", + .stream_name = "Offload1 Playback", + .cpu_dai_name = "Offload1 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Loopback PCM", + .stream_name = "Loopback", + .cpu_dai_name = "Loopback Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 0, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + { + .name = "Capture PCM", + .stream_name = "Capture", + .cpu_dai_name = "Capture Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .be_id = 0, + .cpu_dai_name = "snd-soc-dummy-dai", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt286-aif1", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broadwell_ssp0_fixup, + .ops = &broadwell_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +/* broadwell audio machine driver for WPT + RT286S */ +static struct snd_soc_card broadwell_rt286 = { + .name = "broadwell-rt286", + .owner = THIS_MODULE, + .dai_link = broadwell_rt286_dais, + .num_links = ARRAY_SIZE(broadwell_rt286_dais), + .dapm_widgets = broadwell_widgets, + .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets), + .dapm_routes = broadwell_rt286_map, + .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map), + .fully_routed = true, +}; + +static int broadwell_audio_probe(struct platform_device *pdev) +{ + broadwell_rt286.dev = &pdev->dev; + + return snd_soc_register_card(&broadwell_rt286); +} + +static int broadwell_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&broadwell_rt286); + return 0; +} + +static struct platform_driver broadwell_audio = { + .probe = broadwell_audio_probe, + .remove = broadwell_audio_remove, + .driver = { + .name = "broadwell-audio", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(broadwell_audio) + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:broadwell-audio"); diff --git a/sound/soc/intel/byt-max98090.c b/sound/soc/intel/byt-max98090.c index 5cfb41ec3fab..b8b8af571ef1 100644 --- a/sound/soc/intel/byt-max98090.c +++ b/sound/soc/intel/byt-max98090.c @@ -63,14 +63,6 @@ static struct snd_soc_jack_pin hs_jack_pins[] = { .pin = "Headset Mic", .mask = SND_JACK_MICROPHONE, }, - { - .pin = "Ext Spk", - .mask = SND_JACK_LINEOUT, - }, - { - .pin = "Int Mic", - .mask = SND_JACK_LINEIN, - }, }; static struct snd_soc_jack_gpio hs_jack_gpios[] = { diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index 53d160d39972..234a58de3c53 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -34,6 +34,7 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { }; static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { + {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"IN2N", NULL, "Headset Mic"}, {"DMIC1", NULL, "Internal Mic"}, diff --git a/sound/soc/intel/sst-atom-controls.h b/sound/soc/intel/sst-atom-controls.h new file mode 100644 index 000000000000..14063ab8c7c5 --- /dev/null +++ b/sound/soc/intel/sst-atom-controls.h @@ -0,0 +1,30 @@ +/* + * Copyright (C) 2013-14 Intel Corp + * Author: Ramesh Babu <ramesh.babu.koul@intel.com> + * Omair M Abdullah <omair.m.abdullah@intel.com> + * Samreen Nilofer <samreen.nilofer@intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + */ + +#ifndef __SST_CONTROLS_V2_H__ +#define __SST_CONTROLS_V2_H__ + +enum { + MERR_DPCM_AUDIO = 0, + MERR_DPCM_COMPR, +}; + + +#endif diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index d207b22ea330..67673a2c0f41 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -122,6 +122,26 @@ struct sst_byt_tstamp { u32 channel_peak[8]; } __packed; +struct sst_byt_fw_version { + u8 build; + u8 minor; + u8 major; + u8 type; +} __packed; + +struct sst_byt_fw_build_info { + u8 date[16]; + u8 time[16]; +} __packed; + +struct sst_byt_fw_init { + struct sst_byt_fw_version fw_version; + struct sst_byt_fw_build_info build_info; + u16 result; + u8 module_id; + u8 debug_info; +} __packed; + /* driver internal IPC message structure */ struct ipc_message { struct list_head list; @@ -868,6 +888,7 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) { struct sst_byt *byt; struct sst_fw *byt_sst_fw; + struct sst_byt_fw_init init; int err; dev_dbg(dev, "initialising Byt DSP IPC\n"); @@ -929,6 +950,15 @@ int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) goto boot_err; } + /* show firmware information */ + sst_dsp_inbox_read(byt->dsp, &init, sizeof(init)); + dev_info(byt->dev, "FW version: %02x.%02x.%02x.%02x\n", + init.fw_version.major, init.fw_version.minor, + init.fw_version.build, init.fw_version.type); + dev_info(byt->dev, "Build type: %x\n", init.fw_version.type); + dev_info(byt->dev, "Build date: %s %s\n", + init.build_info.date, init.build_info.time); + pdata->dsp = byt; byt->fw = byt_sst_fw; diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c index 0b715b20a2d7..cd23060a0d86 100644 --- a/sound/soc/intel/sst-dsp.c +++ b/sound/soc/intel/sst-dsp.c @@ -224,19 +224,23 @@ EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64); void sst_dsp_dump(struct sst_dsp *sst) { - sst->ops->dump(sst); + if (sst->ops->dump) + sst->ops->dump(sst); } EXPORT_SYMBOL_GPL(sst_dsp_dump); void sst_dsp_reset(struct sst_dsp *sst) { - sst->ops->reset(sst); + if (sst->ops->reset) + sst->ops->reset(sst); } EXPORT_SYMBOL_GPL(sst_dsp_reset); int sst_dsp_boot(struct sst_dsp *sst) { - sst->ops->boot(sst); + if (sst->ops->boot) + sst->ops->boot(sst); + return 0; } EXPORT_SYMBOL_GPL(sst_dsp_boot); diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h index e44423be66c4..3165dfa97408 100644 --- a/sound/soc/intel/sst-dsp.h +++ b/sound/soc/intel/sst-dsp.h @@ -52,7 +52,11 @@ #define SST_CLKCTL 0x78 #define SST_CSR2 0x80 #define SST_LTRC 0xE0 -#define SST_HDMC 0xE8 +#define SST_HMDC 0xE8 + +#define SST_SHIM_BEGIN SST_CSR +#define SST_SHIM_END SST_HDMC + #define SST_DBGO 0xF0 #define SST_SHIM_SIZE 0x100 @@ -73,6 +77,8 @@ #define SST_CSR_S0IOCS (0x1 << 21) #define SST_CSR_S1IOCS (0x1 << 23) #define SST_CSR_LPCS (0x1 << 31) +#define SST_CSR_24MHZ_LPCS (SST_CSR_SBCS0 | SST_CSR_SBCS1 | SST_CSR_LPCS) +#define SST_CSR_24MHZ_NO_LPCS (SST_CSR_SBCS0 | SST_CSR_SBCS1) #define SST_BYT_CSR_RST (0x1 << 0) #define SST_BYT_CSR_VECTOR_SEL (0x1 << 1) #define SST_BYT_CSR_STALL (0x1 << 2) @@ -92,6 +98,14 @@ #define SST_IMRX_DONE (0x1 << 0) #define SST_BYT_IMRX_REQUEST (0x1 << 1) +/* IMRD / IMD */ +#define SST_IMRD_DONE (0x1 << 0) +#define SST_IMRD_BUSY (0x1 << 1) +#define SST_IMRD_SSP0 (0x1 << 16) +#define SST_IMRD_DMAC0 (0x1 << 21) +#define SST_IMRD_DMAC1 (0x1 << 22) +#define SST_IMRD_DMAC (SST_IMRD_DMAC0 | SST_IMRD_DMAC1) + /* IPCX / IPCC */ #define SST_IPCX_DONE (0x1 << 30) #define SST_IPCX_BUSY (0x1 << 31) @@ -118,9 +132,21 @@ /* LTRC */ #define SST_LTRC_VAL(x) (x << 0) -/* HDMC */ -#define SST_HDMC_HDDA0(x) (x << 0) -#define SST_HDMC_HDDA1(x) (x << 7) +/* HMDC */ +#define SST_HMDC_HDDA0(x) (x << 0) +#define SST_HMDC_HDDA1(x) (x << 7) +#define SST_HMDC_HDDA_E0_CH0 1 +#define SST_HMDC_HDDA_E0_CH1 2 +#define SST_HMDC_HDDA_E0_CH2 4 +#define SST_HMDC_HDDA_E0_CH3 8 +#define SST_HMDC_HDDA_E1_CH0 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH0) +#define SST_HMDC_HDDA_E1_CH1 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH1) +#define SST_HMDC_HDDA_E1_CH2 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH2) +#define SST_HMDC_HDDA_E1_CH3 SST_HMDC_HDDA1(SST_HMDC_HDDA_E0_CH3) +#define SST_HMDC_HDDA_E0_ALLCH (SST_HMDC_HDDA_E0_CH0 | SST_HMDC_HDDA_E0_CH1 | \ + SST_HMDC_HDDA_E0_CH2 | SST_HMDC_HDDA_E0_CH3) +#define SST_HMDC_HDDA_E1_ALLCH (SST_HMDC_HDDA_E1_CH0 | SST_HMDC_HDDA_E1_CH1 | \ + SST_HMDC_HDDA_E1_CH2 | SST_HMDC_HDDA_E1_CH3) /* SST Vendor Defined Registers and bits */ @@ -130,11 +156,16 @@ #define SST_VDRTCTL3 0xaC /* VDRTCTL0 */ +#define SST_VDRTCL0_APLLSE_MASK 1 #define SST_VDRTCL0_DSRAMPGE_SHIFT 16 #define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT) #define SST_VDRTCL0_ISRAMPGE_SHIFT 6 #define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT) +/* PMCS */ +#define SST_PMCS 0x84 +#define SST_PMCS_PS_MASK 0x3 + struct sst_dsp; /* diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c index a33b931181dc..4b6c163c10ff 100644 --- a/sound/soc/intel/sst-haswell-dsp.c +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -28,9 +28,6 @@ #include <linux/firmware.h> #include <linux/pm_runtime.h> -#include <linux/acpi.h> -#include <acpi/acpi_bus.h> - #include "sst-dsp.h" #include "sst-dsp-priv.h" #include "sst-haswell-ipc.h" @@ -272,9 +269,9 @@ static void hsw_boot(struct sst_dsp *sst) SST_CSR2_SDFD_SSP1); /* enable DMA engine 0,1 all channels to access host memory */ - sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC, - SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff), - SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff)); + sst_dsp_shim_update_bits_unlocked(sst, SST_HMDC, + SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff), + SST_HMDC_HDDA1(0xff) | SST_HMDC_HDDA0(0xff)); /* disable all clock gating */ writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2); @@ -313,9 +310,7 @@ static const struct sst_adsp_memregion lp_region[] = { /* wild cat point ADSP mem regions */ static const struct sst_adsp_memregion wpt_region[] = { - {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */ - {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */ - {0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */ + {0x00000, 0xA0000, 20, SST_MEM_DRAM}, /* D-SRAM0,D-SRAM1,D-SRAM2 - 20 * 32kB */ {0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */ }; @@ -339,21 +334,40 @@ static int hsw_acpi_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata) return 0; } +struct sst_sram_shift { + u32 dev_id; /* SST Device IDs */ + u32 iram_shift; + u32 dram_shift; +}; + +static const struct sst_sram_shift sram_shift[] = { + {SST_DEV_ID_LYNX_POINT, 6, 16}, /* lp */ + {SST_DEV_ID_WILDCAT_POINT, 2, 12}, /* wpt */ +}; static u32 hsw_block_get_bit(struct sst_mem_block *block) { - u32 bit = 0, shift = 0; + u32 bit = 0, shift = 0, index; + struct sst_dsp *sst = block->dsp; - switch (block->type) { - case SST_MEM_DRAM: - shift = 16; - break; - case SST_MEM_IRAM: - shift = 6; - break; - default: - return 0; + for (index = 0; index < ARRAY_SIZE(sram_shift); index++) { + if (sram_shift[index].dev_id == sst->id) + break; } + if (index < ARRAY_SIZE(sram_shift)) { + switch (block->type) { + case SST_MEM_DRAM: + shift = sram_shift[index].dram_shift; + break; + case SST_MEM_IRAM: + shift = sram_shift[index].iram_shift; + break; + default: + shift = 0; + } + } else + shift = 0; + bit = 1 << (block->index + shift); return bit; @@ -501,8 +515,9 @@ static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) } } - /* set default power gating mask */ - writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0); + /* set default power gating control, enable power gating control for all blocks. that is, + can't be accessed, please enable each block before accessing. */ + writel(0xffffffff, sst->addr.pci_cfg + SST_VDRTCTL0); return 0; } diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 434236343ddf..b6291516dbbf 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -183,7 +183,7 @@ struct sst_hsw_ipc_fw_ready { u32 inbox_size; u32 outbox_size; u32 fw_info_size; - u8 fw_info[1]; + u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)]; } __attribute__((packed)); struct ipc_message { @@ -457,9 +457,10 @@ static void ipc_tx_msgs(struct kthread_work *work) return; } - /* if the DSP is busy we will TX messages after IRQ */ + /* if the DSP is busy, we will TX messages after IRQ. + * also postpone if we are in the middle of procesing completion irq*/ ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX); - if (ipcx & SST_IPCX_BUSY) { + if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) { spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); return; } @@ -502,6 +503,7 @@ static int tx_wait_done(struct sst_hsw *hsw, struct ipc_message *msg, ipc_shim_dbg(hsw, "message timeout"); trace_ipc_error("error message timeout for", msg->header); + list_del(&msg->list); ret = -ETIMEDOUT; } else { @@ -569,6 +571,9 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) { struct sst_hsw_ipc_fw_ready fw_ready; u32 offset; + u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)]; + char *tmp[5], *pinfo; + int i = 0; offset = (header & 0x1FFFFFFF) << 3; @@ -589,6 +594,19 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) fw_ready.inbox_offset, fw_ready.inbox_size); dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n", fw_ready.outbox_offset, fw_ready.outbox_size); + if (fw_ready.fw_info_size < sizeof(fw_ready.fw_info)) { + fw_ready.fw_info[fw_ready.fw_info_size] = 0; + dev_dbg(hsw->dev, " Firmware info: %s \n", fw_ready.fw_info); + + /* log the FW version info got from the mailbox here. */ + memcpy(fw_info, fw_ready.fw_info, fw_ready.fw_info_size); + pinfo = &fw_info[0]; + for (i = 0; i < sizeof(tmp) / sizeof(char *); i++) + tmp[i] = strsep(&pinfo, " "); + dev_info(hsw->dev, "FW loaded, mailbox readback FW info: type %s, - " + "version: %s.%s, build %s, source commit id: %s\n", + tmp[0], tmp[1], tmp[2], tmp[3], tmp[4]); + } } static void hsw_notification_work(struct work_struct *work) @@ -671,7 +689,9 @@ static void hsw_stream_update(struct sst_hsw *hsw, struct ipc_message *msg) switch (stream_msg) { case IPC_STR_STAGE_MESSAGE: case IPC_STR_NOTIFICATION: + break; case IPC_STR_RESET: + trace_ipc_notification("stream reset", stream->reply.stream_hw_id); break; case IPC_STR_PAUSE: stream->running = false; @@ -762,7 +782,8 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header) } /* update any stream states */ - hsw_stream_update(hsw, msg); + if (msg_get_global_type(header) == IPC_GLB_STREAM_MESSAGE) + hsw_stream_update(hsw, msg); /* wake up and return the error if we have waiters on this message ? */ list_del(&msg->list); @@ -1628,7 +1649,7 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw, enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx) { u32 header, state_; - int ret; + int ret, item; header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE); state_ = state; @@ -1642,6 +1663,13 @@ int sst_hsw_dx_set_state(struct sst_hsw *hsw, return ret; } + for (item = 0; item < dx->entries_no; item++) { + dev_dbg(hsw->dev, + "Item[%d] offset[%x] - size[%x] - source[%x]\n", + item, dx->mem_info[item].offset, + dx->mem_info[item].size, + dx->mem_info[item].source); + } dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n", dx->entries_no, state); @@ -1775,8 +1803,6 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) /* get the FW version */ sst_hsw_fw_get_version(hsw, &version); - dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n", - version.type, version.major, version.minor, version.build); /* get the globalmixer */ ret = sst_hsw_mixer_get_info(hsw); diff --git a/sound/soc/intel/sst-mfld-dsp.h b/sound/soc/intel/sst-mfld-dsp.h index 8d482d76475a..4257263157cd 100644 --- a/sound/soc/intel/sst-mfld-dsp.h +++ b/sound/soc/intel/sst-mfld-dsp.h @@ -3,7 +3,7 @@ /* * sst_mfld_dsp.h - Intel SST Driver for audio engine * - * Copyright (C) 2008-12 Intel Corporation + * Copyright (C) 2008-14 Intel Corporation * Authors: Vinod Koul <vinod.koul@linux.intel.com> * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ * @@ -19,6 +19,142 @@ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#define SST_MAX_BIN_BYTES 1024 + +#define MAX_DBG_RW_BYTES 80 +#define MAX_NUM_SCATTER_BUFFERS 8 +#define MAX_LOOP_BACK_DWORDS 8 +/* IPC base address and mailbox, timestamp offsets */ +#define SST_MAILBOX_SIZE 0x0400 +#define SST_MAILBOX_SEND 0x0000 +#define SST_TIME_STAMP 0x1800 +#define SST_TIME_STAMP_MRFLD 0x800 +#define SST_RESERVED_OFFSET 0x1A00 +#define SST_SCU_LPE_MAILBOX 0x1000 +#define SST_LPE_SCU_MAILBOX 0x1400 +#define SST_SCU_LPE_LOG_BUF (SST_SCU_LPE_MAILBOX+16) +#define PROCESS_MSG 0x80 + +/* Message ID's for IPC messages */ +/* Bits B7: SST or IA/SC ; B6-B4: Msg Category; B3-B0: Msg Type */ + +/* I2L Firmware/Codec Download msgs */ +#define IPC_IA_PREP_LIB_DNLD 0x01 +#define IPC_IA_LIB_DNLD_CMPLT 0x02 +#define IPC_IA_GET_FW_VERSION 0x04 +#define IPC_IA_GET_FW_BUILD_INF 0x05 +#define IPC_IA_GET_FW_INFO 0x06 +#define IPC_IA_GET_FW_CTXT 0x07 +#define IPC_IA_SET_FW_CTXT 0x08 +#define IPC_IA_PREPARE_SHUTDOWN 0x31 +/* I2L Codec Config/control msgs */ +#define IPC_PREP_D3 0x10 +#define IPC_IA_SET_CODEC_PARAMS 0x10 +#define IPC_IA_GET_CODEC_PARAMS 0x11 +#define IPC_IA_SET_PPP_PARAMS 0x12 +#define IPC_IA_GET_PPP_PARAMS 0x13 +#define IPC_SST_PERIOD_ELAPSED_MRFLD 0xA +#define IPC_IA_ALG_PARAMS 0x1A +#define IPC_IA_TUNING_PARAMS 0x1B +#define IPC_IA_SET_RUNTIME_PARAMS 0x1C +#define IPC_IA_SET_PARAMS 0x1 +#define IPC_IA_GET_PARAMS 0x2 + +#define IPC_EFFECTS_CREATE 0xE +#define IPC_EFFECTS_DESTROY 0xF + +/* I2L Stream config/control msgs */ +#define IPC_IA_ALLOC_STREAM_MRFLD 0x2 +#define IPC_IA_ALLOC_STREAM 0x20 /* Allocate a stream ID */ +#define IPC_IA_FREE_STREAM_MRFLD 0x03 +#define IPC_IA_FREE_STREAM 0x21 /* Free the stream ID */ +#define IPC_IA_SET_STREAM_PARAMS 0x22 +#define IPC_IA_SET_STREAM_PARAMS_MRFLD 0x12 +#define IPC_IA_GET_STREAM_PARAMS 0x23 +#define IPC_IA_PAUSE_STREAM 0x24 +#define IPC_IA_PAUSE_STREAM_MRFLD 0x4 +#define IPC_IA_RESUME_STREAM 0x25 +#define IPC_IA_RESUME_STREAM_MRFLD 0x5 +#define IPC_IA_DROP_STREAM 0x26 +#define IPC_IA_DROP_STREAM_MRFLD 0x07 +#define IPC_IA_DRAIN_STREAM 0x27 /* Short msg with str_id */ +#define IPC_IA_DRAIN_STREAM_MRFLD 0x8 +#define IPC_IA_CONTROL_ROUTING 0x29 +#define IPC_IA_VTSV_UPDATE_MODULES 0x20 +#define IPC_IA_VTSV_DETECTED 0x21 + +#define IPC_IA_START_STREAM_MRFLD 0X06 +#define IPC_IA_START_STREAM 0x30 /* Short msg with str_id */ + +#define IPC_IA_SET_GAIN_MRFLD 0x21 +/* Debug msgs */ +#define IPC_IA_DBG_MEM_READ 0x40 +#define IPC_IA_DBG_MEM_WRITE 0x41 +#define IPC_IA_DBG_LOOP_BACK 0x42 +#define IPC_IA_DBG_LOG_ENABLE 0x45 +#define IPC_IA_DBG_SET_PROBE_PARAMS 0x47 + +/* L2I Firmware/Codec Download msgs */ +#define IPC_IA_FW_INIT_CMPLT 0x81 +#define IPC_IA_FW_INIT_CMPLT_MRFLD 0x01 +#define IPC_IA_FW_ASYNC_ERR_MRFLD 0x11 + +/* L2I Codec Config/control msgs */ +#define IPC_SST_FRAGMENT_ELPASED 0x90 /* Request IA more data */ + +#define IPC_SST_BUF_UNDER_RUN 0x92 /* PB Under run and stopped */ +#define IPC_SST_BUF_OVER_RUN 0x93 /* CAP Under run and stopped */ +#define IPC_SST_DRAIN_END 0x94 /* PB Drain complete and stopped */ +#define IPC_SST_CHNGE_SSP_PARAMS 0x95 /* PB SSP parameters changed */ +#define IPC_SST_STREAM_PROCESS_FATAL_ERR 0x96/* error in processing a stream */ +#define IPC_SST_PERIOD_ELAPSED 0x97 /* period elapsed */ + +#define IPC_SST_ERROR_EVENT 0x99 /* Buffer over run occurred */ +/* L2S messages */ +#define IPC_SC_DDR_LINK_UP 0xC0 +#define IPC_SC_DDR_LINK_DOWN 0xC1 +#define IPC_SC_SET_LPECLK_REQ 0xC2 +#define IPC_SC_SSP_BIT_BANG 0xC3 + +/* L2I Error reporting msgs */ +#define IPC_IA_MEM_ALLOC_FAIL 0xE0 +#define IPC_IA_PROC_ERR 0xE1 /* error in processing a + stream can be used by playback and + capture modules */ + +/* L2I Debug msgs */ +#define IPC_IA_PRINT_STRING 0xF0 + +/* Buffer under-run */ +#define IPC_IA_BUF_UNDER_RUN_MRFLD 0x0B + +/* Mrfld specific defines: + * For asynchronous messages(INIT_CMPLT, PERIOD_ELAPSED, ASYNC_ERROR) + * received from FW, the format is: + * - IPC High: pvt_id is set to zero. Always short message. + * - msg_id is in lower 16-bits of IPC low payload. + * - pipe_id is in higher 16-bits of IPC low payload for period_elapsed. + * - error id is in higher 16-bits of IPC low payload for async errors. + */ +#define SST_ASYNC_DRV_ID 0 + +/* Command Response or Acknowledge message to any IPC message will have + * same message ID and stream ID information which is sent. + * There is no specific Ack message ID. The data field is used as response + * meaning. + */ +enum ackData { + IPC_ACK_SUCCESS = 0, + IPC_ACK_FAILURE, +}; + +enum ipc_ia_msg_id { + IPC_CMD = 1, /*!< Task Control message ID */ + IPC_SET_PARAMS = 2,/*!< Task Set param message ID */ + IPC_GET_PARAMS = 3, /*!< Task Get param message ID */ + IPC_INVALID = 0xFF, /*!<Task Get param message ID */ +}; + enum sst_codec_types { /* AUDIO/MUSIC CODEC Type Definitions */ SST_CODEC_TYPE_UNKNOWN = 0, @@ -35,14 +171,157 @@ enum stream_type { SST_STREAM_TYPE_MUSIC = 1, }; +enum sst_error_codes { + /* Error code,response to msgId: Description */ + /* Common error codes */ + SST_SUCCESS = 0, /* Success */ + SST_ERR_INVALID_STREAM_ID = 1, + SST_ERR_INVALID_MSG_ID = 2, + SST_ERR_INVALID_STREAM_OP = 3, + SST_ERR_INVALID_PARAMS = 4, + SST_ERR_INVALID_CODEC = 5, + SST_ERR_INVALID_MEDIA_TYPE = 6, + SST_ERR_STREAM_ERR = 7, + + SST_ERR_STREAM_IN_USE = 15, +}; + +struct ipc_dsp_hdr { + u16 mod_index_id:8; /*!< DSP Command ID specific to tasks */ + u16 pipe_id:8; /*!< instance of the module in the pipeline */ + u16 mod_id; /*!< Pipe_id */ + u16 cmd_id; /*!< Module ID = lpe_algo_types_t */ + u16 length; /*!< Length of the payload only */ +} __packed; + +union ipc_header_high { + struct { + u32 msg_id:8; /* Message ID - Max 256 Message Types */ + u32 task_id:4; /* Task ID associated with this comand */ + u32 drv_id:4; /* Identifier for the driver to track*/ + u32 rsvd1:8; /* Reserved */ + u32 result:4; /* Reserved */ + u32 res_rqd:1; /* Response rqd */ + u32 large:1; /* Large Message if large = 1 */ + u32 done:1; /* bit 30 - Done bit */ + u32 busy:1; /* bit 31 - busy bit*/ + } part; + u32 full; +} __packed; +/* IPC header */ +union ipc_header_mrfld { + struct { + u32 header_low_payload; + union ipc_header_high header_high; + } p; + u64 full; +} __packed; +/* CAUTION NOTE: All IPC message body must be multiple of 32 bits.*/ + +/* IPC Header */ +union ipc_header { + struct { + u32 msg_id:8; /* Message ID - Max 256 Message Types */ + u32 str_id:5; + u32 large:1; /* Large Message if large = 1 */ + u32 reserved:2; /* Reserved for future use */ + u32 data:14; /* Ack/Info for msg, size of msg in Mailbox */ + u32 done:1; /* bit 30 */ + u32 busy:1; /* bit 31 */ + } part; + u32 full; +} __packed; + +/* Firmware build info */ +struct sst_fw_build_info { + unsigned char date[16]; /* Firmware build date */ + unsigned char time[16]; /* Firmware build time */ +} __packed; + +/* Firmware Version info */ +struct snd_sst_fw_version { + u8 build; /* build number*/ + u8 minor; /* minor number*/ + u8 major; /* major number*/ + u8 type; /* build type */ +}; + +struct ipc_header_fw_init { + struct snd_sst_fw_version fw_version;/* Firmware version details */ + struct sst_fw_build_info build_info; + u16 result; /* Fw init result */ + u8 module_id; /* Module ID in case of error */ + u8 debug_info; /* Debug info from Module ID in case of fail */ +} __packed; + +struct snd_sst_tstamp { + u64 ring_buffer_counter; /* PB/CP: Bytes copied from/to DDR. */ + u64 hardware_counter; /* PB/CP: Bytes DMAed to/from SSP. */ + u64 frames_decoded; + u64 bytes_decoded; + u64 bytes_copied; + u32 sampling_frequency; + u32 channel_peak[8]; +} __packed; + +/* Stream type params struture for Alloc stream */ +struct snd_sst_str_type { + u8 codec_type; /* Codec type */ + u8 str_type; /* 1 = voice 2 = music */ + u8 operation; /* Playback or Capture */ + u8 protected_str; /* 0=Non DRM, 1=DRM */ + u8 time_slots; + u8 reserved; /* Reserved */ + u16 result; /* Result used for acknowledgment */ +} __packed; + +/* Library info structure */ +struct module_info { + u32 lib_version; + u32 lib_type;/*TBD- KLOCKWORK u8 lib_type;*/ + u32 media_type; + u8 lib_name[12]; + u32 lib_caps; + unsigned char b_date[16]; /* Lib build date */ + unsigned char b_time[16]; /* Lib build time */ +} __packed; + +/* Library slot info */ +struct lib_slot_info { + u8 slot_num; /* 1 or 2 */ + u8 reserved1; + u16 reserved2; + u32 iram_size; /* slot size in IRAM */ + u32 dram_size; /* slot size in DRAM */ + u32 iram_offset; /* starting offset of slot in IRAM */ + u32 dram_offset; /* starting offset of slot in DRAM */ +} __packed; + +struct snd_ppp_mixer_params { + __u32 type; /*Type of the parameter */ + __u32 size; + __u32 input_stream_bitmap; /*Input stream Bit Map*/ +} __packed; + +struct snd_sst_lib_download { + struct module_info lib_info; /* library info type, capabilities etc */ + struct lib_slot_info slot_info; /* slot info to be downloaded */ + u32 mod_entry_pt; +}; + +struct snd_sst_lib_download_info { + struct snd_sst_lib_download dload_lib; + u16 result; /* Result used for acknowledgment */ + u8 pvt_id; /* Private ID */ + u8 reserved; /* for alignment */ +}; struct snd_pcm_params { u8 num_chan; /* 1=Mono, 2=Stereo */ u8 pcm_wd_sz; /* 16/24 - bit*/ - u32 reserved; /* Bitrate in bits per second */ - u32 sfreq; /* Sampling rate in Hz */ - u8 use_offload_path; + u8 use_offload_path; /* 0-PCM using period elpased & ALSA interfaces + 1-PCM stream via compressed interface */ u8 reserved2; - u16 reserved3; + u32 sfreq; /* Sampling rate in Hz */ u8 channel_map[8]; } __packed; @@ -76,6 +355,7 @@ struct snd_aac_params { struct snd_wma_params { u8 num_chan; /* 1=Mono, 2=Stereo */ u8 pcm_wd_sz; /* 16/24 - bit*/ + u16 reserved1; u32 brate; /* Use the hard coded value. */ u32 sfreq; /* Sampling freq eg. 8000, 441000, 48000 */ u32 channel_mask; /* Channel Mask */ @@ -101,26 +381,153 @@ struct sst_address_info { }; struct snd_sst_alloc_params_ext { - struct sst_address_info ring_buf_info[8]; - u8 sg_count; - u8 reserved; - u16 reserved2; - u32 frag_size; /*Number of samples after which period elapsed + __u16 sg_count; + __u16 reserved; + __u32 frag_size; /*Number of samples after which period elapsed message is sent valid only if path = 0*/ -} __packed; + struct sst_address_info ring_buf_info[8]; +}; struct snd_sst_stream_params { union snd_sst_codec_params uc; } __packed; struct snd_sst_params { + u32 result; u32 stream_id; u8 codec; u8 ops; u8 stream_type; u8 device_type; + u8 task; struct snd_sst_stream_params sparams; struct snd_sst_alloc_params_ext aparams; }; +struct snd_sst_alloc_mrfld { + u16 codec_type; + u8 operation; + u8 sg_count; + struct sst_address_info ring_buf_info[8]; + u32 frag_size; + u32 ts; + struct snd_sst_stream_params codec_params; +} __packed; + +/* Alloc stream params structure */ +struct snd_sst_alloc_params { + struct snd_sst_str_type str_type; + struct snd_sst_stream_params stream_params; + struct snd_sst_alloc_params_ext alloc_params; +} __packed; + +/* Alloc stream response message */ +struct snd_sst_alloc_response { + struct snd_sst_str_type str_type; /* Stream type for allocation */ + struct snd_sst_lib_download lib_dnld; /* Valid only for codec dnld */ +}; + +/* Drop response */ +struct snd_sst_drop_response { + u32 result; + u32 bytes; +}; + +struct snd_sst_async_msg { + u32 msg_id; /* Async msg id */ + u32 payload[0]; +}; + +struct snd_sst_async_err_msg { + u32 fw_resp; /* Firmware Result */ + u32 lib_resp; /*Library result */ +} __packed; + +struct snd_sst_vol { + u32 stream_id; + s32 volume; + u32 ramp_duration; + u32 ramp_type; /* Ramp type, default=0 */ +}; + +/* Gain library parameters for mrfld + * based on DSP command spec v0.82 + */ +struct snd_sst_gain_v2 { + u16 gain_cell_num; /* num of gain cells to modify*/ + u8 cell_nbr_idx; /* instance index*/ + u8 cell_path_idx; /* pipe-id */ + u16 module_id; /*module id */ + u16 left_cell_gain; /* left gain value in dB*/ + u16 right_cell_gain; /* right gain value in dB*/ + u16 gain_time_const; /* gain time constant*/ +} __packed; + +struct snd_sst_mute { + u32 stream_id; + u32 mute; +}; + +struct snd_sst_runtime_params { + u8 type; + u8 str_id; + u8 size; + u8 rsvd; + void *addr; +} __packed; + +enum stream_param_type { + SST_SET_TIME_SLOT = 0, + SST_SET_CHANNEL_INFO = 1, + OTHERS = 2, /*reserved for future params*/ +}; + +/* CSV Voice call routing structure */ +struct snd_sst_control_routing { + u8 control; /* 0=start, 1=Stop */ + u8 reserved[3]; /* Reserved- for 32 bit alignment */ +}; + +struct ipc_post { + struct list_head node; + union ipc_header header; /* driver specific */ + bool is_large; + bool is_process_reply; + union ipc_header_mrfld mrfld_header; + char *mailbox_data; +}; + +struct snd_sst_ctxt_params { + u32 address; /* Physical Address in DDR where the context is stored */ + u32 size; /* size of the context */ +}; + +struct snd_sst_lpe_log_params { + u8 dbg_type; + u8 module_id; + u8 log_level; + u8 reserved; +} __packed; + +enum snd_sst_bytes_type { + SND_SST_BYTES_SET = 0x1, + SND_SST_BYTES_GET = 0x2, +}; + +struct snd_sst_bytes_v2 { + u8 type; + u8 ipc_msg; + u8 block; + u8 task_id; + u8 pipe_id; + u8 rsvd; + u16 len; + char bytes[0]; +}; + +#define MAX_VTSV_FILES 2 +struct snd_sst_vtsv_info { + struct sst_address_info vfiles[MAX_VTSV_FILES]; +} __packed; + #endif /* __SST_MFLD_DSP_H__ */ diff --git a/sound/soc/intel/sst-mfld-platform-compress.c b/sound/soc/intel/sst-mfld-platform-compress.c index 02abd19fce1d..29c059ca19e8 100644 --- a/sound/soc/intel/sst-mfld-platform-compress.c +++ b/sound/soc/intel/sst-mfld-platform-compress.c @@ -100,14 +100,19 @@ static int sst_platform_compr_set_params(struct snd_compr_stream *cstream, int retval; struct snd_sst_params str_params; struct sst_compress_cb cb; + struct snd_soc_pcm_runtime *rtd = cstream->private_data; + struct snd_soc_platform *platform = rtd->platform; + struct sst_data *ctx = snd_soc_platform_get_drvdata(platform); stream = cstream->runtime->private_data; /* construct fw structure for this*/ memset(&str_params, 0, sizeof(str_params)); - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.stream_type = SST_STREAM_TYPE_MUSIC; - str_params.device_type = SND_SST_DEVICE_COMPRESS; + /* fill the device type and stream id to pass to SST driver */ + retval = sst_fill_stream_params(cstream, ctx, &str_params, true); + pr_debug("compr_set_params: fill stream params ret_val = 0x%x\n", retval); + if (retval < 0) + return retval; switch (params->codec.id) { case SND_AUDIOCODEC_MP3: { diff --git a/sound/soc/intel/sst-mfld-platform-pcm.c b/sound/soc/intel/sst-mfld-platform-pcm.c index 7c790f51d259..706212a6a68c 100644 --- a/sound/soc/intel/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/sst-mfld-platform-pcm.c @@ -1,7 +1,7 @@ /* * sst_mfld_platform.c - Intel MID Platform driver * - * Copyright (C) 2010-2013 Intel Corp + * Copyright (C) 2010-2014 Intel Corp * Author: Vinod Koul <vinod.koul@intel.com> * Author: Harsha Priya <priya.harsha@intel.com> * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ @@ -27,7 +27,9 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/compress_driver.h> +#include <asm/platform_sst_audio.h> #include "sst-mfld-platform.h" +#include "sst-atom-controls.h" struct sst_device *sst; static DEFINE_MUTEX(sst_lock); @@ -92,6 +94,13 @@ static struct snd_pcm_hardware sst_platform_pcm_hw = { .fifo_size = SST_FIFO_SIZE, }; +static struct sst_dev_stream_map dpcm_strm_map[] = { + {0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF}, /* Reserved, not in use */ + {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA1_IN, SST_TASK_ID_MEDIA, 0}, + {MERR_DPCM_COMPR, 0, SNDRV_PCM_STREAM_PLAYBACK, PIPE_MEDIA0_IN, SST_TASK_ID_MEDIA, 0}, + {MERR_DPCM_AUDIO, 0, SNDRV_PCM_STREAM_CAPTURE, PIPE_PCM1_OUT, SST_TASK_ID_MEDIA, 0}, +}; + /* MFLD - MSIC */ static struct snd_soc_dai_driver sst_platform_dai[] = { { @@ -143,58 +152,142 @@ static inline int sst_get_stream_status(struct sst_runtime_stream *stream) return state; } +static void sst_fill_alloc_params(struct snd_pcm_substream *substream, + struct snd_sst_alloc_params_ext *alloc_param) +{ + unsigned int channels; + snd_pcm_uframes_t period_size; + ssize_t periodbytes; + ssize_t buffer_bytes = snd_pcm_lib_buffer_bytes(substream); + u32 buffer_addr = virt_to_phys(substream->dma_buffer.area); + + channels = substream->runtime->channels; + period_size = substream->runtime->period_size; + periodbytes = samples_to_bytes(substream->runtime, period_size); + alloc_param->ring_buf_info[0].addr = buffer_addr; + alloc_param->ring_buf_info[0].size = buffer_bytes; + alloc_param->sg_count = 1; + alloc_param->reserved = 0; + alloc_param->frag_size = periodbytes * channels; + +} static void sst_fill_pcm_params(struct snd_pcm_substream *substream, - struct sst_pcm_params *param) + struct snd_sst_stream_params *param) { + param->uc.pcm_params.num_chan = (u8) substream->runtime->channels; + param->uc.pcm_params.pcm_wd_sz = substream->runtime->sample_bits; + param->uc.pcm_params.sfreq = substream->runtime->rate; + + /* PCM stream via ALSA interface */ + param->uc.pcm_params.use_offload_path = 0; + param->uc.pcm_params.reserved2 = 0; + memset(param->uc.pcm_params.channel_map, 0, sizeof(u8)); - param->num_chan = (u8) substream->runtime->channels; - param->pcm_wd_sz = substream->runtime->sample_bits; - param->reserved = 0; - param->sfreq = substream->runtime->rate; - param->ring_buffer_size = snd_pcm_lib_buffer_bytes(substream); - param->period_count = substream->runtime->period_size; - param->ring_buffer_addr = virt_to_phys(substream->dma_buffer.area); - pr_debug("period_cnt = %d\n", param->period_count); - pr_debug("sfreq= %d, wd_sz = %d\n", param->sfreq, param->pcm_wd_sz); } -static int sst_platform_alloc_stream(struct snd_pcm_substream *substream) +static int sst_get_stream_mapping(int dev, int sdev, int dir, + struct sst_dev_stream_map *map, int size) +{ + int i; + + if (map == NULL) + return -EINVAL; + + + /* index 0 is not used in stream map */ + for (i = 1; i < size; i++) { + if ((map[i].dev_num == dev) && (map[i].direction == dir)) + return i; + } + return 0; +} + +int sst_fill_stream_params(void *substream, + const struct sst_data *ctx, struct snd_sst_params *str_params, bool is_compress) +{ + int map_size; + int index; + struct sst_dev_stream_map *map; + struct snd_pcm_substream *pstream = NULL; + struct snd_compr_stream *cstream = NULL; + + map = ctx->pdata->pdev_strm_map; + map_size = ctx->pdata->strm_map_size; + + if (is_compress == true) + cstream = (struct snd_compr_stream *)substream; + else + pstream = (struct snd_pcm_substream *)substream; + + str_params->stream_type = SST_STREAM_TYPE_MUSIC; + + /* For pcm streams */ + if (pstream) { + index = sst_get_stream_mapping(pstream->pcm->device, + pstream->number, pstream->stream, + map, map_size); + if (index <= 0) + return -EINVAL; + + str_params->stream_id = index; + str_params->device_type = map[index].device_id; + str_params->task = map[index].task_id; + + str_params->ops = (u8)pstream->stream; + } + + if (cstream) { + index = sst_get_stream_mapping(cstream->device->device, + 0, cstream->direction, + map, map_size); + if (index <= 0) + return -EINVAL; + str_params->stream_id = index; + str_params->device_type = map[index].device_id; + str_params->task = map[index].task_id; + + str_params->ops = (u8)cstream->direction; + } + return 0; +} + +static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, + struct snd_soc_platform *platform) { struct sst_runtime_stream *stream = substream->runtime->private_data; - struct sst_pcm_params param = {0}; - struct sst_stream_params str_params = {0}; - int ret_val; + struct snd_sst_stream_params param = {{{0,},},}; + struct snd_sst_params str_params = {0}; + struct snd_sst_alloc_params_ext alloc_params = {0}; + int ret_val = 0; + struct sst_data *ctx = snd_soc_platform_get_drvdata(platform); /* set codec params and inform SST driver the same */ sst_fill_pcm_params(substream, ¶m); + sst_fill_alloc_params(substream, &alloc_params); substream->runtime->dma_area = substream->dma_buffer.area; str_params.sparams = param; - str_params.codec = param.codec; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - str_params.ops = STREAM_OPS_PLAYBACK; - str_params.device_type = substream->pcm->device + 1; - pr_debug("Playbck stream,Device %d\n", - substream->pcm->device); - } else { - str_params.ops = STREAM_OPS_CAPTURE; - str_params.device_type = SND_SST_DEVICE_CAPTURE; - pr_debug("Capture stream,Device %d\n", - substream->pcm->device); - } - ret_val = stream->ops->open(&str_params); - pr_debug("SST_SND_PLAY/CAPTURE ret_val = %x\n", ret_val); + str_params.aparams = alloc_params; + str_params.codec = SST_CODEC_TYPE_PCM; + + /* fill the device type and stream id to pass to SST driver */ + ret_val = sst_fill_stream_params(substream, ctx, &str_params, false); if (ret_val < 0) return ret_val; - stream->stream_info.str_id = ret_val; - pr_debug("str id : %d\n", stream->stream_info.str_id); + stream->stream_info.str_id = str_params.stream_id; + + ret_val = stream->ops->open(&str_params); + if (ret_val <= 0) + return ret_val; + + return ret_val; } -static void sst_period_elapsed(void *mad_substream) +static void sst_period_elapsed(void *arg) { - struct snd_pcm_substream *substream = mad_substream; + struct snd_pcm_substream *substream = arg; struct sst_runtime_stream *stream; int status; @@ -218,7 +311,7 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) pr_debug("setting buffer ptr param\n"); sst_set_stream_status(stream, SST_PLATFORM_INIT); stream->stream_info.period_elapsed = sst_period_elapsed; - stream->stream_info.mad_substream = substream; + stream->stream_info.arg = substream; stream->stream_info.buffer_ptr = 0; stream->stream_info.sfreq = substream->runtime->rate; ret_val = stream->ops->device_control( @@ -230,19 +323,12 @@ static int sst_platform_init_stream(struct snd_pcm_substream *substream) } /* end -- helper functions */ -static int sst_platform_open(struct snd_pcm_substream *substream) +static int sst_media_open(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { + int ret_val = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct sst_runtime_stream *stream; - int ret_val; - - pr_debug("sst_platform_open called\n"); - - snd_soc_set_runtime_hwparams(substream, &sst_platform_pcm_hw); - ret_val = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret_val < 0) - return ret_val; stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) @@ -251,50 +337,69 @@ static int sst_platform_open(struct snd_pcm_substream *substream) /* get the sst ops */ mutex_lock(&sst_lock); - if (!sst) { + if (!sst || + !try_module_get(sst->dev->driver->owner)) { pr_err("no device available to run\n"); - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; - } - if (!try_module_get(sst->dev->driver->owner)) { - mutex_unlock(&sst_lock); - kfree(stream); - return -ENODEV; + ret_val = -ENODEV; + goto out_ops; } stream->ops = sst->ops; mutex_unlock(&sst_lock); stream->stream_info.str_id = 0; - sst_set_stream_status(stream, SST_PLATFORM_INIT); - stream->stream_info.mad_substream = substream; + + stream->stream_info.arg = substream; /* allocate memory for SST API set */ runtime->private_data = stream; - return 0; + /* Make sure, that the period size is always even */ + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIODS, 2); + + return snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); +out_ops: + kfree(stream); + mutex_unlock(&sst_lock); + return ret_val; } -static int sst_platform_close(struct snd_pcm_substream *substream) +static void sst_media_close(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct sst_runtime_stream *stream; int ret_val = 0, str_id; - pr_debug("sst_platform_close called\n"); stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (str_id) ret_val = stream->ops->close(str_id); module_put(sst->dev->driver->owner); kfree(stream); - return ret_val; } -static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) +static inline unsigned int get_current_pipe_id(struct snd_soc_platform *platform, + struct snd_pcm_substream *substream) +{ + struct sst_data *sst = snd_soc_platform_get_drvdata(platform); + struct sst_dev_stream_map *map = sst->pdata->pdev_strm_map; + struct sst_runtime_stream *stream = + substream->runtime->private_data; + u32 str_id = stream->stream_info.str_id; + unsigned int pipe_id; + pipe_id = map[str_id].device_id; + + pr_debug("%s: got pipe_id = %#x for str_id = %d\n", + __func__, pipe_id, str_id); + return pipe_id; +} + +static int sst_media_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct sst_runtime_stream *stream; int ret_val = 0, str_id; - pr_debug("sst_platform_pcm_prepare called\n"); stream = substream->runtime->private_data; str_id = stream->stream_info.str_id; if (stream->stream_info.str_id) { @@ -303,8 +408,8 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) return ret_val; } - ret_val = sst_platform_alloc_stream(substream); - if (ret_val < 0) + ret_val = sst_platform_alloc_stream(substream, dai->platform); + if (ret_val <= 0) return ret_val; snprintf(substream->pcm->id, sizeof(substream->pcm->id), "%d", stream->stream_info.str_id); @@ -316,6 +421,41 @@ static int sst_platform_pcm_prepare(struct snd_pcm_substream *substream) return ret_val; } +static int sst_media_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); + return 0; +} + +static int sst_media_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + return snd_pcm_lib_free_pages(substream); +} + +static struct snd_soc_dai_ops sst_media_dai_ops = { + .startup = sst_media_open, + .shutdown = sst_media_close, + .prepare = sst_media_prepare, + .hw_params = sst_media_hw_params, + .hw_free = sst_media_hw_free, +}; + +static int sst_platform_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + + if (substream->pcm->internal) + return 0; + + runtime = substream->runtime; + runtime->hw = sst_platform_pcm_hw; + return 0; +} + static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { @@ -331,7 +471,7 @@ static int sst_platform_pcm_trigger(struct snd_pcm_substream *substream, pr_debug("sst: Trigger Start\n"); str_cmd = SST_SND_START; status = SST_PLATFORM_RUNNING; - stream->stream_info.mad_substream = substream; + stream->stream_info.arg = substream; break; case SNDRV_PCM_TRIGGER_STOP: pr_debug("sst: in stop\n"); @@ -377,32 +517,15 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer pr_err("sst: error code = %d\n", ret_val); return ret_val; } - return stream->stream_info.buffer_ptr; -} - -static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); - - return 0; -} - -static int sst_platform_pcm_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); + substream->runtime->delay = str_info->pcm_delay; + return str_info->buffer_ptr; } static struct snd_pcm_ops sst_platform_ops = { .open = sst_platform_open, - .close = sst_platform_close, .ioctl = snd_pcm_lib_ioctl, - .prepare = sst_platform_pcm_prepare, .trigger = sst_platform_pcm_trigger, .pointer = sst_platform_pcm_pointer, - .hw_params = sst_platform_pcm_hw_params, - .hw_free = sst_platform_pcm_hw_free, }; static void sst_pcm_free(struct snd_pcm *pcm) @@ -413,15 +536,15 @@ static void sst_pcm_free(struct snd_pcm *pcm) static int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; int retval = 0; - pr_debug("sst_pcm_new called\n"); - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || - pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + if (dai->driver->playback.channels_min || + dai->driver->capture.channels_min) { retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), + snd_dma_continuous_data(GFP_DMA), SST_MIN_BUFFER, SST_MAX_BUFFER); if (retval) { pr_err("dma buffer allocationf fail\n"); @@ -445,10 +568,28 @@ static const struct snd_soc_component_driver sst_component = { static int sst_platform_probe(struct platform_device *pdev) { + struct sst_data *drv; int ret; + struct sst_platform_data *pdata; + + drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); + if (drv == NULL) { + pr_err("kzalloc failed\n"); + return -ENOMEM; + } + + pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); + if (pdata == NULL) { + pr_err("kzalloc failed for pdata\n"); + return -ENOMEM; + } + + pdata->pdev_strm_map = dpcm_strm_map; + pdata->strm_map_size = ARRAY_SIZE(dpcm_strm_map); + drv->pdata = pdata; + mutex_init(&drv->lock); + dev_set_drvdata(&pdev->dev, drv); - pr_debug("sst_platform_probe called\n"); - sst = NULL; ret = snd_soc_register_platform(&pdev->dev, &sst_soc_platform_drv); if (ret) { pr_err("registering soc platform failed\n"); diff --git a/sound/soc/intel/sst-mfld-platform.h b/sound/soc/intel/sst-mfld-platform.h index 6c5e7dc49e3c..6c6a42c08e24 100644 --- a/sound/soc/intel/sst-mfld-platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -39,9 +39,10 @@ extern struct sst_device *sst; struct pcm_stream_info { int str_id; - void *mad_substream; - void (*period_elapsed) (void *mad_substream); + void *arg; + void (*period_elapsed) (void *arg); unsigned long long buffer_ptr; + unsigned long long pcm_delay; int sfreq; }; @@ -62,7 +63,9 @@ enum sst_controls { SST_SND_BUFFER_POINTER = 0x05, SST_SND_STREAM_INIT = 0x06, SST_SND_START = 0x07, - SST_MAX_CONTROLS = 0x07, + SST_SET_BYTE_STREAM = 0x100A, + SST_GET_BYTE_STREAM = 0x100B, + SST_MAX_CONTROLS = SST_GET_BYTE_STREAM, }; enum sst_stream_ops { @@ -124,8 +127,9 @@ struct compress_sst_ops { }; struct sst_ops { - int (*open) (struct sst_stream_params *str_param); + int (*open) (struct snd_sst_params *str_param); int (*device_control) (int cmd, void *arg); + int (*set_generic_params)(enum sst_controls cmd, void *arg); int (*close) (unsigned int str_id); }; @@ -143,10 +147,27 @@ struct sst_device { char *name; struct device *dev; struct sst_ops *ops; + struct platform_device *pdev; struct compress_sst_ops *compr_ops; }; +struct sst_data; void sst_set_stream_status(struct sst_runtime_stream *stream, int state); +int sst_fill_stream_params(void *substream, const struct sst_data *ctx, + struct snd_sst_params *str_params, bool is_compress); + +struct sst_algo_int_control_v2 { + struct soc_mixer_control mc; + u16 module_id; /* module identifieer */ + u16 pipe_id; /* location info: pipe_id + instance_id */ + u16 instance_id; + unsigned int value; /* Value received is stored here */ +}; +struct sst_data { + struct platform_device *pdev; + struct sst_platform_data *pdata; + struct mutex lock; +}; int sst_register_dsp(struct sst_device *sst); int sst_unregister_dsp(struct sst_device *sst); #endif diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 06f4e8aa93ae..132bb83f8e99 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,6 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood and Dove chips" - depends on ARCH_KIRKWOOD || ARCH_DOVE || ARCH_MVEBU || MACH_KIRKWOOD || COMPILE_TEST + depends on ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the @@ -15,20 +15,3 @@ config SND_KIRKWOOD_SOC_ARMADA370_DB Say Y if you want to add support for SoC audio on the Armada 370 Development Board. -config SND_KIRKWOOD_SOC_OPENRD - tristate "SoC Audio support for Kirkwood Openrd Client" - depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST) - depends on I2C - select SND_SOC_CS42L51 - help - Say Y if you want to add support for SoC audio on - Openrd Client. - -config SND_KIRKWOOD_SOC_T5325 - tristate "SoC Audio support for HP t5325" - depends on SND_KIRKWOOD_SOC && (MACH_T5325 || COMPILE_TEST) && I2C - select SND_SOC_ALC5623 - help - Say Y if you want to add support for SoC audio on - the HP t5325 thin client. - diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 7c1d8fe09e6b..c36b03d8006c 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -2,10 +2,6 @@ snd-soc-kirkwood-objs := kirkwood-dma.o kirkwood-i2s.o obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o -snd-soc-openrd-objs := kirkwood-openrd.o -snd-soc-t5325-objs := kirkwood-t5325.o snd-soc-armada-370-db-objs := armada-370-db.o obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o -obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index aac22fccdcdc..4cf2245950d7 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -28,11 +28,12 @@ static struct kirkwood_dma_data *kirkwood_priv(struct snd_pcm_substream *subs) } static struct snd_pcm_hardware kirkwood_dma_snd_hw = { - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_PAUSE), + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, .buffer_bytes_max = KIRKWOOD_SND_MAX_BUFFER_BYTES, .period_bytes_min = KIRKWOOD_SND_MIN_PERIOD_BYTES, .period_bytes_max = KIRKWOOD_SND_MAX_PERIOD_BYTES, diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 9f842222e798..0704cd6d2314 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -212,7 +212,8 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, KIRKWOOD_PLAYCTL_SIZE_MASK); priv->ctl_play |= ctl_play; } else { - priv->ctl_rec &= ~KIRKWOOD_RECCTL_SIZE_MASK; + priv->ctl_rec &= ~(KIRKWOOD_RECCTL_ENABLE_MASK | + KIRKWOOD_RECCTL_SIZE_MASK); priv->ctl_rec |= ctl_rec; } @@ -221,14 +222,24 @@ static int kirkwood_i2s_hw_params(struct snd_pcm_substream *substream, return 0; } +static unsigned kirkwood_i2s_play_mute(unsigned ctl) +{ + if (!(ctl & KIRKWOOD_PLAYCTL_I2S_EN)) + ctl |= KIRKWOOD_PLAYCTL_I2S_MUTE; + if (!(ctl & KIRKWOOD_PLAYCTL_SPDIF_EN)) + ctl |= KIRKWOOD_PLAYCTL_SPDIF_MUTE; + return ctl; +} + static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { + struct snd_pcm_runtime *runtime = substream->runtime; struct kirkwood_dma_data *priv = snd_soc_dai_get_drvdata(dai); uint32_t ctl, value; ctl = readl(priv->io + KIRKWOOD_PLAYCTL); - if (ctl & KIRKWOOD_PLAYCTL_PAUSE) { + if ((ctl & KIRKWOOD_PLAYCTL_ENABLE_MASK) == 0) { unsigned timeout = 5000; /* * The Armada510 spec says that if we enter pause mode, the @@ -256,14 +267,16 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ else ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ - + ctl = kirkwood_i2s_play_mute(ctl); value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_PLAYCTL); /* enable interrupts */ - value = readl(priv->io + KIRKWOOD_INT_MASK); - value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES; - writel(value, priv->io + KIRKWOOD_INT_MASK); + if (!runtime->no_period_wakeup) { + value = readl(priv->io + KIRKWOOD_INT_MASK); + value |= KIRKWOOD_INT_CAUSE_PLAY_BYTES; + writel(value, priv->io + KIRKWOOD_INT_MASK); + } /* enable playback */ writel(ctl, priv->io + KIRKWOOD_PLAYCTL); @@ -295,6 +308,7 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ctl &= ~(KIRKWOOD_PLAYCTL_PAUSE | KIRKWOOD_PLAYCTL_I2S_MUTE | KIRKWOOD_PLAYCTL_SPDIF_MUTE); + ctl = kirkwood_i2s_play_mute(ctl); writel(ctl, priv->io + KIRKWOOD_PLAYCTL); break; @@ -322,8 +336,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, else ctl &= ~KIRKWOOD_RECCTL_I2S_EN; /* spdif */ - value = ctl & ~(KIRKWOOD_RECCTL_I2S_EN | - KIRKWOOD_RECCTL_SPDIF_EN); + value = ctl & ~KIRKWOOD_RECCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_RECCTL); /* enable interrupts */ @@ -347,7 +360,7 @@ static int kirkwood_i2s_rec_trigger(struct snd_pcm_substream *substream, /* disable all records */ value = readl(priv->io + KIRKWOOD_RECCTL); - value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN); + value &= ~KIRKWOOD_RECCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_RECCTL); break; @@ -411,7 +424,7 @@ static int kirkwood_i2s_init(struct kirkwood_dma_data *priv) writel(value, priv->io + KIRKWOOD_PLAYCTL); value = readl(priv->io + KIRKWOOD_RECCTL); - value &= ~(KIRKWOOD_RECCTL_I2S_EN | KIRKWOOD_RECCTL_SPDIF_EN); + value &= ~KIRKWOOD_RECCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_RECCTL); return 0; diff --git a/sound/soc/kirkwood/kirkwood-openrd.c b/sound/soc/kirkwood/kirkwood-openrd.c deleted file mode 100644 index 65f2a5b9ec3b..000000000000 --- a/sound/soc/kirkwood/kirkwood-openrd.c +++ /dev/null @@ -1,109 +0,0 @@ -/* - * kirkwood-openrd.c - * - * (c) 2010 Arnaud Patard <apatard@mandriva.com> - * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org> - * - * 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 <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <sound/soc.h> -#include <linux/platform_data/asoc-kirkwood.h> -#include "../codecs/cs42l51.h" - -static int openrd_client_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; - unsigned int freq; - - switch (params_rate(params)) { - default: - case 44100: - freq = 11289600; - break; - case 48000: - freq = 12288000; - break; - case 96000: - freq = 24576000; - break; - } - - return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); - -} - -static struct snd_soc_ops openrd_client_ops = { - .hw_params = openrd_client_hw_params, -}; - - -static struct snd_soc_dai_link openrd_client_dai[] = { -{ - .name = "CS42L51", - .stream_name = "CS42L51 HiFi", - .cpu_dai_name = "i2s", - .platform_name = "mvebu-audio", - .codec_dai_name = "cs42l51-hifi", - .codec_name = "cs42l51-codec.0-004a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, - .ops = &openrd_client_ops, -}, -}; - - -static struct snd_soc_card openrd_client = { - .name = "OpenRD Client", - .owner = THIS_MODULE, - .dai_link = openrd_client_dai, - .num_links = ARRAY_SIZE(openrd_client_dai), -}; - -static int openrd_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &openrd_client; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static int openrd_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 openrd_driver = { - .driver = { - .name = "openrd-client-audio", - .owner = THIS_MODULE, - }, - .probe = openrd_probe, - .remove = openrd_remove, -}; - -module_platform_driver(openrd_driver); - -/* Module information */ -MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); -MODULE_DESCRIPTION("ALSA SoC OpenRD Client"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:openrd-client-audio"); diff --git a/sound/soc/kirkwood/kirkwood-t5325.c b/sound/soc/kirkwood/kirkwood-t5325.c deleted file mode 100644 index 844b8415a011..000000000000 --- a/sound/soc/kirkwood/kirkwood-t5325.c +++ /dev/null @@ -1,116 +0,0 @@ -/* - * kirkwood-t5325.c - * - * (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org> - * - * 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 <linux/interrupt.h> -#include <linux/platform_device.h> -#include <linux/slab.h> -#include <sound/soc.h> -#include <linux/platform_data/asoc-kirkwood.h> -#include "../codecs/alc5623.h" - -static int t5325_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; - unsigned int freq; - - freq = params_rate(params) * 256; - - return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); - -} - -static struct snd_soc_ops t5325_ops = { - .hw_params = t5325_hw_params, -}; - -static const struct snd_soc_dapm_widget t5325_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route t5325_route[] = { - { "Headphone Jack", NULL, "HPL" }, - { "Headphone Jack", NULL, "HPR" }, - - {"Speaker", NULL, "SPKOUT"}, - {"Speaker", NULL, "SPKOUTN"}, - - { "MIC1", NULL, "Mic Jack" }, - { "MIC2", NULL, "Mic Jack" }, -}; - -static struct snd_soc_dai_link t5325_dai[] = { -{ - .name = "ALC5621", - .stream_name = "ALC5621 HiFi", - .cpu_dai_name = "i2s", - .platform_name = "mvebu-audio", - .codec_dai_name = "alc5621-hifi", - .codec_name = "alc562x-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, - .ops = &t5325_ops, -}, -}; - -static struct snd_soc_card t5325 = { - .name = "t5325", - .owner = THIS_MODULE, - .dai_link = t5325_dai, - .num_links = ARRAY_SIZE(t5325_dai), - - .dapm_widgets = t5325_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(t5325_dapm_widgets), - .dapm_routes = t5325_route, - .num_dapm_routes = ARRAY_SIZE(t5325_route), -}; - -static int t5325_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &t5325; - int ret; - - card->dev = &pdev->dev; - - ret = snd_soc_register_card(card); - if (ret) - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", - ret); - return ret; -} - -static int t5325_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 t5325_driver = { - .driver = { - .name = "t5325-audio", - .owner = THIS_MODULE, - }, - .probe = t5325_probe, - .remove = t5325_remove, -}; - -module_platform_driver(t5325_driver); - -MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>"); -MODULE_DESCRIPTION("ALSA SoC t5325 audio client"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:t5325-audio"); diff --git a/sound/soc/kirkwood/kirkwood.h b/sound/soc/kirkwood/kirkwood.h index bf23afbba1d7..90e32a781424 100644 --- a/sound/soc/kirkwood/kirkwood.h +++ b/sound/soc/kirkwood/kirkwood.h @@ -38,6 +38,9 @@ #define KIRKWOOD_RECCTL_SIZE_24 (1<<0) #define KIRKWOOD_RECCTL_SIZE_32 (0<<0) +#define KIRKWOOD_RECCTL_ENABLE_MASK (KIRKWOOD_RECCTL_SPDIF_EN | \ + KIRKWOOD_RECCTL_I2S_EN) + #define KIRKWOOD_REC_BUF_ADDR 0x1004 #define KIRKWOOD_REC_BUF_SIZE 0x1008 #define KIRKWOOD_REC_BYTE_COUNT 0x100C @@ -121,9 +124,9 @@ /* Theses values come from the marvell alsa driver */ /* need to find where they come from */ -#define KIRKWOOD_SND_MIN_PERIODS 8 +#define KIRKWOOD_SND_MIN_PERIODS 2 #define KIRKWOOD_SND_MAX_PERIODS 16 -#define KIRKWOOD_SND_MIN_PERIOD_BYTES 0x800 +#define KIRKWOOD_SND_MIN_PERIOD_BYTES 256 #define KIRKWOOD_SND_MAX_PERIOD_BYTES 0x8000 #define KIRKWOOD_SND_MAX_BUFFER_BYTES (KIRKWOOD_SND_MAX_PERIOD_BYTES \ * KIRKWOOD_SND_MAX_PERIODS) |