diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index eabf66af12cd..5d230cee3fa7 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -789,13 +789,14 @@ static struct snd_soc_dai_driver atmel_ssc_dai[NUM_SSC_DEVICES] = { static __devinit int asoc_ssc_probe(struct platform_device *pdev) { - return snd_soc_register_dais(&pdev->dev, atmel_ssc_dai, - ARRAY_SIZE(atmel_ssc_dai)); + BUG_ON(pdev->id < 0); + BUG_ON(pdev->id >= ARRAY_SIZE(atmel_ssc_dai)); + return snd_soc_register_dai(&pdev->dev, &atmel_ssc_dai[pdev->id]); } static int __devexit asoc_ssc_remove(struct platform_device *pdev) { - snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(atmel_ssc_dai)); + snd_soc_unregister_dai(&pdev->dev); return 0; } @@ -809,6 +810,56 @@ static struct platform_driver asoc_ssc_driver = { .remove = __devexit_p(asoc_ssc_remove), }; +/** + * atmel_ssc_set_audio - Allocate the specified SSC for audio use. + */ +int atmel_ssc_set_audio(int ssc_id) +{ + struct ssc_device *ssc; + static struct platform_device *dma_pdev; + struct platform_device *ssc_pdev; + int ret; + + if (ssc_id < 0 || ssc_id >= ARRAY_SIZE(atmel_ssc_dai)) + return -EINVAL; + + /* Allocate a dummy device for DMA if we don't have one already */ + if (!dma_pdev) { + dma_pdev = platform_device_alloc("atmel-pcm-audio", -1); + if (!dma_pdev) + return -ENOMEM; + + ret = platform_device_add(dma_pdev); + if (ret < 0) { + platform_device_put(dma_pdev); + dma_pdev = NULL; + return ret; + } + } + + ssc_pdev = platform_device_alloc("atmel-ssc-dai", ssc_id); + if (!ssc_pdev) { + ssc_free(ssc); + return -ENOMEM; + } + + /* If we can grab the SSC briefly to parent the DAI device off it */ + ssc = ssc_request(ssc_id); + if (IS_ERR(ssc)) + pr_warn("Unable to parent ASoC SSC DAI on SSC: %ld\n", + PTR_ERR(ssc)); + else + ssc_pdev->dev.parent = &(ssc->pdev->dev); + ssc_free(ssc); + + ret = platform_device_add(ssc_pdev); + if (ret < 0) + platform_device_put(ssc_pdev); + + return ret; +} +EXPORT_SYMBOL_GPL(atmel_ssc_set_audio); + static int __init snd_atmel_ssc_init(void) { return platform_driver_register(&asoc_ssc_driver); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index 392a46953112..5d4f0f9b4d9a 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -117,4 +117,6 @@ struct atmel_ssc_info { struct atmel_ssc_state ssc_state; }; +int atmel_ssc_set_audio(int ssc); + #endif /* _AT91_SSC_DAI_H */ diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 66a6f1879689..293569dfd0ed 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -146,7 +146,7 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) "at91sam9g20ek_wm8731 " ": at91sam9g20ek_wm8731_init() called\n"); - ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, MCLK_RATE, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "Failed to set WM8731 SYSCLK: %d\n", ret); @@ -183,8 +183,8 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { .cpu_dai_name = "atmel-ssc-dai.0", .codec_dai_name = "wm8731-hifi", .init = at91sam9g20ek_wm8731_init, - .platform_name = "atmel_pcm-audio", - .codec_name = "wm8731-codec.0-001a", + .platform_name = "atmel-pcm-audio", + .codec_name = "wm8731-codec.0-001b", .ops = &at91sam9g20ek_ops, }; @@ -205,6 +205,12 @@ static int __init at91sam9g20ek_init(void) if (!(machine_is_at91sam9g20ek() || machine_is_at91sam9g20ek_2mmc())) return -ENODEV; + ret = atmel_ssc_set_audio(0); + if (ret != 0) { + pr_err("Failed to set SSC 0 for audio: %d\n", ret); + return ret; + } + /* * Codec MCLK is supplied by PCK0 - set it up. */ diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 8780c90107fc..d8dc8225576a 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -49,7 +49,7 @@ static int db1200_i2s_startup(struct snd_pcm_substream *substream) int ret; /* WM8731 has its own 12MHz crystal */ - snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, 12000000, SND_SOC_CLOCK_IN); /* codec is bitclock and lrclk master */ diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c new file mode 100644 index 000000000000..01d19e9f53f9 --- /dev/null +++ b/sound/soc/codecs/88pm860x-codec.c @@ -0,0 +1,1486 @@ +/* + * 88pm860x-codec.c -- 88PM860x ALSA SoC Audio Driver + * + * Copyright 2010 Marvell International Ltd. + * Author: Haojian Zhuang + * + * 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 +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "88pm860x-codec.h" + +#define MAX_NAME_LEN 20 +#define REG_CACHE_SIZE 0x40 +#define REG_CACHE_BASE 0xb0 + +/* Status Register 1 (0x01) */ +#define REG_STATUS_1 0x01 +#define MIC_STATUS (1 << 7) +#define HOOK_STATUS (1 << 6) +#define HEADSET_STATUS (1 << 5) + +/* Mic Detection Register (0x37) */ +#define REG_MIC_DET 0x37 +#define CONTINUOUS_POLLING (3 << 1) +#define EN_MIC_DET (1 << 0) +#define MICDET_MASK 0x07 + +/* Headset Detection Register (0x38) */ +#define REG_HS_DET 0x38 +#define EN_HS_DET (1 << 0) + +/* Misc2 Register (0x42) */ +#define REG_MISC2 0x42 +#define AUDIO_PLL (1 << 5) +#define AUDIO_SECTION_RESET (1 << 4) +#define AUDIO_SECTION_ON (1 << 3) + +/* PCM Interface Register 2 (0xb1) */ +#define PCM_INF2_BCLK (1 << 6) /* Bit clock polarity */ +#define PCM_INF2_FS (1 << 5) /* Frame Sync polarity */ +#define PCM_INF2_MASTER (1 << 4) /* Master / Slave */ +#define PCM_INF2_18WL (1 << 3) /* 18 / 16 bits */ +#define PCM_GENERAL_I2S 0 +#define PCM_EXACT_I2S 1 +#define PCM_LEFT_I2S 2 +#define PCM_RIGHT_I2S 3 +#define PCM_SHORT_FS 4 +#define PCM_LONG_FS 5 +#define PCM_MODE_MASK 7 + +/* I2S Interface Register 4 (0xbe) */ +#define I2S_EQU_BYP (1 << 6) + +/* DAC Offset Register (0xcb) */ +#define DAC_MUTE (1 << 7) +#define MUTE_LEFT (1 << 6) +#define MUTE_RIGHT (1 << 2) + +/* ADC Analog Register 1 (0xd0) */ +#define REG_ADC_ANA_1 0xd0 +#define MIC1BIAS_MASK 0x60 + +/* Earpiece/Speaker Control Register 2 (0xda) */ +#define REG_EAR2 0xda +#define RSYNC_CHANGE (1 << 2) + +/* Audio Supplies Register 2 (0xdc) */ +#define REG_SUPPLIES2 0xdc +#define LDO15_READY (1 << 4) +#define LDO15_EN (1 << 3) +#define CPUMP_READY (1 << 2) +#define CPUMP_EN (1 << 1) +#define AUDIO_EN (1 << 0) +#define SUPPLY_MASK (LDO15_EN | CPUMP_EN | AUDIO_EN) + +/* Audio Enable Register 1 (0xdd) */ +#define ADC_MOD_RIGHT (1 << 1) +#define ADC_MOD_LEFT (1 << 0) + +/* Audio Enable Register 2 (0xde) */ +#define ADC_LEFT (1 << 5) +#define ADC_RIGHT (1 << 4) + +/* DAC Enable Register 2 (0xe1) */ +#define DAC_LEFT (1 << 5) +#define DAC_RIGHT (1 << 4) +#define MODULATOR (1 << 3) + +/* Shorts Register (0xeb) */ +#define REG_SHORTS 0xeb +#define CLR_SHORT_LO2 (1 << 7) +#define SHORT_LO2 (1 << 6) +#define CLR_SHORT_LO1 (1 << 5) +#define SHORT_LO1 (1 << 4) +#define CLR_SHORT_HS2 (1 << 3) +#define SHORT_HS2 (1 << 2) +#define CLR_SHORT_HS1 (1 << 1) +#define SHORT_HS1 (1 << 0) + +/* + * This widget should be just after DAC & PGA in DAPM power-on sequence and + * before DAC & PGA in DAPM power-off sequence. + */ +#define PM860X_DAPM_OUTPUT(wname, wevent) \ +{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, \ + .shift = 0, .invert = 0, .kcontrols = NULL, \ + .num_kcontrols = 0, .event = wevent, \ + .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD, } + +struct pm860x_det { + struct snd_soc_jack *hp_jack; + struct snd_soc_jack *mic_jack; + int hp_det; + int mic_det; + int hook_det; + int hs_shrt; + int lo_shrt; +}; + +struct pm860x_priv { + unsigned int sysclk; + unsigned int pcmclk; + unsigned int dir; + unsigned int filter; + struct snd_soc_codec *codec; + struct i2c_client *i2c; + struct pm860x_chip *chip; + struct pm860x_det det; + + int irq[4]; + unsigned char name[4][MAX_NAME_LEN]; + unsigned char reg_cache[REG_CACHE_SIZE]; +}; + +/* -9450dB to 0dB in 150dB steps ( mute instead of -9450dB) */ +static const DECLARE_TLV_DB_SCALE(dpga_tlv, -9450, 150, 1); + +/* -9dB to 0db in 3dB steps */ +static const DECLARE_TLV_DB_SCALE(adc_tlv, -900, 300, 0); + +/* {-23, -17, -13.5, -11, -9, -6, -3, 0}dB */ +static const unsigned int mic_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 0, TLV_DB_SCALE_ITEM(-2300, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-1700, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-1350, 0, 0), + 3, 3, TLV_DB_SCALE_ITEM(-1100, 0, 0), + 4, 7, TLV_DB_SCALE_ITEM(-900, 300, 0), +}; + +/* {0, 0, 0, -6, 0, 6, 12, 18}dB */ +static const unsigned int aux_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(0, 0, 0), + 3, 7, TLV_DB_SCALE_ITEM(-600, 600, 0), +}; + +/* {-16, -13, -10, -7, -5.2, -3,3, -2.2, 0}dB, mute instead of -16dB */ +static const unsigned int out_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 3, TLV_DB_SCALE_ITEM(-1600, 300, 1), + 4, 4, TLV_DB_SCALE_ITEM(-520, 0, 0), + 5, 5, TLV_DB_SCALE_ITEM(-330, 0, 0), + 6, 7, TLV_DB_SCALE_ITEM(-220, 220, 0), +}; + +static const unsigned int st_tlv[] = { + TLV_DB_RANGE_HEAD(8), + 0, 1, TLV_DB_SCALE_ITEM(-12041, 602, 0), + 2, 3, TLV_DB_SCALE_ITEM(-11087, 250, 0), + 4, 5, TLV_DB_SCALE_ITEM(-10643, 158, 0), + 6, 7, TLV_DB_SCALE_ITEM(-10351, 116, 0), + 8, 9, TLV_DB_SCALE_ITEM(-10133, 92, 0), + 10, 13, TLV_DB_SCALE_ITEM(-9958, 70, 0), + 14, 17, TLV_DB_SCALE_ITEM(-9689, 53, 0), + 18, 271, TLV_DB_SCALE_ITEM(-9484, 37, 0), +}; + +/* Sidetone Gain = M * 2^(-5-N) */ +struct st_gain { + unsigned int db; + unsigned int m; + unsigned int n; +}; + +static struct st_gain st_table[] = { + {-12041, 1, 15}, {-11439, 1, 14}, {-11087, 3, 15}, {-10837, 1, 13}, + {-10643, 5, 15}, {-10485, 3, 14}, {-10351, 7, 15}, {-10235, 1, 12}, + {-10133, 9, 15}, {-10041, 5, 14}, { -9958, 11, 15}, { -9883, 3, 13}, + { -9813, 13, 15}, { -9749, 7, 14}, { -9689, 15, 15}, { -9633, 1, 11}, + { -9580, 17, 15}, { -9531, 9, 14}, { -9484, 19, 15}, { -9439, 5, 13}, + { -9397, 21, 15}, { -9356, 11, 14}, { -9318, 23, 15}, { -9281, 3, 12}, + { -9245, 25, 15}, { -9211, 13, 14}, { -9178, 27, 15}, { -9147, 7, 13}, + { -9116, 29, 15}, { -9087, 15, 14}, { -9058, 31, 15}, { -9031, 1, 10}, + { -8978, 17, 14}, { -8929, 9, 13}, { -8882, 19, 14}, { -8837, 5, 12}, + { -8795, 21, 14}, { -8754, 11, 13}, { -8716, 23, 14}, { -8679, 3, 11}, + { -8643, 25, 14}, { -8609, 13, 13}, { -8576, 27, 14}, { -8545, 7, 12}, + { -8514, 29, 14}, { -8485, 15, 13}, { -8456, 31, 14}, { -8429, 1, 9}, + { -8376, 17, 13}, { -8327, 9, 12}, { -8280, 19, 13}, { -8235, 5, 11}, + { -8193, 21, 13}, { -8152, 11, 12}, { -8114, 23, 13}, { -8077, 3, 10}, + { -8041, 25, 13}, { -8007, 13, 12}, { -7974, 27, 13}, { -7943, 7, 11}, + { -7912, 29, 13}, { -7883, 15, 12}, { -7854, 31, 13}, { -7827, 1, 8}, + { -7774, 17, 12}, { -7724, 9, 11}, { -7678, 19, 12}, { -7633, 5, 10}, + { -7591, 21, 12}, { -7550, 11, 11}, { -7512, 23, 12}, { -7475, 3, 9}, + { -7439, 25, 12}, { -7405, 13, 11}, { -7372, 27, 12}, { -7341, 7, 10}, + { -7310, 29, 12}, { -7281, 15, 11}, { -7252, 31, 12}, { -7225, 1, 7}, + { -7172, 17, 11}, { -7122, 9, 10}, { -7075, 19, 11}, { -7031, 5, 9}, + { -6989, 21, 11}, { -6948, 11, 10}, { -6910, 23, 11}, { -6873, 3, 8}, + { -6837, 25, 11}, { -6803, 13, 10}, { -6770, 27, 11}, { -6739, 7, 9}, + { -6708, 29, 11}, { -6679, 15, 10}, { -6650, 31, 11}, { -6623, 1, 6}, + { -6570, 17, 10}, { -6520, 9, 9}, { -6473, 19, 10}, { -6429, 5, 8}, + { -6386, 21, 10}, { -6346, 11, 9}, { -6307, 23, 10}, { -6270, 3, 7}, + { -6235, 25, 10}, { -6201, 13, 9}, { -6168, 27, 10}, { -6137, 7, 8}, + { -6106, 29, 10}, { -6077, 15, 9}, { -6048, 31, 10}, { -6021, 1, 5}, + { -5968, 17, 9}, { -5918, 9, 8}, { -5871, 19, 9}, { -5827, 5, 7}, + { -5784, 21, 9}, { -5744, 11, 8}, { -5705, 23, 9}, { -5668, 3, 6}, + { -5633, 25, 9}, { -5599, 13, 8}, { -5566, 27, 9}, { -5535, 7, 7}, + { -5504, 29, 9}, { -5475, 15, 8}, { -5446, 31, 9}, { -5419, 1, 4}, + { -5366, 17, 8}, { -5316, 9, 7}, { -5269, 19, 8}, { -5225, 5, 6}, + { -5182, 21, 8}, { -5142, 11, 7}, { -5103, 23, 8}, { -5066, 3, 5}, + { -5031, 25, 8}, { -4997, 13, 7}, { -4964, 27, 8}, { -4932, 7, 6}, + { -4902, 29, 8}, { -4873, 15, 7}, { -4844, 31, 8}, { -4816, 1, 3}, + { -4764, 17, 7}, { -4714, 9, 6}, { -4667, 19, 7}, { -4623, 5, 5}, + { -4580, 21, 7}, { -4540, 11, 6}, { -4501, 23, 7}, { -4464, 3, 4}, + { -4429, 25, 7}, { -4395, 13, 6}, { -4362, 27, 7}, { -4330, 7, 5}, + { -4300, 29, 7}, { -4270, 15, 6}, { -4242, 31, 7}, { -4214, 1, 2}, + { -4162, 17, 6}, { -4112, 9, 5}, { -4065, 19, 6}, { -4021, 5, 4}, + { -3978, 21, 6}, { -3938, 11, 5}, { -3899, 23, 6}, { -3862, 3, 3}, + { -3827, 25, 6}, { -3793, 13, 5}, { -3760, 27, 6}, { -3728, 7, 4}, + { -3698, 29, 6}, { -3668, 15, 5}, { -3640, 31, 6}, { -3612, 1, 1}, + { -3560, 17, 5}, { -3510, 9, 4}, { -3463, 19, 5}, { -3419, 5, 3}, + { -3376, 21, 5}, { -3336, 11, 4}, { -3297, 23, 5}, { -3260, 3, 2}, + { -3225, 25, 5}, { -3191, 13, 4}, { -3158, 27, 5}, { -3126, 7, 3}, + { -3096, 29, 5}, { -3066, 15, 4}, { -3038, 31, 5}, { -3010, 1, 0}, + { -2958, 17, 4}, { -2908, 9, 3}, { -2861, 19, 4}, { -2816, 5, 2}, + { -2774, 21, 4}, { -2734, 11, 3}, { -2695, 23, 4}, { -2658, 3, 1}, + { -2623, 25, 4}, { -2589, 13, 3}, { -2556, 27, 4}, { -2524, 7, 2}, + { -2494, 29, 4}, { -2464, 15, 3}, { -2436, 31, 4}, { -2408, 2, 0}, + { -2356, 17, 3}, { -2306, 9, 2}, { -2259, 19, 3}, { -2214, 5, 1}, + { -2172, 21, 3}, { -2132, 11, 2}, { -2093, 23, 3}, { -2056, 3, 0}, + { -2021, 25, 3}, { -1987, 13, 2}, { -1954, 27, 3}, { -1922, 7, 1}, + { -1892, 29, 3}, { -1862, 15, 2}, { -1834, 31, 3}, { -1806, 4, 0}, + { -1754, 17, 2}, { -1704, 9, 1}, { -1657, 19, 2}, { -1612, 5, 0}, + { -1570, 21, 2}, { -1530, 11, 1}, { -1491, 23, 2}, { -1454, 6, 0}, + { -1419, 25, 2}, { -1384, 13, 1}, { -1352, 27, 2}, { -1320, 7, 0}, + { -1290, 29, 2}, { -1260, 15, 1}, { -1232, 31, 2}, { -1204, 8, 0}, + { -1151, 17, 1}, { -1102, 9, 0}, { -1055, 19, 1}, { -1010, 10, 0}, + { -968, 21, 1}, { -928, 11, 0}, { -889, 23, 1}, { -852, 12, 0}, + { -816, 25, 1}, { -782, 13, 0}, { -750, 27, 1}, { -718, 14, 0}, + { -688, 29, 1}, { -658, 15, 0}, { -630, 31, 1}, { -602, 16, 0}, + { -549, 17, 0}, { -500, 18, 0}, { -453, 19, 0}, { -408, 20, 0}, + { -366, 21, 0}, { -325, 22, 0}, { -287, 23, 0}, { -250, 24, 0}, + { -214, 25, 0}, { -180, 26, 0}, { -148, 27, 0}, { -116, 28, 0}, + { -86, 29, 0}, { -56, 30, 0}, { -28, 31, 0}, { 0, 0, 0}, +}; + +static int pm860x_volatile(unsigned int reg) +{ + BUG_ON(reg >= REG_CACHE_SIZE); + + switch (reg) { + case PM860X_AUDIO_SUPPLIES_2: + return 1; + } + + return 0; +} + +static unsigned int pm860x_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + unsigned char *cache = codec->reg_cache; + + BUG_ON(reg >= REG_CACHE_SIZE); + + if (pm860x_volatile(reg)) + return cache[reg]; + + reg += REG_CACHE_BASE; + + return pm860x_reg_read(codec->control_data, reg); +} + +static int pm860x_write_reg_cache(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + unsigned char *cache = codec->reg_cache; + + BUG_ON(reg >= REG_CACHE_SIZE); + + if (!pm860x_volatile(reg)) + cache[reg] = (unsigned char)value; + + reg += REG_CACHE_BASE; + + return pm860x_reg_write(codec->control_data, reg, value); +} + +static int snd_soc_get_volsw_2r_st(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + int val[2], val2[2], i; + + val[0] = snd_soc_read(codec, reg) & 0x3f; + val[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT) >> 4) & 0xf; + val2[0] = snd_soc_read(codec, reg2) & 0x3f; + val2[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT)) & 0xf; + + for (i = 0; i < ARRAY_SIZE(st_table); i++) { + if ((st_table[i].m == val[0]) && (st_table[i].n == val[1])) + ucontrol->value.integer.value[0] = i; + if ((st_table[i].m == val2[0]) && (st_table[i].n == val2[1])) + ucontrol->value.integer.value[1] = i; + } + return 0; +} + +static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + int err; + unsigned int val, val2; + + val = ucontrol->value.integer.value[0]; + val2 = ucontrol->value.integer.value[1]; + + err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m); + if (err < 0) + return err; + err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0xf0, + st_table[val].n << 4); + if (err < 0) + return err; + + err = snd_soc_update_bits(codec, reg2, 0x3f, st_table[val2].m); + if (err < 0) + return err; + err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0x0f, + st_table[val2].n); + return err; +} + +static int snd_soc_get_volsw_2r_out(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + int max = mc->max, val, val2; + unsigned int mask = (1 << fls(max)) - 1; + + val = snd_soc_read(codec, reg) >> shift; + val2 = snd_soc_read(codec, reg2) >> shift; + ucontrol->value.integer.value[0] = (max - val) & mask; + ucontrol->value.integer.value[1] = (max - val2) & mask; + + return 0; +} + +static int snd_soc_put_volsw_2r_out(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + unsigned int shift = mc->shift; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + int err; + unsigned int val, val2, val_mask; + + val_mask = mask << shift; + val = ((max - ucontrol->value.integer.value[0]) & mask); + val2 = ((max - ucontrol->value.integer.value[1]) & mask); + + val = val << shift; + val2 = val2 << shift; + + err = snd_soc_update_bits(codec, reg, val_mask, val); + if (err < 0) + return err; + + err = snd_soc_update_bits(codec, reg2, val_mask, val2); + return err; +} + +/* DAPM Widget Events */ +/* + * A lot registers are belong to RSYNC domain. It requires enabling RSYNC bit + * after updating these registers. Otherwise, these updated registers won't + * be effective. + */ +static int pm860x_rsync_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + /* + * In order to avoid current on the load, mute power-on and power-off + * should be transients. + * Unmute by DAC_MUTE. It should be unmuted when DAPM sequence is + * finished. + */ + snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, 0); + snd_soc_update_bits(codec, PM860X_EAR_CTRL_2, + RSYNC_CHANGE, RSYNC_CHANGE); + return 0; +} + +static int pm860x_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int dac = 0; + int data; + + if (!strcmp(w->name, "Left DAC")) + dac = DAC_LEFT; + if (!strcmp(w->name, "Right DAC")) + dac = DAC_RIGHT; + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (dac) { + /* Auto mute in power-on sequence. */ + dac |= MODULATOR; + snd_soc_update_bits(codec, PM860X_DAC_OFFSET, + DAC_MUTE, DAC_MUTE); + snd_soc_update_bits(codec, PM860X_EAR_CTRL_2, + RSYNC_CHANGE, RSYNC_CHANGE); + /* update dac */ + snd_soc_update_bits(codec, PM860X_DAC_EN_2, + dac, dac); + } + break; + case SND_SOC_DAPM_PRE_PMD: + if (dac) { + /* Auto mute in power-off sequence. */ + snd_soc_update_bits(codec, PM860X_DAC_OFFSET, + DAC_MUTE, DAC_MUTE); + snd_soc_update_bits(codec, PM860X_EAR_CTRL_2, + RSYNC_CHANGE, RSYNC_CHANGE); + /* update dac */ + data = snd_soc_read(codec, PM860X_DAC_EN_2); + data &= ~dac; + if (!(data & (DAC_LEFT | DAC_RIGHT))) + data &= ~MODULATOR; + snd_soc_write(codec, PM860X_DAC_EN_2, data); + } + break; + } + return 0; +} + +static const char *pm860x_opamp_texts[] = {"-50%", "-25%", "0%", "75%"}; + +static const char *pm860x_pa_texts[] = {"-33%", "0%", "33%", "66%"}; + +static const struct soc_enum pm860x_hs1_opamp_enum = + SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts); + +static const struct soc_enum pm860x_hs2_opamp_enum = + SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 5, 4, pm860x_opamp_texts); + +static const struct soc_enum pm860x_hs1_pa_enum = + SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 3, 4, pm860x_pa_texts); + +static const struct soc_enum pm860x_hs2_pa_enum = + SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 3, 4, pm860x_pa_texts); + +static const struct soc_enum pm860x_lo1_opamp_enum = + SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 5, 4, pm860x_opamp_texts); + +static const struct soc_enum pm860x_lo2_opamp_enum = + SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 5, 4, pm860x_opamp_texts); + +static const struct soc_enum pm860x_lo1_pa_enum = + SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 3, 4, pm860x_pa_texts); + +static const struct soc_enum pm860x_lo2_pa_enum = + SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 3, 4, pm860x_pa_texts); + +static const struct soc_enum pm860x_spk_pa_enum = + SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 5, 4, pm860x_pa_texts); + +static const struct soc_enum pm860x_ear_pa_enum = + SOC_ENUM_SINGLE(PM860X_EAR_CTRL_2, 0, 4, pm860x_pa_texts); + +static const struct soc_enum pm860x_spk_ear_opamp_enum = + SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 3, 4, pm860x_opamp_texts); + +static const struct snd_kcontrol_new pm860x_snd_controls[] = { + SOC_DOUBLE_R_TLV("ADC Capture Volume", PM860X_ADC_ANA_2, + PM860X_ADC_ANA_3, 6, 3, 0, adc_tlv), + SOC_DOUBLE_TLV("AUX Capture Volume", PM860X_ADC_ANA_3, 0, 3, 7, 0, + aux_tlv), + SOC_SINGLE_TLV("MIC1 Capture Volume", PM860X_ADC_ANA_2, 0, 7, 0, + mic_tlv), + SOC_SINGLE_TLV("MIC3 Capture Volume", PM860X_ADC_ANA_2, 3, 7, 0, + mic_tlv), + SOC_DOUBLE_R_EXT_TLV("Sidetone Volume", PM860X_SIDETONE_L_GAIN, + PM860X_SIDETONE_R_GAIN, 0, ARRAY_SIZE(st_table)-1, + 0, snd_soc_get_volsw_2r_st, + snd_soc_put_volsw_2r_st, st_tlv), + SOC_SINGLE_TLV("Speaker Playback Volume", PM860X_EAR_CTRL_1, + 0, 7, 0, out_tlv), + SOC_DOUBLE_R_TLV("Line Playback Volume", PM860X_LO1_CTRL, + PM860X_LO2_CTRL, 0, 7, 0, out_tlv), + SOC_DOUBLE_R_TLV("Headset Playback Volume", PM860X_HS1_CTRL, + PM860X_HS2_CTRL, 0, 7, 0, out_tlv), + SOC_DOUBLE_R_EXT_TLV("Hifi Left Playback Volume", + PM860X_HIFIL_GAIN_LEFT, + PM860X_HIFIL_GAIN_RIGHT, 0, 63, 0, + snd_soc_get_volsw_2r_out, + snd_soc_put_volsw_2r_out, dpga_tlv), + SOC_DOUBLE_R_EXT_TLV("Hifi Right Playback Volume", + PM860X_HIFIR_GAIN_LEFT, + PM860X_HIFIR_GAIN_RIGHT, 0, 63, 0, + snd_soc_get_volsw_2r_out, + snd_soc_put_volsw_2r_out, dpga_tlv), + SOC_DOUBLE_R_EXT_TLV("Lofi Playback Volume", PM860X_LOFI_GAIN_LEFT, + PM860X_LOFI_GAIN_RIGHT, 0, 63, 0, + snd_soc_get_volsw_2r_out, + snd_soc_put_volsw_2r_out, dpga_tlv), + SOC_ENUM("Headset1 Operational Amplifier Current", + pm860x_hs1_opamp_enum), + SOC_ENUM("Headset2 Operational Amplifier Current", + pm860x_hs2_opamp_enum), + SOC_ENUM("Headset1 Amplifier Current", pm860x_hs1_pa_enum), + SOC_ENUM("Headset2 Amplifier Current", pm860x_hs2_pa_enum), + SOC_ENUM("Lineout1 Operational Amplifier Current", + pm860x_lo1_opamp_enum), + SOC_ENUM("Lineout2 Operational Amplifier Current", + pm860x_lo2_opamp_enum), + SOC_ENUM("Lineout1 Amplifier Current", pm860x_lo1_pa_enum), + SOC_ENUM("Lineout2 Amplifier Current", pm860x_lo2_pa_enum), + SOC_ENUM("Speaker Operational Amplifier Current", + pm860x_spk_ear_opamp_enum), + SOC_ENUM("Speaker Amplifier Current", pm860x_spk_pa_enum), + SOC_ENUM("Earpiece Amplifier Current", pm860x_ear_pa_enum), +}; + +/* + * DAPM Controls + */ + +/* PCM Switch / PCM Interface */ +static const struct snd_kcontrol_new pcm_switch_controls = + SOC_DAPM_SINGLE("Switch", PM860X_ADC_EN_2, 0, 1, 0); + +/* AUX1 Switch */ +static const struct snd_kcontrol_new aux1_switch_controls = + SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 4, 1, 0); + +/* AUX2 Switch */ +static const struct snd_kcontrol_new aux2_switch_controls = + SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 5, 1, 0); + +/* Left Ex. PA Switch */ +static const struct snd_kcontrol_new lepa_switch_controls = + SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 2, 1, 0); + +/* Right Ex. PA Switch */ +static const struct snd_kcontrol_new repa_switch_controls = + SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 1, 1, 0); + +/* PCM Mux / Mux7 */ +static const char *aif1_text[] = { + "PCM L", "PCM R", +}; + +static const struct soc_enum aif1_enum = + SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 6, 2, aif1_text); + +static const struct snd_kcontrol_new aif1_mux = + SOC_DAPM_ENUM("PCM Mux", aif1_enum); + +/* I2S Mux / Mux9 */ +static const char *i2s_din_text[] = { + "DIN", "DIN1", +}; + +static const struct soc_enum i2s_din_enum = + SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 1, 2, i2s_din_text); + +static const struct snd_kcontrol_new i2s_din_mux = + SOC_DAPM_ENUM("I2S DIN Mux", i2s_din_enum); + +/* I2S Mic Mux / Mux8 */ +static const char *i2s_mic_text[] = { + "Ex PA", "ADC", +}; + +static const struct soc_enum i2s_mic_enum = + SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 4, 2, i2s_mic_text); + +static const struct snd_kcontrol_new i2s_mic_mux = + SOC_DAPM_ENUM("I2S Mic Mux", i2s_mic_enum); + +/* ADCL Mux / Mux2 */ +static const char *adcl_text[] = { + "ADCR", "ADCL", +}; + +static const struct soc_enum adcl_enum = + SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 4, 2, adcl_text); + +static const struct snd_kcontrol_new adcl_mux = + SOC_DAPM_ENUM("ADC Left Mux", adcl_enum); + +/* ADCR Mux / Mux3 */ +static const char *adcr_text[] = { + "ADCL", "ADCR", +}; + +static const struct soc_enum adcr_enum = + SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 2, 2, adcr_text); + +static const struct snd_kcontrol_new adcr_mux = + SOC_DAPM_ENUM("ADC Right Mux", adcr_enum); + +/* ADCR EC Mux / Mux6 */ +static const char *adcr_ec_text[] = { + "ADCR", "EC", +}; + +static const struct soc_enum adcr_ec_enum = + SOC_ENUM_SINGLE(PM860X_ADC_EN_2, 3, 2, adcr_ec_text); + +static const struct snd_kcontrol_new adcr_ec_mux = + SOC_DAPM_ENUM("ADCR EC Mux", adcr_ec_enum); + +/* EC Mux / Mux4 */ +static const char *ec_text[] = { + "Left", "Right", "Left + Right", +}; + +static const struct soc_enum ec_enum = + SOC_ENUM_SINGLE(PM860X_EC_PATH, 1, 3, ec_text); + +static const struct snd_kcontrol_new ec_mux = + SOC_DAPM_ENUM("EC Mux", ec_enum); + +static const char *dac_text[] = { + "No input", "Right", "Left", "No input", +}; + +/* DAC Headset 1 Mux / Mux10 */ +static const struct soc_enum dac_hs1_enum = + SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 0, 4, dac_text); + +static const struct snd_kcontrol_new dac_hs1_mux = + SOC_DAPM_ENUM("DAC HS1 Mux", dac_hs1_enum); + +/* DAC Headset 2 Mux / Mux11 */ +static const struct soc_enum dac_hs2_enum = + SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 2, 4, dac_text); + +static const struct snd_kcontrol_new dac_hs2_mux = + SOC_DAPM_ENUM("DAC HS2 Mux", dac_hs2_enum); + +/* DAC Lineout 1 Mux / Mux12 */ +static const struct soc_enum dac_lo1_enum = + SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 4, 4, dac_text); + +static const struct snd_kcontrol_new dac_lo1_mux = + SOC_DAPM_ENUM("DAC LO1 Mux", dac_lo1_enum); + +/* DAC Lineout 2 Mux / Mux13 */ +static const struct soc_enum dac_lo2_enum = + SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 6, 4, dac_text); + +static const struct snd_kcontrol_new dac_lo2_mux = + SOC_DAPM_ENUM("DAC LO2 Mux", dac_lo2_enum); + +/* DAC Spearker Earphone Mux / Mux14 */ +static const struct soc_enum dac_spk_ear_enum = + SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_2, 0, 4, dac_text); + +static const struct snd_kcontrol_new dac_spk_ear_mux = + SOC_DAPM_ENUM("DAC SP Mux", dac_spk_ear_enum); + +/* Headset 1 Mux / Mux15 */ +static const char *in_text[] = { + "Digital", "Analog", +}; + +static const struct soc_enum hs1_enum = + SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 0, 2, in_text); + +static const struct snd_kcontrol_new hs1_mux = + SOC_DAPM_ENUM("Headset1 Mux", hs1_enum); + +/* Headset 2 Mux / Mux16 */ +static const struct soc_enum hs2_enum = + SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 1, 2, in_text); + +static const struct snd_kcontrol_new hs2_mux = + SOC_DAPM_ENUM("Headset2 Mux", hs2_enum); + +/* Lineout 1 Mux / Mux17 */ +static const struct soc_enum lo1_enum = + SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 2, 2, in_text); + +static const struct snd_kcontrol_new lo1_mux = + SOC_DAPM_ENUM("Lineout1 Mux", lo1_enum); + +/* Lineout 2 Mux / Mux18 */ +static const struct soc_enum lo2_enum = + SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 3, 2, in_text); + +static const struct snd_kcontrol_new lo2_mux = + SOC_DAPM_ENUM("Lineout2 Mux", lo2_enum); + +/* Speaker Earpiece Demux */ +static const char *spk_text[] = { + "Earpiece", "Speaker", +}; + +static const struct soc_enum spk_enum = + SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 6, 2, spk_text); + +static const struct snd_kcontrol_new spk_demux = + SOC_DAPM_ENUM("Speaker Earpiece Demux", spk_enum); + +/* MIC Mux / Mux1 */ +static const char *mic_text[] = { + "Mic 1", "Mic 2", +}; + +static const struct soc_enum mic_enum = + SOC_ENUM_SINGLE(PM860X_ADC_ANA_4, 4, 2, mic_text); + +static const struct snd_kcontrol_new mic_mux = + SOC_DAPM_ENUM("MIC Mux", mic_enum); + +static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("PCM SDI", "PCM Playback", 0, + PM860X_ADC_EN_2, 0, 0), + SND_SOC_DAPM_AIF_OUT("PCM SDO", "PCM Capture", 0, + PM860X_PCM_IFACE_3, 1, 1), + + + SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0, + PM860X_DAC_EN_2, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0, + PM860X_DAC_EN_2, 0, 0), + SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0, + PM860X_I2S_IFACE_3, 5, 1), + SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux), + SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux), + SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux), + SND_SOC_DAPM_MUX("EC Mux", SND_SOC_NOPM, 0, 0, &ec_mux), + SND_SOC_DAPM_MUX("ADCR EC Mux", SND_SOC_NOPM, 0, 0, &adcr_ec_mux), + SND_SOC_DAPM_SWITCH("Left EPA", SND_SOC_NOPM, 0, 0, + &lepa_switch_controls), + SND_SOC_DAPM_SWITCH("Right EPA", SND_SOC_NOPM, 0, 0, + &repa_switch_controls), + + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "Left ADC MOD", PM860X_ADC_EN_1, + 0, 1, 1, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "Right ADC MOD", PM860X_ADC_EN_1, + 1, 1, 1, 0), + SND_SOC_DAPM_ADC("Left ADC", NULL, PM860X_ADC_EN_2, 5, 0), + SND_SOC_DAPM_ADC("Right ADC", NULL, PM860X_ADC_EN_2, 4, 0), + + SND_SOC_DAPM_SWITCH("AUX1 Switch", SND_SOC_NOPM, 0, 0, + &aux1_switch_controls), + SND_SOC_DAPM_SWITCH("AUX2 Switch", SND_SOC_NOPM, 0, 0, + &aux2_switch_controls), + + SND_SOC_DAPM_MUX("MIC Mux", SND_SOC_NOPM, 0, 0, &mic_mux), + SND_SOC_DAPM_MICBIAS("Mic1 Bias", PM860X_ADC_ANA_1, 2, 0), + SND_SOC_DAPM_MICBIAS("Mic3 Bias", PM860X_ADC_ANA_1, 7, 0), + SND_SOC_DAPM_PGA("MIC1 Volume", PM860X_ADC_EN_1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC3 Volume", PM860X_ADC_EN_1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX1 Volume", PM860X_ADC_EN_1, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX2 Volume", PM860X_ADC_EN_1, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Sidetone PGA", PM860X_ADC_EN_2, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("Lofi PGA", PM860X_ADC_EN_2, 2, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("AUX1"), + SND_SOC_DAPM_INPUT("AUX2"), + SND_SOC_DAPM_INPUT("MIC1P"), + SND_SOC_DAPM_INPUT("MIC1N"), + SND_SOC_DAPM_INPUT("MIC2P"), + SND_SOC_DAPM_INPUT("MIC2N"), + SND_SOC_DAPM_INPUT("MIC3P"), + SND_SOC_DAPM_INPUT("MIC3N"), + + SND_SOC_DAPM_DAC_E("Left DAC", NULL, SND_SOC_NOPM, 0, 0, + pm860x_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("Right DAC", NULL, SND_SOC_NOPM, 0, 0, + pm860x_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_MUX("I2S DIN Mux", SND_SOC_NOPM, 0, 0, &i2s_din_mux), + SND_SOC_DAPM_MUX("DAC HS1 Mux", SND_SOC_NOPM, 0, 0, &dac_hs1_mux), + SND_SOC_DAPM_MUX("DAC HS2 Mux", SND_SOC_NOPM, 0, 0, &dac_hs2_mux), + SND_SOC_DAPM_MUX("DAC LO1 Mux", SND_SOC_NOPM, 0, 0, &dac_lo1_mux), + SND_SOC_DAPM_MUX("DAC LO2 Mux", SND_SOC_NOPM, 0, 0, &dac_lo2_mux), + SND_SOC_DAPM_MUX("DAC SP Mux", SND_SOC_NOPM, 0, 0, &dac_spk_ear_mux), + SND_SOC_DAPM_MUX("Headset1 Mux", SND_SOC_NOPM, 0, 0, &hs1_mux), + SND_SOC_DAPM_MUX("Headset2 Mux", SND_SOC_NOPM, 0, 0, &hs2_mux), + SND_SOC_DAPM_MUX("Lineout1 Mux", SND_SOC_NOPM, 0, 0, &lo1_mux), + SND_SOC_DAPM_MUX("Lineout2 Mux", SND_SOC_NOPM, 0, 0, &lo2_mux), + SND_SOC_DAPM_MUX("Speaker Earpiece Demux", SND_SOC_NOPM, 0, 0, + &spk_demux), + + + SND_SOC_DAPM_PGA("Headset1 PGA", PM860X_DAC_EN_1, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Headset2 PGA", PM860X_DAC_EN_1, 1, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("HS1"), + SND_SOC_DAPM_OUTPUT("HS2"), + SND_SOC_DAPM_PGA("Lineout1 PGA", PM860X_DAC_EN_1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("Lineout2 PGA", PM860X_DAC_EN_1, 3, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("LINEOUT1"), + SND_SOC_DAPM_OUTPUT("LINEOUT2"), + SND_SOC_DAPM_PGA("Earpiece PGA", PM860X_DAC_EN_1, 4, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("EARP"), + SND_SOC_DAPM_OUTPUT("EARN"), + SND_SOC_DAPM_PGA("Speaker PGA", PM860X_DAC_EN_1, 5, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("LSP"), + SND_SOC_DAPM_OUTPUT("LSN"), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "VCODEC", PM860X_AUDIO_SUPPLIES_2, + 0, SUPPLY_MASK, SUPPLY_MASK, 0), + + PM860X_DAPM_OUTPUT("RSYNC", pm860x_rsync_event), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* supply */ + {"Left DAC", NULL, "VCODEC"}, + {"Right DAC", NULL, "VCODEC"}, + {"Left ADC", NULL, "VCODEC"}, + {"Right ADC", NULL, "VCODEC"}, + {"Left ADC", NULL, "Left ADC MOD"}, + {"Right ADC", NULL, "Right ADC MOD"}, + + /* PCM/AIF1 Inputs */ + {"PCM SDO", NULL, "ADC Left Mux"}, + {"PCM SDO", NULL, "ADCR EC Mux"}, + + /* PCM/AFI2 Outputs */ + {"Lofi PGA", NULL, "PCM SDI"}, + {"Lofi PGA", NULL, "Sidetone PGA"}, + {"Left DAC", NULL, "Lofi PGA"}, + {"Right DAC", NULL, "Lofi PGA"}, + + /* I2S/AIF2 Inputs */ + {"MIC Mux", "Mic 1", "MIC1P"}, + {"MIC Mux", "Mic 1", "MIC1N"}, + {"MIC Mux", "Mic 2", "MIC2P"}, + {"MIC Mux", "Mic 2", "MIC2N"}, + {"MIC1 Volume", NULL, "MIC Mux"}, + {"MIC3 Volume", NULL, "MIC3P"}, + {"MIC3 Volume", NULL, "MIC3N"}, + {"Left ADC", NULL, "MIC1 Volume"}, + {"Right ADC", NULL, "MIC3 Volume"}, + {"ADC Left Mux", "ADCR", "Right ADC"}, + {"ADC Left Mux", "ADCL", "Left ADC"}, + {"ADC Right Mux", "ADCL", "Left ADC"}, + {"ADC Right Mux", "ADCR", "Right ADC"}, + {"Left EPA", "Switch", "Left DAC"}, + {"Right EPA", "Switch", "Right DAC"}, + {"EC Mux", "Left", "Left DAC"}, + {"EC Mux", "Right", "Right DAC"}, + {"EC Mux", "Left + Right", "Left DAC"}, + {"EC Mux", "Left + Right", "Right DAC"}, + {"ADCR EC Mux", "ADCR", "ADC Right Mux"}, + {"ADCR EC Mux", "EC", "EC Mux"}, + {"I2S Mic Mux", "Ex PA", "Left EPA"}, + {"I2S Mic Mux", "Ex PA", "Right EPA"}, + {"I2S Mic Mux", "ADC", "ADC Left Mux"}, + {"I2S Mic Mux", "ADC", "ADCR EC Mux"}, + {"I2S DOUT", NULL, "I2S Mic Mux"}, + + /* I2S/AIF2 Outputs */ + {"I2S DIN Mux", "DIN", "I2S DIN"}, + {"I2S DIN Mux", "DIN1", "I2S DIN1"}, + {"Left DAC", NULL, "I2S DIN Mux"}, + {"Right DAC", NULL, "I2S DIN Mux"}, + {"DAC HS1 Mux", "Left", "Left DAC"}, + {"DAC HS1 Mux", "Right", "Right DAC"}, + {"DAC HS2 Mux", "Left", "Left DAC"}, + {"DAC HS2 Mux", "Right", "Right DAC"}, + {"DAC LO1 Mux", "Left", "Left DAC"}, + {"DAC LO1 Mux", "Right", "Right DAC"}, + {"DAC LO2 Mux", "Left", "Left DAC"}, + {"DAC LO2 Mux", "Right", "Right DAC"}, + {"Headset1 Mux", "Digital", "DAC HS1 Mux"}, + {"Headset2 Mux", "Digital", "DAC HS2 Mux"}, + {"Lineout1 Mux", "Digital", "DAC LO1 Mux"}, + {"Lineout2 Mux", "Digital", "DAC LO2 Mux"}, + {"Headset1 PGA", NULL, "Headset1 Mux"}, + {"Headset2 PGA", NULL, "Headset2 Mux"}, + {"Lineout1 PGA", NULL, "Lineout1 Mux"}, + {"Lineout2 PGA", NULL, "Lineout2 Mux"}, + {"DAC SP Mux", "Left", "Left DAC"}, + {"DAC SP Mux", "Right", "Right DAC"}, + {"Speaker Earpiece Demux", "Speaker", "DAC SP Mux"}, + {"Speaker PGA", NULL, "Speaker Earpiece Demux"}, + {"Earpiece PGA", NULL, "Speaker Earpiece Demux"}, + + {"RSYNC", NULL, "Headset1 PGA"}, + {"RSYNC", NULL, "Headset2 PGA"}, + {"RSYNC", NULL, "Lineout1 PGA"}, + {"RSYNC", NULL, "Lineout2 PGA"}, + {"RSYNC", NULL, "Speaker PGA"}, + {"RSYNC", NULL, "Speaker PGA"}, + {"RSYNC", NULL, "Earpiece PGA"}, + {"RSYNC", NULL, "Earpiece PGA"}, + + {"HS1", NULL, "RSYNC"}, + {"HS2", NULL, "RSYNC"}, + {"LINEOUT1", NULL, "RSYNC"}, + {"LINEOUT2", NULL, "RSYNC"}, + {"LSP", NULL, "RSYNC"}, + {"LSN", NULL, "RSYNC"}, + {"EARP", NULL, "RSYNC"}, + {"EARN", NULL, "RSYNC"}, +}; + +/* + * Use MUTE_LEFT & MUTE_RIGHT to implement digital mute. + * These bits can also be used to mute. + */ +static int pm860x_digital_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int data = 0, mask = MUTE_LEFT | MUTE_RIGHT; + + if (mute) + data = mask; + snd_soc_update_bits(codec, PM860X_DAC_OFFSET, mask, data); + snd_soc_update_bits(codec, PM860X_EAR_CTRL_2, + RSYNC_CHANGE, RSYNC_CHANGE); + return 0; +} + +static int pm860x_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned char inf = 0, mask = 0; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + inf &= ~PCM_INF2_18WL; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + inf |= PCM_INF2_18WL; + break; + default: + return -EINVAL; + } + mask |= PCM_INF2_18WL; + snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf); + + /* sample rate */ + switch (params_rate(params)) { + case 8000: + inf = 0; + break; + case 16000: + inf = 3; + break; + case 32000: + inf = 6; + break; + case 48000: + inf = 8; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, PM860X_PCM_RATE, 0x0f, inf); + + return 0; +} + +static int pm860x_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); + unsigned char inf = 0, mask = 0; + int ret = -EINVAL; + + mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + if (pm860x->dir == PM860X_CLK_DIR_OUT) { + inf |= PCM_INF2_MASTER; + ret = 0; + } + break; + case SND_SOC_DAIFMT_CBS_CFS: + if (pm860x->dir == PM860X_CLK_DIR_IN) { + inf &= ~PCM_INF2_MASTER; + ret = 0; + } + break; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + inf |= PCM_EXACT_I2S; + ret = 0; + break; + } + mask |= PCM_MODE_MASK; + if (ret) + return ret; + snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf); + return 0; +} + +static int pm860x_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); + + if (dir == PM860X_CLK_DIR_OUT) + pm860x->dir = PM860X_CLK_DIR_OUT; + else { + pm860x->dir = PM860X_CLK_DIR_IN; + return -EINVAL; + } + + return 0; +} + +static int pm860x_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned char inf; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + inf = 0; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + inf = PCM_INF2_18WL; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, PCM_INF2_18WL, inf); + + /* sample rate */ + switch (params_rate(params)) { + case 8000: + inf = 0; + break; + case 11025: + inf = 1; + break; + case 16000: + inf = 3; + break; + case 22050: + inf = 4; + break; + case 32000: + inf = 6; + break; + case 44100: + inf = 7; + break; + case 48000: + inf = 8; + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, PM860X_I2S_IFACE_4, 0xf, inf); + + return 0; +} + +static int pm860x_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); + unsigned char inf = 0, mask = 0; + + mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + if (pm860x->dir == PM860X_CLK_DIR_OUT) + inf |= PCM_INF2_MASTER; + else + return -EINVAL; + break; + case SND_SOC_DAIFMT_CBS_CFS: + if (pm860x->dir == PM860X_CLK_DIR_IN) + inf &= ~PCM_INF2_MASTER; + else + return -EINVAL; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + inf |= PCM_EXACT_I2S; + break; + default: + return -EINVAL; + } + mask |= PCM_MODE_MASK; + snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, mask, inf); + return 0; +} + +static int pm860x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + int data; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Enable Audio PLL & Audio section */ + data = AUDIO_PLL | AUDIO_SECTION_RESET + | AUDIO_SECTION_ON; + pm860x_reg_write(codec->control_data, REG_MISC2, data); + } + break; + + case SND_SOC_BIAS_OFF: + data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; + pm860x_set_bits(codec->control_data, REG_MISC2, data, 0); + break; + } + codec->bias_level = level; + return 0; +} + +static struct snd_soc_dai_ops pm860x_pcm_dai_ops = { + .digital_mute = pm860x_digital_mute, + .hw_params = pm860x_pcm_hw_params, + .set_fmt = pm860x_pcm_set_dai_fmt, + .set_sysclk = pm860x_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops pm860x_i2s_dai_ops = { + .digital_mute = pm860x_digital_mute, + .hw_params = pm860x_i2s_hw_params, + .set_fmt = pm860x_i2s_set_dai_fmt, + .set_sysclk = pm860x_set_dai_sysclk, +}; + +#define PM860X_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000) + +static struct snd_soc_dai_driver pm860x_dai[] = { + { + /* DAI PCM */ + .name = "88pm860x-pcm", + .id = 1, + .playback = { + .stream_name = "PCM Playback", + .channels_min = 2, + .channels_max = 2, + .rates = PM860X_RATES, + .formats = SNDRV_PCM_FORMAT_S16_LE | \ + SNDRV_PCM_FORMAT_S18_3LE, + }, + .capture = { + .stream_name = "PCM Capture", + .channels_min = 2, + .channels_max = 2, + .rates = PM860X_RATES, + .formats = SNDRV_PCM_FORMAT_S16_LE | \ + SNDRV_PCM_FORMAT_S18_3LE, + }, + .ops = &pm860x_pcm_dai_ops, + }, { + /* DAI I2S */ + .name = "88pm860x-i2s", + .id = 2, + .playback = { + .stream_name = "I2S Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FORMAT_S16_LE | \ + SNDRV_PCM_FORMAT_S18_3LE, + }, + .capture = { + .stream_name = "I2S Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FORMAT_S16_LE | \ + SNDRV_PCM_FORMAT_S18_3LE, + }, + .ops = &pm860x_i2s_dai_ops, + }, +}; + +static irqreturn_t pm860x_codec_handler(int irq, void *data) +{ + struct pm860x_priv *pm860x = data; + int status, shrt, report = 0, mic_report = 0; + int mask; + + status = pm860x_reg_read(pm860x->i2c, REG_STATUS_1); + shrt = pm860x_reg_read(pm860x->i2c, REG_SHORTS); + mask = pm860x->det.hs_shrt | pm860x->det.hook_det | pm860x->det.lo_shrt + | pm860x->det.hp_det; + + if ((pm860x->det.hp_det & SND_JACK_HEADPHONE) + && (status & HEADSET_STATUS)) + report |= SND_JACK_HEADPHONE; + + if ((pm860x->det.mic_det & SND_JACK_MICROPHONE) + && (status & MIC_STATUS)) + mic_report |= SND_JACK_MICROPHONE; + + if (pm860x->det.hs_shrt && (shrt & (SHORT_HS1 | SHORT_HS2))) + report |= pm860x->det.hs_shrt; + + if (pm860x->det.hook_det && (status & HOOK_STATUS)) + report |= pm860x->det.hook_det; + + if (pm860x->det.lo_shrt && (shrt & (SHORT_LO1 | SHORT_LO2))) + report |= pm860x->det.lo_shrt; + + if (report) + snd_soc_jack_report(pm860x->det.hp_jack, report, mask); + if (mic_report) + snd_soc_jack_report(pm860x->det.mic_jack, SND_JACK_MICROPHONE, + SND_JACK_MICROPHONE); + + dev_dbg(pm860x->codec->dev, "headphone report:0x%x, mask:%x\n", + report, mask); + dev_dbg(pm860x->codec->dev, "microphone report:0x%x\n", mic_report); + return IRQ_HANDLED; +} + +int pm860x_hs_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, + int det, int hook, int hs_shrt, int lo_shrt) +{ + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); + int data; + + pm860x->det.hp_jack = jack; + pm860x->det.hp_det = det; + pm860x->det.hook_det = hook; + pm860x->det.hs_shrt = hs_shrt; + pm860x->det.lo_shrt = lo_shrt; + + if (det & SND_JACK_HEADPHONE) + pm860x_set_bits(codec->control_data, REG_HS_DET, + EN_HS_DET, EN_HS_DET); + /* headset short detect */ + if (hs_shrt) { + data = CLR_SHORT_HS2 | CLR_SHORT_HS1; + pm860x_set_bits(codec->control_data, REG_SHORTS, data, data); + } + /* Lineout short detect */ + if (lo_shrt) { + data = CLR_SHORT_LO2 | CLR_SHORT_LO1; + pm860x_set_bits(codec->control_data, REG_SHORTS, data, data); + } + + /* sync status */ + pm860x_codec_handler(0, pm860x); + return 0; +} +EXPORT_SYMBOL_GPL(pm860x_hs_jack_detect); + +int pm860x_mic_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack, int det) +{ + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); + + pm860x->det.mic_jack = jack; + pm860x->det.mic_det = det; + + if (det & SND_JACK_MICROPHONE) + pm860x_set_bits(codec->control_data, REG_MIC_DET, + MICDET_MASK, MICDET_MASK); + + /* sync status */ + pm860x_codec_handler(0, pm860x); + return 0; +} +EXPORT_SYMBOL_GPL(pm860x_mic_jack_detect); + +static int pm860x_probe(struct snd_soc_codec *codec) +{ + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); + int i, ret; + + pm860x->codec = codec; + + codec->control_data = pm860x->i2c; + + for (i = 0; i < 4; i++) { + ret = request_threaded_irq(pm860x->irq[i], NULL, + pm860x_codec_handler, IRQF_ONESHOT, + pm860x->name[i], pm860x); + if (ret < 0) { + dev_err(codec->dev, "Failed to request IRQ!\n"); + goto out_irq; + } + } + + pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + ret = pm860x_bulk_read(codec->control_data, REG_CACHE_BASE, + REG_CACHE_SIZE, codec->reg_cache); + if (ret < 0) { + dev_err(codec->dev, "Failed to fill register cache: %d\n", + ret); + goto out_codec; + } + + snd_soc_add_controls(codec, pm860x_snd_controls, + ARRAY_SIZE(pm860x_snd_controls)); + snd_soc_dapm_new_controls(codec, pm860x_dapm_widgets, + ARRAY_SIZE(pm860x_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + return 0; + +out_codec: + i = 3; +out_irq: + for (; i >= 0; i--) + free_irq(pm860x->irq[i], pm860x); + return -EINVAL; +} + +static int pm860x_remove(struct snd_soc_codec *codec) +{ + struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec); + int i; + + for (i = 3; i >= 0; i--) + free_irq(pm860x->irq[i], pm860x); + pm860x_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_pm860x = { + .probe = pm860x_probe, + .remove = pm860x_remove, + .read = pm860x_read_reg_cache, + .write = pm860x_write_reg_cache, + .reg_cache_size = REG_CACHE_SIZE, + .reg_word_size = sizeof(u8), + .set_bias_level = pm860x_set_bias_level, +}; + +static int __devinit pm860x_codec_probe(struct platform_device *pdev) +{ + struct pm860x_chip *chip = dev_get_drvdata(pdev->dev.parent); + struct pm860x_priv *pm860x; + struct resource *res; + int i, ret; + + pm860x = kzalloc(sizeof(struct pm860x_priv), GFP_KERNEL); + if (pm860x == NULL) + return -ENOMEM; + + pm860x->chip = chip; + pm860x->i2c = (chip->id == CHIP_PM8607) ? chip->client + : chip->companion; + platform_set_drvdata(pdev, pm860x); + + for (i = 0; i < 4; i++) { + res = platform_get_resource(pdev, IORESOURCE_IRQ, i); + if (!res) { + dev_err(&pdev->dev, "Failed to get IRQ resources\n"); + goto out; + } + pm860x->irq[i] = res->start + chip->irq_base; + strncpy(pm860x->name[i], res->name, MAX_NAME_LEN); + } + + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pm860x, + pm860x_dai, ARRAY_SIZE(pm860x_dai)); + if (ret) { + dev_err(&pdev->dev, "Failed to register codec\n"); + goto out; + } + return ret; + +out: + platform_set_drvdata(pdev, NULL); + kfree(pm860x); + return -EINVAL; +} + +static int __devexit pm860x_codec_remove(struct platform_device *pdev) +{ + struct pm860x_priv *pm860x = platform_get_drvdata(pdev); + + snd_soc_unregister_codec(&pdev->dev); + platform_set_drvdata(pdev, NULL); + kfree(pm860x); + return 0; +} + +static struct platform_driver pm860x_codec_driver = { + .driver = { + .name = "88pm860x-codec", + .owner = THIS_MODULE, + }, + .probe = pm860x_codec_probe, + .remove = __devexit_p(pm860x_codec_remove), +}; + +static __init int pm860x_init(void) +{ + return platform_driver_register(&pm860x_codec_driver); +} +module_init(pm860x_init); + +static __exit void pm860x_exit(void) +{ + platform_driver_unregister(&pm860x_codec_driver); +} +module_exit(pm860x_exit); + +MODULE_DESCRIPTION("ASoC 88PM860x driver"); +MODULE_AUTHOR("Haojian Zhuang "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:88pm860x-codec"); + diff --git a/sound/soc/codecs/88pm860x-codec.h b/sound/soc/codecs/88pm860x-codec.h new file mode 100644 index 000000000000..3364ba4a3607 --- /dev/null +++ b/sound/soc/codecs/88pm860x-codec.h @@ -0,0 +1,97 @@ +/* + * 88pm860x-codec.h -- 88PM860x ALSA SoC Audio Driver + * + * Copyright 2010 Marvell International Ltd. + * Haojian Zhuang + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __88PM860X_H +#define __88PM860X_H + +/* The offset of these registers are 0xb0 */ +#define PM860X_PCM_IFACE_1 0x00 +#define PM860X_PCM_IFACE_2 0x01 +#define PM860X_PCM_IFACE_3 0x02 +#define PM860X_PCM_RATE 0x03 +#define PM860X_EC_PATH 0x04 +#define PM860X_SIDETONE_L_GAIN 0x05 +#define PM860X_SIDETONE_R_GAIN 0x06 +#define PM860X_SIDETONE_SHIFT 0x07 +#define PM860X_ADC_OFFSET_1 0x08 +#define PM860X_ADC_OFFSET_2 0x09 +#define PM860X_DMIC_DELAY 0x0a + +#define PM860X_I2S_IFACE_1 0x0b +#define PM860X_I2S_IFACE_2 0x0c +#define PM860X_I2S_IFACE_3 0x0d +#define PM860X_I2S_IFACE_4 0x0e +#define PM860X_EQUALIZER_N0_1 0x0f +#define PM860X_EQUALIZER_N0_2 0x10 +#define PM860X_EQUALIZER_N1_1 0x11 +#define PM860X_EQUALIZER_N1_2 0x12 +#define PM860X_EQUALIZER_D1_1 0x13 +#define PM860X_EQUALIZER_D1_2 0x14 +#define PM860X_LOFI_GAIN_LEFT 0x15 +#define PM860X_LOFI_GAIN_RIGHT 0x16 +#define PM860X_HIFIL_GAIN_LEFT 0x17 +#define PM860X_HIFIL_GAIN_RIGHT 0x18 +#define PM860X_HIFIR_GAIN_LEFT 0x19 +#define PM860X_HIFIR_GAIN_RIGHT 0x1a +#define PM860X_DAC_OFFSET 0x1b +#define PM860X_OFFSET_LEFT_1 0x1c +#define PM860X_OFFSET_LEFT_2 0x1d +#define PM860X_OFFSET_RIGHT_1 0x1e +#define PM860X_OFFSET_RIGHT_2 0x1f +#define PM860X_ADC_ANA_1 0x20 +#define PM860X_ADC_ANA_2 0x21 +#define PM860X_ADC_ANA_3 0x22 +#define PM860X_ADC_ANA_4 0x23 +#define PM860X_ANA_TO_ANA 0x24 +#define PM860X_HS1_CTRL 0x25 +#define PM860X_HS2_CTRL 0x26 +#define PM860X_LO1_CTRL 0x27 +#define PM860X_LO2_CTRL 0x28 +#define PM860X_EAR_CTRL_1 0x29 +#define PM860X_EAR_CTRL_2 0x2a +#define PM860X_AUDIO_SUPPLIES_1 0x2b +#define PM860X_AUDIO_SUPPLIES_2 0x2c +#define PM860X_ADC_EN_1 0x2d +#define PM860X_ADC_EN_2 0x2e +#define PM860X_DAC_EN_1 0x2f +#define PM860X_DAC_EN_2 0x31 +#define PM860X_AUDIO_CAL_1 0x32 +#define PM860X_AUDIO_CAL_2 0x33 +#define PM860X_AUDIO_CAL_3 0x34 +#define PM860X_AUDIO_CAL_4 0x35 +#define PM860X_AUDIO_CAL_5 0x36 +#define PM860X_ANA_INPUT_SEL_1 0x37 +#define PM860X_ANA_INPUT_SEL_2 0x38 + +#define PM860X_PCM_IFACE_4 0x39 +#define PM860X_I2S_IFACE_5 0x3a + +#define PM860X_SHORTS 0x3b +#define PM860X_PLL_ADJ_1 0x3c +#define PM860X_PLL_ADJ_2 0x3d + +/* bits definition */ +#define PM860X_CLK_DIR_IN 0 +#define PM860X_CLK_DIR_OUT 1 + +#define PM860X_DET_HEADSET (1 << 0) +#define PM860X_DET_MIC (1 << 1) +#define PM860X_DET_HOOK (1 << 2) +#define PM860X_SHORT_HEADSET (1 << 3) +#define PM860X_SHORT_LINEOUT (1 << 4) +#define PM860X_DET_MASK 0x1F + +extern int pm860x_hs_jack_detect(struct snd_soc_codec *, struct snd_soc_jack *, + int, int, int, int); +extern int pm860x_mic_jack_detect(struct snd_soc_codec *, struct snd_soc_jack *, + int); + +#endif /* __88PM860X_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bfdd92b78fb6..a3cfc184ee50 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" + select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS select SND_SOC_AD1836 if SPI_MASTER @@ -85,6 +86,9 @@ config SND_SOC_ALL_CODECS If unsure select "N". +config SND_SOC_88PM860X + tristate + config SND_SOC_WM_HUBS tristate default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 9c3c39fd99ad..b9c43582c5bd 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,3 +1,4 @@ +snd-soc-88pm860x-objs := 88pm860x-codec.o snd-soc-ac97-objs := ac97.o snd-soc-ad1836-objs := ad1836.o snd-soc-ad193x-objs := ad193x.o @@ -67,6 +68,7 @@ snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-wm2000-objs := wm2000.o snd-soc-wm9090-objs := wm9090.o +obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index cf4323dbf9c4..e8d27c8f9ba3 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -318,7 +318,7 @@ EXPORT_SYMBOL_GPL(v253_ops); */ static struct snd_soc_dai_driver cx20442_dai = { - .name = "cx20442-hifi", + .name = "cx20442-voice", .playback = { .stream_name = "Playback", .channels_min = 1, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 0b80e242a66d..efae8b53fd64 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -12,11 +12,11 @@ * * Notes: * The AIC3X is a driver for a low power stereo audio - * codecs aic31, aic32, aic33. + * codecs aic31, aic32, aic33, aic3007. * * It supports full aic33 codec functionality. - * The compatibility with aic32, aic31 is as follows: - * aic32 | aic31 + * The compatibility with aic32, aic31 and aic3007 is as follows: + * aic32/aic3007 | aic31 * --------------------------------------- * MONO_LOUT -> N/A | MONO_LOUT -> N/A * | IN1L -> LINE1L @@ -70,6 +70,10 @@ struct aic3x_priv { unsigned int sysclk; int master; int gpio_reset; +#define AIC3X_MODEL_3X 0 +#define AIC3X_MODEL_33 1 +#define AIC3X_MODEL_3007 2 + u16 model; }; /* @@ -361,6 +365,14 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), }; +/* + * Class-D amplifier gain. From 0 to 18 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(classd_amp_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl = + SOC_DOUBLE_TLV("Class-D Amplifier Gain", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv); + /* Left DAC Mux */ static const struct snd_kcontrol_new aic3x_left_dac_mux_controls = SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]); @@ -589,6 +601,15 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { SND_SOC_DAPM_INPUT("LINE2R"), }; +static const struct snd_soc_dapm_widget aic3007_dapm_widgets[] = { + /* Class-D outputs */ + SND_SOC_DAPM_PGA("Left Class-D Out", CLASSD_CTRL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Class-D Out", CLASSD_CTRL, 2, 0, NULL, 0), + + SND_SOC_DAPM_OUTPUT("SPOP"), + SND_SOC_DAPM_OUTPUT("SPOM"), +}; + static const struct snd_soc_dapm_route intercon[] = { /* Left Output */ {"Left DAC Mux", "DAC_L1", "Left DAC"}, @@ -759,14 +780,30 @@ static const struct snd_soc_dapm_route intercon[] = { {"GPIO1 dmic modclk", NULL, "DMic Rate 32"}, }; +static const struct snd_soc_dapm_route intercon_3007[] = { + /* Class-D outputs */ + {"Left Class-D Out", NULL, "Left Line Out"}, + {"Right Class-D Out", NULL, "Left Line Out"}, + {"SPOP", NULL, "Left Class-D Out"}, + {"SPOM", NULL, "Right Class-D Out"}, +}; + static int aic3x_add_widgets(struct snd_soc_codec *codec) { + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); /* set up audio path interconnects */ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + if (aic3x->model == AIC3X_MODEL_3007) { + snd_soc_dapm_new_controls(codec, aic3007_dapm_widgets, + ARRAY_SIZE(aic3007_dapm_widgets)); + snd_soc_dapm_add_routes(codec, intercon_3007, ARRAY_SIZE(intercon_3007)); + } + return 0; } @@ -1117,6 +1154,7 @@ static struct snd_soc_dai_driver aic3x_dai = { .rates = AIC3X_RATES, .formats = AIC3X_FORMATS,}, .ops = &aic3x_dai_ops, + .symmetric_rates = 1, }; static int aic3x_suspend(struct snd_soc_codec *codec, pm_message_t state) @@ -1150,6 +1188,7 @@ static int aic3x_resume(struct snd_soc_codec *codec) */ static int aic3x_init(struct snd_soc_codec *codec) { + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); int reg; aic3x_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); @@ -1218,6 +1257,17 @@ static int aic3x_init(struct snd_soc_codec *codec) aic3x_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL); aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL); + if (aic3x->model == AIC3X_MODEL_3007) { + /* Class-D speaker driver init; datasheet p. 46 */ + aic3x_write(codec, AIC3X_PAGE_SELECT, 0x0D); + aic3x_write(codec, 0xD, 0x0D); + aic3x_write(codec, 0x8, 0x5C); + aic3x_write(codec, 0x8, 0x5D); + aic3x_write(codec, 0x8, 0x5C); + aic3x_write(codec, AIC3X_PAGE_SELECT, 0x00); + aic3x_write(codec, CLASSD_CTRL, 0); + } + /* off, with power on */ aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1243,6 +1293,8 @@ static int aic3x_probe(struct snd_soc_codec *codec) snd_soc_add_controls(codec, aic3x_snd_controls, ARRAY_SIZE(aic3x_snd_controls)); + if (aic3x->model == AIC3X_MODEL_3007) + snd_soc_add_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); aic3x_add_widgets(codec); @@ -1274,6 +1326,14 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = { * 0x18, 0x19, 0x1A, 0x1B */ +static const struct i2c_device_id aic3x_i2c_id[] = { + [AIC3X_MODEL_3X] = { "tlv320aic3x", 0 }, + [AIC3X_MODEL_33] = { "tlv320aic33", 0 }, + [AIC3X_MODEL_3007] = { "tlv320aic3007", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); + /* * If the i2c layer weren't so broken, we could pass this kind of data * around @@ -1285,6 +1345,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, struct aic3x_setup_data *setup = pdata->setup; struct aic3x_priv *aic3x; int ret, i; + const struct i2c_device_id *tbl; aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL); if (aic3x == NULL) { @@ -1305,6 +1366,12 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, gpio_direction_output(aic3x->gpio_reset, 0); } + for (tbl = aic3x_i2c_id; tbl->name[0]; tbl++) { + if (!strcmp(tbl->name, id->name)) + break; + } + aic3x->model = tbl - aic3x_i2c_id; + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) aic3x->supplies[i].supply = aic3x_supply_names[i]; @@ -1359,13 +1426,6 @@ static int aic3x_i2c_remove(struct i2c_client *client) return 0; } -static const struct i2c_device_id aic3x_i2c_id[] = { - { "tlv320aic3x", 0 }, - { "tlv320aic33", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); - /* machine i2c codec control layer */ static struct i2c_driver aic3x_i2c_driver = { .driver = { diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index f6e3d9b42daf..98e44395b662 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -111,6 +111,8 @@ #define DACL1_2_MONOLOPM_VOL 75 #define DACR1_2_MONOLOPM_VOL 78 #define MONOLOPM_CTRL 79 +/* Class-D speaker driver on tlv320aic3007 */ +#define CLASSD_CTRL 73 /* Line Output Plus/Minus control registers */ #define LINE2L_2_LLOPM_VOL 80 #define LINE2L_2_RLOPM_VOL 87 diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 19844fc8cb1d..56f540838745 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -46,6 +46,7 @@ struct wm8731_priv { struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES]; u16 reg_cache[WM8731_CACHEREGNUM]; unsigned int sysclk; + int sysclk_type; }; @@ -110,6 +111,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls = SOC_DAPM_ENUM("Input Select", wm8731_enum[0]); static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0), SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, &wm8731_output_mixer_controls[0], ARRAY_SIZE(wm8731_output_mixer_controls)), @@ -127,7 +129,18 @@ SND_SOC_DAPM_INPUT("RLINEIN"), SND_SOC_DAPM_INPUT("LLINEIN"), }; +static int wm8731_check_osc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(source->codec); + + return wm8731->sysclk_type == WM8731_SYSCLK_MCLK; +} + static const struct snd_soc_dapm_route intercon[] = { + {"DAC", NULL, "OSC", wm8731_check_osc}, + {"ADC", NULL, "OSC", wm8731_check_osc}, + /* output mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, {"Output Mixer", "HiFi Playback Switch", "DAC"}, @@ -285,6 +298,15 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); + switch (clk_id) { + case WM8731_SYSCLK_XTAL: + case WM8731_SYSCLK_MCLK: + wm8731->sysclk_type = clk_id; + break; + default: + return -EINVAL; + } + switch (freq) { case 11289600: case 12000000: @@ -292,9 +314,14 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, case 16934400: case 18432000: wm8731->sysclk = freq; - return 0; + break; + default: + return -EINVAL; } - return -EINVAL; + + snd_soc_dapm_sync(codec); + + return 0; } diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h index 73a70e206ba9..e9c0c76ab73b 100644 --- a/sound/soc/codecs/wm8731.h +++ b/sound/soc/codecs/wm8731.h @@ -31,7 +31,9 @@ #define WM8731_CACHEREGNUM 10 -#define WM8731_SYSCLK 0 +#define WM8731_SYSCLK_XTAL 1 +#define WM8731_SYSCLK_MCLK 2 + #define WM8731_DAI 0 #endif diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 981868700388..d754d34d68a6 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,24 +1,36 @@ config SND_MPC52xx_DMA tristate -# ASoC platform support for the Freescale MPC8610 SOC. This compiles drivers -# for the SSI and the Elo DMA controller. You will still need to select -# a platform driver and a codec driver. -config SND_SOC_MPC8610 +# ASoC platform support for the Freescale PowerPC SOCs that have an SSI and +# an Elo DMA controller, such as the MPC8610 and P1022. You will still need to +# select a platform driver and a codec driver. +config SND_SOC_POWERPC_SSI tristate - depends on MPC8610 + depends on FSL_SOC config SND_SOC_MPC8610_HPCD tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" # I2C is necessary for the CS4270 driver depends on MPC8610_HPCD && I2C - select SND_SOC_MPC8610 + select SND_SOC_POWERPC_SSI select SND_SOC_CS4270 select SND_SOC_CS4270_VD33_ERRATA default y if MPC8610_HPCD help Say Y if you want to enable audio on the Freescale MPC8610 HPCD. +config SND_SOC_P1022_DS + tristate "ALSA SoC support for the Freescale P1022 DS board" + # I2C is necessary for the WM8776 driver + depends on P1022_DS && I2C + select SND_SOC_POWERPC_SSI + select SND_SOC_WM8776 + default y if P1022_DS + help + Say Y if you want to enable audio on the Freescale P1022 DS board. + This will also include the Wolfson Microelectronics WM8776 codec + driver. + config SND_SOC_MPC5200_I2S tristate "Freescale MPC5200 PSC in I2S mode driver" depends on PPC_MPC52xx && PPC_BESTCOMM diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 7e472a53fcd3..b4a38c0ac58c 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -2,10 +2,14 @@ snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o -# MPC8610 Platform Support +# P1022 DS Machine Support +snd-soc-p1022-ds-objs := p1022_ds.o +obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o + +# Freescale PowerPC SSI/DMA Platform Support snd-soc-fsl-ssi-objs := fsl_ssi.o snd-soc-fsl-dma-objs := fsl_dma.o -obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o +obj-$(CONFIG_SND_SOC_POWERPC_SSI) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o # MPC5200 Platform Support obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index dfe1cb94a70f..4cf98c03af22 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include @@ -305,21 +306,29 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = fsl_dma_dmamask; - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, - fsl_dma_hardware.buffer_bytes_max, - &pcm->streams[0].substream->dma_buffer); - if (ret) { - dev_err(card->dev, "can't allocate playback dma buffer\n"); - return ret; + /* Some codecs have separate DAIs for playback and capture, so we + * should allocate a DMA buffer only for the streams that are valid. + */ + + if (dai->driver->playback.channels_min) { + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, + fsl_dma_hardware.buffer_bytes_max, + &pcm->streams[0].substream->dma_buffer); + if (ret) { + dev_err(card->dev, "can't alloc playback dma buffer\n"); + return ret; + } } - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, - fsl_dma_hardware.buffer_bytes_max, - &pcm->streams[1].substream->dma_buffer); - if (ret) { - snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); - dev_err(card->dev, "can't allocate capture dma buffer\n"); - return ret; + if (dai->driver->capture.channels_min) { + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, + fsl_dma_hardware.buffer_bytes_max, + &pcm->streams[1].substream->dma_buffer); + if (ret) { + snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); + dev_err(card->dev, "can't alloc capture dma buffer\n"); + return ret; + } } return 0; @@ -887,11 +896,11 @@ static struct snd_pcm_ops fsl_dma_ops = { .pointer = fsl_dma_pointer, }; -static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, +static int __devinit fsl_soc_dma_probe(struct platform_device *pdev, const struct of_device_id *match) { struct dma_object *dma; - struct device_node *np = of_dev->dev.of_node; + struct device_node *np = pdev->dev.of_node; struct device_node *ssi_np; struct resource res; const uint32_t *iprop; @@ -900,13 +909,13 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, /* Find the SSI node that points to us. */ ssi_np = find_ssi_node(np); if (!ssi_np) { - dev_err(&of_dev->dev, "cannot find parent SSI node\n"); + dev_err(&pdev->dev, "cannot find parent SSI node\n"); return -ENODEV; } ret = of_address_to_resource(ssi_np, 0, &res); if (ret) { - dev_err(&of_dev->dev, "could not determine resources for %s\n", + dev_err(&pdev->dev, "could not determine resources for %s\n", ssi_np->full_name); of_node_put(ssi_np); return ret; @@ -914,7 +923,7 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL); if (!dma) { - dev_err(&of_dev->dev, "could not allocate dma object\n"); + dev_err(&pdev->dev, "could not allocate dma object\n"); of_node_put(ssi_np); return -ENOMEM; } @@ -937,9 +946,9 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, of_node_put(ssi_np); - ret = snd_soc_register_platform(&of_dev->dev, &dma->dai); + ret = snd_soc_register_platform(&pdev->dev, &dma->dai); if (ret) { - dev_err(&of_dev->dev, "could not register platform\n"); + dev_err(&pdev->dev, "could not register platform\n"); kfree(dma); return ret; } @@ -947,16 +956,16 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev, dma->channel = of_iomap(np, 0); dma->irq = irq_of_parse_and_map(np, 0); - dev_set_drvdata(&of_dev->dev, dma); + dev_set_drvdata(&pdev->dev, dma); return 0; } -static int __devexit fsl_soc_dma_remove(struct of_device *of_dev) +static int __devexit fsl_soc_dma_remove(struct platform_device *pdev) { - struct dma_object *dma = dev_get_drvdata(&of_dev->dev); + struct dma_object *dma = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_platform(&of_dev->dev); + snd_soc_unregister_platform(&pdev->dev); iounmap(dma->channel); irq_dispose_mapping(dma->irq); kfree(dma); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index d1c855ade8fb..4cc167a7aeb8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -624,13 +624,13 @@ static void make_lowercase(char *s) } } -static int __devinit fsl_ssi_probe(struct of_device *of_dev, +static int __devinit fsl_ssi_probe(struct platform_device *pdev, const struct of_device_id *match) { struct fsl_ssi_private *ssi_private; int ret = 0; struct device_attribute *dev_attr = NULL; - struct device_node *np = of_dev->dev.of_node; + struct device_node *np = pdev->dev.of_node; const char *p, *sprop; const uint32_t *iprop; struct resource res; @@ -645,14 +645,14 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev, /* Check for a codec-handle property. */ if (!of_get_property(np, "codec-handle", NULL)) { - dev_err(&of_dev->dev, "missing codec-handle property\n"); + dev_err(&pdev->dev, "missing codec-handle property\n"); return -ENODEV; } /* We only support the SSI in "I2S Slave" mode */ sprop = of_get_property(np, "fsl,mode", NULL); if (!sprop || strcmp(sprop, "i2s-slave")) { - dev_notice(&of_dev->dev, "mode %s is unsupported\n", sprop); + dev_notice(&pdev->dev, "mode %s is unsupported\n", sprop); return -ENODEV; } @@ -661,7 +661,7 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev, ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + strlen(p), GFP_KERNEL); if (!ssi_private) { - dev_err(&of_dev->dev, "could not allocate DAI object\n"); + dev_err(&pdev->dev, "could not allocate DAI object\n"); return -ENOMEM; } @@ -675,7 +675,7 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev, /* Get the addresses and IRQ */ ret = of_address_to_resource(np, 0, &res); if (ret) { - dev_err(&of_dev->dev, "could not determine device resources\n"); + dev_err(&pdev->dev, "could not determine device resources\n"); kfree(ssi_private); return ret; } @@ -703,19 +703,19 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev, dev_attr->attr.mode = S_IRUGO; dev_attr->show = fsl_sysfs_ssi_show; - ret = device_create_file(&of_dev->dev, dev_attr); + ret = device_create_file(&pdev->dev, dev_attr); if (ret) { - dev_err(&of_dev->dev, "could not create sysfs %s file\n", + dev_err(&pdev->dev, "could not create sysfs %s file\n", ssi_private->dev_attr.attr.name); goto error; } /* Register with ASoC */ - dev_set_drvdata(&of_dev->dev, ssi_private); + dev_set_drvdata(&pdev->dev, ssi_private); - ret = snd_soc_register_dai(&of_dev->dev, &ssi_private->cpu_dai_drv); + ret = snd_soc_register_dai(&pdev->dev, &ssi_private->cpu_dai_drv); if (ret) { - dev_err(&of_dev->dev, "failed to register DAI: %d\n", ret); + dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); goto error; } @@ -733,20 +733,20 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev, make_lowercase(name); ssi_private->pdev = - platform_device_register_data(&of_dev->dev, name, 0, NULL, 0); + platform_device_register_data(&pdev->dev, name, 0, NULL, 0); if (IS_ERR(ssi_private->pdev)) { ret = PTR_ERR(ssi_private->pdev); - dev_err(&of_dev->dev, "failed to register platform: %d\n", ret); + dev_err(&pdev->dev, "failed to register platform: %d\n", ret); goto error; } return 0; error: - snd_soc_unregister_dai(&of_dev->dev); - dev_set_drvdata(&of_dev->dev, NULL); + snd_soc_unregister_dai(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); if (dev_attr) - device_remove_file(&of_dev->dev, dev_attr); + device_remove_file(&pdev->dev, dev_attr); irq_dispose_mapping(ssi_private->irq); iounmap(ssi_private->ssi); kfree(ssi_private); @@ -754,16 +754,16 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev, return ret; } -static int fsl_ssi_remove(struct of_device *of_dev) +static int fsl_ssi_remove(struct platform_device *pdev) { - struct fsl_ssi_private *ssi_private = dev_get_drvdata(&of_dev->dev); + struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev); platform_device_unregister(ssi_private->pdev); - snd_soc_unregister_dai(&of_dev->dev); - device_remove_file(&of_dev->dev, &ssi_private->dev_attr); + snd_soc_unregister_dai(&pdev->dev); + device_remove_file(&pdev->dev, &ssi_private->dev_attr); kfree(ssi_private); - dev_set_drvdata(&of_dev->dev, NULL); + dev_set_drvdata(&pdev->dev, NULL); return 0; } diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 38339c158ed9..0d7dcf1e4863 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include @@ -323,9 +324,10 @@ static int get_dma_channel(struct device_node *ssi_np, static int mpc8610_hpcd_probe(struct platform_device *pdev) { struct device *dev = pdev->dev.parent; - /* of_dev is the OF device for the SSI node that probed us */ - struct of_device *of_dev = container_of(dev, struct of_device, dev); - struct device_node *np = of_dev->dev.of_node; + /* ssi_pdev is the platform device for the SSI node that probed us */ + struct platform_device *ssi_pdev = + container_of(dev, struct platform_device, dev); + struct device_node *np = ssi_pdev->dev.of_node; struct device_node *codec_np = NULL; struct platform_device *sound_device = NULL; struct mpc8610_hpcd_data *machine_data; @@ -348,7 +350,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) if (!machine_data) return -ENOMEM; - machine_data->dai[0].cpu_dai_name = dev_name(&of_dev->dev); + machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); machine_data->dai[0].ops = &mpc8610_hpcd_ops; /* Determine the codec name, it will be used as the codec DAI name */ diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c new file mode 100644 index 000000000000..f8176e8e1adf --- /dev/null +++ b/sound/soc/fsl/p1022_ds.c @@ -0,0 +1,590 @@ +/** + * Freescale P1022DS ALSA SoC Machine driver + * + * Author: Timur Tabi + * + * Copyright 2010 Freescale Semiconductor, Inc. + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include +#include +#include +#include +#include +#include + +#include "fsl_dma.h" +#include "fsl_ssi.h" + +/* P1022-specific PMUXCR and DMUXCR bit definitions */ + +#define CCSR_GUTS_PMUXCR_UART0_I2C1_MASK 0x0001c000 +#define CCSR_GUTS_PMUXCR_UART0_I2C1_UART0_SSI 0x00010000 +#define CCSR_GUTS_PMUXCR_UART0_I2C1_SSI 0x00018000 + +#define CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK 0x00000c00 +#define CCSR_GUTS_PMUXCR_SSI_DMA_TDM_SSI 0x00000000 + +#define CCSR_GUTS_DMUXCR_PAD 1 /* DMA controller/channel set to pad */ +#define CCSR_GUTS_DMUXCR_SSI 2 /* DMA controller/channel set to SSI */ + +/* + * Set the DMACR register in the GUTS + * + * The DMACR register determines the source of initiated transfers for each + * channel on each DMA controller. Rather than have a bunch of repetitive + * macros for the bit patterns, we just have a function that calculates + * them. + * + * guts: Pointer to GUTS structure + * co: The DMA controller (0 or 1) + * ch: The channel on the DMA controller (0, 1, 2, or 3) + * device: The device to set as the target (CCSR_GUTS_DMUXCR_xxx) + */ +static inline void guts_set_dmuxcr(struct ccsr_guts_85xx __iomem *guts, + unsigned int co, unsigned int ch, unsigned int device) +{ + unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch)); + + clrsetbits_be32(&guts->dmuxcr, 3 << shift, device << shift); +} + +/* There's only one global utilities register */ +static phys_addr_t guts_phys; + +#define DAI_NAME_SIZE 32 + +/** + * machine_data: machine-specific ASoC device data + * + * This structure contains data for a single sound platform device on an + * P1022 DS. Some of the data is taken from the device tree. + */ +struct machine_data { + struct snd_soc_dai_link dai[2]; + struct snd_soc_card card; + unsigned int dai_format; + unsigned int codec_clk_direction; + unsigned int cpu_clk_direction; + unsigned int clk_frequency; + unsigned int ssi_id; /* 0 = SSI1, 1 = SSI2, etc */ + unsigned int dma_id[2]; /* 0 = DMA1, 1 = DMA2, etc */ + unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/ + char codec_name[DAI_NAME_SIZE]; + char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */ +}; + +/** + * p1022_ds_machine_probe: initialize the board + * + * This function is used to initialize the board-specific hardware. + * + * Here we program the DMACR and PMUXCR registers. + */ +static int p1022_ds_machine_probe(struct platform_device *sound_device) +{ + struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct machine_data *mdata = + container_of(card, struct machine_data, card); + struct ccsr_guts_85xx __iomem *guts; + + guts = ioremap(guts_phys, sizeof(struct ccsr_guts_85xx)); + if (!guts) { + dev_err(card->dev, "could not map global utilities\n"); + return -ENOMEM; + } + + /* Enable SSI Tx signal */ + clrsetbits_be32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_UART0_I2C1_MASK, + CCSR_GUTS_PMUXCR_UART0_I2C1_UART0_SSI); + + /* Enable SSI Rx signal */ + clrsetbits_be32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK, + CCSR_GUTS_PMUXCR_SSI_DMA_TDM_SSI); + + /* Enable DMA Channel for SSI */ + guts_set_dmuxcr(guts, mdata->dma_id[0], mdata->dma_channel_id[0], + CCSR_GUTS_DMUXCR_SSI); + + guts_set_dmuxcr(guts, mdata->dma_id[1], mdata->dma_channel_id[1], + CCSR_GUTS_DMUXCR_SSI); + + iounmap(guts); + + return 0; +} + +/** + * p1022_ds_startup: program the board with various hardware parameters + * + * This function takes board-specific information, like clock frequencies + * and serial data formats, and passes that information to the codec and + * transport drivers. + */ +static int p1022_ds_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct machine_data *mdata = + container_of(rtd->card, struct machine_data, card); + struct device *dev = rtd->card->dev; + int ret = 0; + + /* Tell the codec driver what the serial protocol is. */ + ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format); + if (ret < 0) { + dev_err(dev, "could not set codec driver audio format\n"); + return ret; + } + + /* + * Tell the codec driver what the MCLK frequency is, and whether it's + * a slave or master. + */ + ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0, mdata->clk_frequency, + mdata->codec_clk_direction); + if (ret < 0) { + dev_err(dev, "could not set codec driver clock params\n"); + return ret; + } + + return 0; +} + +/** + * p1022_ds_machine_remove: Remove the sound device + * + * This function is called to remove the sound device for one SSI. We + * de-program the DMACR and PMUXCR register. + */ +static int p1022_ds_machine_remove(struct platform_device *sound_device) +{ + struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct machine_data *mdata = + container_of(card, struct machine_data, card); + struct ccsr_guts_85xx __iomem *guts; + + guts = ioremap(guts_phys, sizeof(struct ccsr_guts_85xx)); + if (!guts) { + dev_err(card->dev, "could not map global utilities\n"); + return -ENOMEM; + } + + /* Restore the signal routing */ + clrbits32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_UART0_I2C1_MASK); + clrbits32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK); + guts_set_dmuxcr(guts, mdata->dma_id[0], mdata->dma_channel_id[0], 0); + guts_set_dmuxcr(guts, mdata->dma_id[1], mdata->dma_channel_id[1], 0); + + iounmap(guts); + + return 0; +} + +/** + * p1022_ds_ops: ASoC machine driver operations + */ +static struct snd_soc_ops p1022_ds_ops = { + .startup = p1022_ds_startup, +}; + +/** + * get_node_by_phandle_name - get a node by its phandle name + * + * This function takes a node, the name of a property in that node, and a + * compatible string. Assuming the property is a phandle to another node, + * it returns that node, (optionally) if that node is compatible. + * + * If the property is not a phandle, or the node it points to is not compatible + * with the specific string, then NULL is returned. + */ +static struct device_node *get_node_by_phandle_name(struct device_node *np, + const char *name, const char *compatible) +{ + np = of_parse_phandle(np, name, 0); + if (!np) + return NULL; + + if (!of_device_is_compatible(np, compatible)) { + of_node_put(np); + return NULL; + } + + return np; +} + +/** + * get_parent_cell_index -- return the cell-index of the parent of a node + * + * Return the value of the cell-index property of the parent of the given + * node. This is used for DMA channel nodes that need to know the DMA ID + * of the controller they are on. + */ +static int get_parent_cell_index(struct device_node *np) +{ + struct device_node *parent = of_get_parent(np); + const u32 *iprop; + int ret = -1; + + if (!parent) + return -1; + + iprop = of_get_property(parent, "cell-index", NULL); + if (iprop) + ret = *iprop; + + of_node_put(parent); + + return ret; +} + +/** + * codec_node_dev_name - determine the dev_name for a codec node + * + * This function determines the dev_name for an I2C node. This is the name + * that would be returned by dev_name() if this device_node were part of a + * 'struct device' It's ugly and hackish, but it works. + * + * The dev_name for such devices include the bus number and I2C address. For + * example, "cs4270-codec.0-004f". + */ +static int codec_node_dev_name(struct device_node *np, char *buf, size_t len) +{ + const u32 *iprop; + int bus, addr; + char temp[DAI_NAME_SIZE]; + + of_modalias_node(np, temp, DAI_NAME_SIZE); + + iprop = of_get_property(np, "reg", NULL); + if (!iprop) + return -EINVAL; + + addr = *iprop; + + bus = get_parent_cell_index(np); + if (bus < 0) + return bus; + + snprintf(buf, len, "%s-codec.%u-%04x", temp, bus, addr); + + return 0; +} + +static int get_dma_channel(struct device_node *ssi_np, + const char *compatible, + struct snd_soc_dai_link *dai, + unsigned int *dma_channel_id, + unsigned int *dma_id) +{ + struct resource res; + struct device_node *dma_channel_np; + const u32 *iprop; + int ret; + + dma_channel_np = get_node_by_phandle_name(ssi_np, compatible, + "fsl,ssi-dma-channel"); + if (!dma_channel_np) + return -EINVAL; + + /* Determine the dev_name for the device_node. This code mimics the + * behavior of of_device_make_bus_id(). We need this because ASoC uses + * the dev_name() of the device to match the platform (DMA) device with + * the CPU (SSI) device. It's all ugly and hackish, but it works (for + * now). + * + * dai->platform name should already point to an allocated buffer. + */ + ret = of_address_to_resource(dma_channel_np, 0, &res); + if (ret) + return ret; + snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s", + (unsigned long long) res.start, dma_channel_np->name); + + iprop = of_get_property(dma_channel_np, "cell-index", NULL); + if (!iprop) { + of_node_put(dma_channel_np); + return -EINVAL; + } + + *dma_channel_id = *iprop; + *dma_id = get_parent_cell_index(dma_channel_np); + of_node_put(dma_channel_np); + + return 0; +} + +/** + * p1022_ds_probe: platform probe function for the machine driver + * + * Although this is a machine driver, the SSI node is the "master" node with + * respect to audio hardware connections. Therefore, we create a new ASoC + * device for each new SSI node that has a codec attached. + */ +static int p1022_ds_probe(struct platform_device *pdev) +{ + struct device *dev = pdev->dev.parent; + /* ssi_pdev is the platform device for the SSI node that probed us */ + struct platform_device *ssi_pdev = + container_of(dev, struct platform_device, dev); + struct device_node *np = ssi_pdev->dev.of_node; + struct device_node *codec_np = NULL; + struct platform_device *sound_device = NULL; + struct machine_data *mdata; + int ret = -ENODEV; + const char *sprop; + const u32 *iprop; + + /* Find the codec node for this SSI. */ + codec_np = of_parse_phandle(np, "codec-handle", 0); + if (!codec_np) { + dev_err(dev, "could not find codec node\n"); + return -EINVAL; + } + + mdata = kzalloc(sizeof(struct machine_data), GFP_KERNEL); + if (!mdata) + return -ENOMEM; + + mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev); + mdata->dai[0].ops = &p1022_ds_ops; + + /* Determine the codec name, it will be used as the codec DAI name */ + ret = codec_node_dev_name(codec_np, mdata->codec_name, DAI_NAME_SIZE); + if (ret) { + dev_err(&pdev->dev, "invalid codec node %s\n", + codec_np->full_name); + ret = -EINVAL; + goto error; + } + mdata->dai[0].codec_name = mdata->codec_name; + + /* We register two DAIs per SSI, one for playback and the other for + * capture. We support codecs that have separate DAIs for both playback + * and capture. + */ + memcpy(&mdata->dai[1], &mdata->dai[0], sizeof(struct snd_soc_dai_link)); + + /* The DAI names from the codec (snd_soc_dai_driver.name) */ + mdata->dai[0].codec_dai_name = "wm8776-hifi-playback"; + mdata->dai[1].codec_dai_name = "wm8776-hifi-capture"; + + /* Get the device ID */ + iprop = of_get_property(np, "cell-index", NULL); + if (!iprop) { + dev_err(&pdev->dev, "cell-index property not found\n"); + ret = -EINVAL; + goto error; + } + mdata->ssi_id = *iprop; + + /* Get the serial format and clock direction. */ + sprop = of_get_property(np, "fsl,mode", NULL); + if (!sprop) { + dev_err(&pdev->dev, "fsl,mode property not found\n"); + ret = -EINVAL; + goto error; + } + + if (strcasecmp(sprop, "i2s-slave") == 0) { + mdata->dai_format = SND_SOC_DAIFMT_I2S; + mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; + mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; + + /* In i2s-slave mode, the codec has its own clock source, so we + * need to get the frequency from the device tree and pass it to + * the codec driver. + */ + iprop = of_get_property(codec_np, "clock-frequency", NULL); + if (!iprop || !*iprop) { + dev_err(&pdev->dev, "codec bus-frequency " + "property is missing or invalid\n"); + ret = -EINVAL; + goto error; + } + mdata->clk_frequency = *iprop; + } else if (strcasecmp(sprop, "i2s-master") == 0) { + mdata->dai_format = SND_SOC_DAIFMT_I2S; + mdata->codec_clk_direction = SND_SOC_CLOCK_IN; + mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "lj-slave") == 0) { + mdata->dai_format = SND_SOC_DAIFMT_LEFT_J; + mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; + mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "lj-master") == 0) { + mdata->dai_format = SND_SOC_DAIFMT_LEFT_J; + mdata->codec_clk_direction = SND_SOC_CLOCK_IN; + mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "rj-slave") == 0) { + mdata->dai_format = SND_SOC_DAIFMT_RIGHT_J; + mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; + mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "rj-master") == 0) { + mdata->dai_format = SND_SOC_DAIFMT_RIGHT_J; + mdata->codec_clk_direction = SND_SOC_CLOCK_IN; + mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else if (strcasecmp(sprop, "ac97-slave") == 0) { + mdata->dai_format = SND_SOC_DAIFMT_AC97; + mdata->codec_clk_direction = SND_SOC_CLOCK_OUT; + mdata->cpu_clk_direction = SND_SOC_CLOCK_IN; + } else if (strcasecmp(sprop, "ac97-master") == 0) { + mdata->dai_format = SND_SOC_DAIFMT_AC97; + mdata->codec_clk_direction = SND_SOC_CLOCK_IN; + mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT; + } else { + dev_err(&pdev->dev, + "unrecognized fsl,mode property '%s'\n", sprop); + ret = -EINVAL; + goto error; + } + + if (!mdata->clk_frequency) { + dev_err(&pdev->dev, "unknown clock frequency\n"); + ret = -EINVAL; + goto error; + } + + /* Find the playback DMA channel to use. */ + mdata->dai[0].platform_name = mdata->platform_name[0]; + ret = get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0], + &mdata->dma_channel_id[0], + &mdata->dma_id[0]); + if (ret) { + dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n"); + goto error; + } + + /* Find the capture DMA channel to use. */ + mdata->dai[1].platform_name = mdata->platform_name[1]; + ret = get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1], + &mdata->dma_channel_id[1], + &mdata->dma_id[1]); + if (ret) { + dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n"); + goto error; + } + + /* Initialize our DAI data structure. */ + mdata->dai[0].stream_name = "playback"; + mdata->dai[1].stream_name = "capture"; + mdata->dai[0].name = mdata->dai[0].stream_name; + mdata->dai[1].name = mdata->dai[1].stream_name; + + mdata->card.probe = p1022_ds_machine_probe; + mdata->card.remove = p1022_ds_machine_remove; + mdata->card.name = pdev->name; /* The platform driver name */ + mdata->card.num_links = 2; + mdata->card.dai_link = mdata->dai; + + /* Allocate a new audio platform device structure */ + sound_device = platform_device_alloc("soc-audio", -1); + if (!sound_device) { + dev_err(&pdev->dev, "platform device alloc failed\n"); + ret = -ENOMEM; + goto error; + } + + /* Associate the card data with the sound device */ + platform_set_drvdata(sound_device, &mdata->card); + + /* Register with ASoC */ + ret = platform_device_add(sound_device); + if (ret) { + dev_err(&pdev->dev, "platform device add failed\n"); + goto error; + } + + of_node_put(codec_np); + + return 0; + +error: + of_node_put(codec_np); + + if (sound_device) + platform_device_unregister(sound_device); + + kfree(mdata); + + return ret; +} + +/** + * p1022_ds_remove: remove the platform device + * + * This function is called when the platform device is removed. + */ +static int __devexit p1022_ds_remove(struct platform_device *pdev) +{ + struct platform_device *sound_device = dev_get_drvdata(&pdev->dev); + struct snd_soc_card *card = platform_get_drvdata(sound_device); + struct machine_data *mdata = + container_of(card, struct machine_data, card); + + platform_device_unregister(sound_device); + + kfree(mdata); + sound_device->dev.platform_data = NULL; + + dev_set_drvdata(&pdev->dev, NULL); + + return 0; +} + +static struct platform_driver p1022_ds_driver = { + .probe = p1022_ds_probe, + .remove = __devexit_p(p1022_ds_remove), + .driver = { + /* The name must match the 'model' property in the device tree, + * in lowercase letters, but only the part after that last + * comma. This is because some model properties have a "fsl," + * prefix. + */ + .name = "snd-soc-p1022", + .owner = THIS_MODULE, + }, +}; + +/** + * p1022_ds_init: machine driver initialization. + * + * This function is called when this module is loaded. + */ +static int __init p1022_ds_init(void) +{ + struct device_node *guts_np; + struct resource res; + + pr_info("Freescale P1022 DS ALSA SoC machine driver\n"); + + /* Get the physical address of the global utilities registers */ + guts_np = of_find_compatible_node(NULL, NULL, "fsl,p1022-guts"); + if (of_address_to_resource(guts_np, 0, &res)) { + pr_err("p1022-ds: missing/invalid global utilities node\n"); + return -EINVAL; + } + guts_phys = res.start; + of_node_put(guts_np); + + return platform_driver_register(&p1022_ds_driver); +} + +/** + * p1022_ds_exit: machine driver exit + * + * This function is called when this driver is unloaded. + */ +static void __exit p1022_ds_exit(void) +{ + platform_driver_unregister(&p1022_ds_driver); +} + +module_init(p1022_ds_init); +module_exit(p1022_ds_exit); + +MODULE_AUTHOR("Timur Tabi "); +MODULE_DESCRIPTION("Freescale P1022 DS ALSA SoC machine driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 9d88efa06e3c..438146addbb8 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -584,7 +584,7 @@ static struct snd_soc_dai_link ams_delta_dai_link = { .name = "CX20442", .stream_name = "CX20442", .cpu_dai_name ="omap-mcbsp-dai.0", - .codec_dai_name = "cx20442-hifi", + .codec_dai_name = "cx20442-voice", .init = ams_delta_cx20442_init, .platform_name = "omap-pcm-audio", .codec_name = "cx20442-codec", diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index e30c8325f35e..37f191bbfdd9 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -117,6 +117,24 @@ config SND_PXA2XX_SOC_PALM27X Say Y if you want to add support for SoC audio on Palm T|X, T5, E2 or LifeDrive handheld computer. +config SND_SOC_SAARB + tristate "SoC Audio support for Marvell Saarb" + depends on SND_PXA2XX_SOC && MACH_SAARB + select SND_PXA_SOC_SSP + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on the + Marvell Saarb reference platform. + +config SND_SOC_TAVOREVB3 + tristate "SoC Audio support for Marvell Tavor EVB3" + depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 + select SND_PXA_SOC_SSP + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on the + Marvell Saarb reference platform. + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index caa03d8f4789..07660165ec8d 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -19,6 +19,8 @@ snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o snd-soc-palm27x-objs := palm27x.o +snd-soc-saarb-objs := saarb.o +snd-soc-tavorevb3-objs := tavorevb3.o snd-soc-zylonite-objs := zylonite.o snd-soc-magician-objs := magician.o snd-soc-mioa701-objs := mioa701_wm9713.o @@ -38,6 +40,8 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o +obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o +obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 555689cf6727..97e9423615c9 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -149,7 +149,7 @@ static int corgi_hw_params(struct snd_pcm_substream *substream, return ret; /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index f614607b2055..c82cedb602fd 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -198,6 +198,9 @@ static int __init e740_init(void) static void __exit e740_exit(void) { platform_device_unregister(e740_snd_device); + gpio_free(GPIO_E740_WM9705_nAVDD2); + gpio_free(GPIO_E740_AMP_ON); + gpio_free(GPIO_E740_MIC_ON); } module_init(e740_init); diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index add0e1c25bc8..fa752f6ec37d 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -128,7 +128,7 @@ static int poodle_hw_params(struct snd_pcm_substream *substream, return ret; /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 9c2bafa112ad..ac51c6d25c42 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -24,7 +24,6 @@ #include #include -#include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c new file mode 100644 index 000000000000..d63cb474b4e1 --- /dev/null +++ b/sound/soc/pxa/saarb.c @@ -0,0 +1,200 @@ +/* + * saarb.c -- SoC audio for saarb + * + * Copyright (C) 2010 Marvell International Ltd. + * Haojian Zhuang + * + * 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 +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include + +#include "../codecs/88pm860x-codec.h" +#include "pxa-ssp.h" + +static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd); + +static struct platform_device *saarb_snd_device; + +static struct snd_soc_jack hs_jack, mic_jack; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, +}; + +static struct snd_soc_jack_pin mic_jack_pins[] = { + { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, +}; + +/* saarb machine dapm widgets */ +static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Stereophone", NULL), + SND_SOC_DAPM_LINE("Lineout Out 1", NULL), + SND_SOC_DAPM_LINE("Lineout Out 2", NULL), + SND_SOC_DAPM_SPK("Ext Speaker", NULL), + SND_SOC_DAPM_MIC("Ext Mic 1", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Ext Mic 3", NULL), +}; + +/* saarb machine audio map */ +static const struct snd_soc_dapm_route audio_map[] = { + {"Headset Stereophone", NULL, "HS1"}, + {"Headset Stereophone", NULL, "HS2"}, + + {"Ext Speaker", NULL, "LSP"}, + {"Ext Speaker", NULL, "LSN"}, + + {"Lineout Out 1", NULL, "LINEOUT1"}, + {"Lineout Out 2", NULL, "LINEOUT2"}, + + {"MIC1P", NULL, "Mic1 Bias"}, + {"MIC1N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Ext Mic 1"}, + + {"MIC2P", NULL, "Mic1 Bias"}, + {"MIC2N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Headset Mic 2"}, + + {"MIC3P", NULL, "Mic3 Bias"}, + {"MIC3N", NULL, "Mic3 Bias"}, + {"Mic3 Bias", NULL, "Ext Mic 3"}, +}; + +static int saarb_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int width = snd_pcm_format_physical_width(params_format(params)); + int ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, + PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); + + return ret; +} + +static struct snd_soc_ops saarb_i2s_ops = { + .hw_params = saarb_i2s_hw_params, +}; + +static struct snd_soc_dai_link saarb_dai[] = { + { + .name = "88PM860x I2S", + .stream_name = "I2S Audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .platform_name = "pxa-pcm-audio", + .codec_name = "88pm860x-codec", + .init = saarb_pm860x_init, + .ops = &saarb_i2s_ops, + }, +}; + +static struct snd_soc_card snd_soc_card_saarb = { + .name = "Saarb", + .dai_link = saarb_dai, + .num_links = ARRAY_SIZE(saarb_dai), +}; + +static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + snd_soc_dapm_new_controls(codec, saarb_dapm_widgets, + ARRAY_SIZE(saarb_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Speaker"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 1"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 3"); + snd_soc_dapm_disable_pin(codec, "Headset Mic 2"); + snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); + + ret = snd_soc_dapm_sync(codec); + if (ret) + return ret; + + /* Headset jack detection */ + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE + | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, + &hs_jack); + snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, + &mic_jack); + snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), + mic_jack_pins); + + /* headphone, microphone detection & headset short detection */ + pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, + SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); + pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE); + return 0; +} + +static int __init saarb_init(void) +{ + int ret; + + if (!machine_is_saarb()) + return -ENODEV; + saarb_snd_device = platform_device_alloc("soc-audio", -1); + if (!saarb_snd_device) + return -ENOMEM; + + platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb); + + ret = platform_device_add(saarb_snd_device); + if (ret) + platform_device_put(saarb_snd_device); + + return ret; +} + +static void __exit saarb_exit(void) +{ + platform_device_unregister(saarb_snd_device); +} + +module_init(saarb_init); +module_exit(saarb_exit); + +MODULE_AUTHOR("Haojian Zhuang "); +MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c new file mode 100644 index 000000000000..248c283fc4df --- /dev/null +++ b/sound/soc/pxa/tavorevb3.c @@ -0,0 +1,200 @@ +/* + * tavorevb3.c -- SoC audio for Tavor EVB3 + * + * Copyright (C) 2010 Marvell International Ltd. + * Haojian Zhuang + * + * 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 +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include + +#include "../codecs/88pm860x-codec.h" +#include "pxa-ssp.h" + +static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd); + +static struct platform_device *evb3_snd_device; + +static struct snd_soc_jack hs_jack, mic_jack; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, +}; + +static struct snd_soc_jack_pin mic_jack_pins[] = { + { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, +}; + +/* tavorevb3 machine dapm widgets */ +static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_LINE("Lineout Out 1", NULL), + SND_SOC_DAPM_LINE("Lineout Out 2", NULL), + SND_SOC_DAPM_SPK("Ext Speaker", NULL), + SND_SOC_DAPM_MIC("Ext Mic 1", NULL), + SND_SOC_DAPM_MIC("Headset Mic 2", NULL), + SND_SOC_DAPM_MIC("Ext Mic 3", NULL), +}; + +/* tavorevb3 machine audio map */ +static const struct snd_soc_dapm_route audio_map[] = { + {"Headset Stereophone", NULL, "HS1"}, + {"Headset Stereophone", NULL, "HS2"}, + + {"Ext Speaker", NULL, "LSP"}, + {"Ext Speaker", NULL, "LSN"}, + + {"Lineout Out 1", NULL, "LINEOUT1"}, + {"Lineout Out 2", NULL, "LINEOUT2"}, + + {"MIC1P", NULL, "Mic1 Bias"}, + {"MIC1N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Ext Mic 1"}, + + {"MIC2P", NULL, "Mic1 Bias"}, + {"MIC2N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Headset Mic 2"}, + + {"MIC3P", NULL, "Mic3 Bias"}, + {"MIC3N", NULL, "Mic3 Bias"}, + {"Mic3 Bias", NULL, "Ext Mic 3"}, +}; + +static int evb3_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int width = snd_pcm_format_physical_width(params_format(params)); + int ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0, + PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width); + return ret; +} + +static struct snd_soc_ops evb3_i2s_ops = { + .hw_params = evb3_i2s_hw_params, +}; + +static struct snd_soc_dai_link evb3_dai[] = { + { + .name = "88PM860x I2S", + .stream_name = "I2S Audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .platform_name = "pxa-pcm-audio", + .codec_name = "88pm860x-codec", + .init = evb3_pm860x_init, + .ops = &evb3_i2s_ops, + }, +}; + +static struct snd_soc_card snd_soc_card_evb3 = { + .name = "Tavor EVB3", + .dai_link = evb3_dai, + .num_links = ARRAY_SIZE(evb3_dai), +}; + +static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + snd_soc_dapm_new_controls(codec, evb3_dapm_widgets, + ARRAY_SIZE(evb3_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Speaker"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 1"); + snd_soc_dapm_enable_pin(codec, "Ext Mic 3"); + snd_soc_dapm_disable_pin(codec, "Headset Mic 2"); + snd_soc_dapm_disable_pin(codec, "Headset Stereophone"); + + ret = snd_soc_dapm_sync(codec); + if (ret) + return ret; + + /* Headset jack detection */ + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE + | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, + &hs_jack); + snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, + &mic_jack); + snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), + mic_jack_pins); + + /* headphone, microphone detection & headset short detection */ + pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, + SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); + pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE); + return 0; +} + +static int __init tavorevb3_init(void) +{ + int ret; + + if (!machine_is_tavorevb3()) + return -ENODEV; + evb3_snd_device = platform_device_alloc("soc-audio", -1); + if (!evb3_snd_device) + return -ENOMEM; + + platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3); + + ret = platform_device_add(evb3_snd_device); + if (ret) + platform_device_put(evb3_snd_device); + + return ret; +} + +static void __exit tavorevb3_exit(void) +{ + platform_device_unregister(evb3_snd_device); +} + +module_init(tavorevb3_init); +module_exit(tavorevb3_exit); + +MODULE_AUTHOR("Haojian Zhuang "); +MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7093c1787128..65352c7d4b7f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3122,10 +3122,12 @@ int snd_soc_register_codec(struct device *dev, fixup_codec_formats(&dai_drv[i].capture); } - /* register DAIs */ - ret = snd_soc_register_dais(dev, dai_drv, num_dai); - if (ret < 0) + /* register any DAIs */ + if (num_dai) { + ret = snd_soc_register_dais(dev, dai_drv, num_dai); + if (ret < 0) goto error; + } mutex_lock(&client_mutex); list_add(&codec->list, &codec_list); @@ -3164,8 +3166,9 @@ void snd_soc_unregister_codec(struct device *dev) return; found: - for (i = 0; i < codec->num_dai; i++) - snd_soc_unregister_dai(dev); + if (codec->num_dai) + for (i = 0; i < codec->num_dai; i++) + snd_soc_unregister_dai(dev); mutex_lock(&client_mutex); list_del(&codec->list);