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;
}
.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);
struct atmel_ssc_state ssc_state;
};
+int atmel_ssc_set_audio(int ssc);
+
#endif /* _AT91_SSC_DAI_H */
.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,
};
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.
*/
--- /dev/null
+/*
+ * 88pm860x-codec.c -- 88PM860x ALSA SoC Audio Driver
+ *
+ * Copyright 2010 Marvell International Ltd.
+ * Author: Haojian Zhuang <haojian.zhuang@marvell.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/88pm860x.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <sound/jack.h>
+
+#include "88pm860x-codec.h"
+
+#define MAX_NAME_LEN 20
+#define REG_CACHE_SIZE 0x40
+#define REG_CACHE_BASE 0xb0
+
+/* Status Register 1 (0x01) */
+#define REG_STATUS_1 0x01
+#define MIC_STATUS (1 << 7)
+#define HOOK_STATUS (1 << 6)
+#define HEADSET_STATUS (1 << 5)
+
+/* Mic Detection Register (0x37) */
+#define REG_MIC_DET 0x37
+#define CONTINUOUS_POLLING (3 << 1)
+#define EN_MIC_DET (1 << 0)
+#define MICDET_MASK 0x07
+
+/* Headset Detection Register (0x38) */
+#define REG_HS_DET 0x38
+#define EN_HS_DET (1 << 0)
+
+/* Misc2 Register (0x42) */
+#define REG_MISC2 0x42
+#define AUDIO_PLL (1 << 5)
+#define AUDIO_SECTION_RESET (1 << 4)
+#define AUDIO_SECTION_ON (1 << 3)
+
+/* PCM Interface Register 2 (0xb1) */
+#define PCM_INF2_BCLK (1 << 6) /* Bit clock polarity */
+#define PCM_INF2_FS (1 << 5) /* Frame Sync polarity */
+#define PCM_INF2_MASTER (1 << 4) /* Master / Slave */
+#define PCM_INF2_18WL (1 << 3) /* 18 / 16 bits */
+#define PCM_GENERAL_I2S 0
+#define PCM_EXACT_I2S 1
+#define PCM_LEFT_I2S 2
+#define PCM_RIGHT_I2S 3
+#define PCM_SHORT_FS 4
+#define PCM_LONG_FS 5
+#define PCM_MODE_MASK 7
+
+/* I2S Interface Register 4 (0xbe) */
+#define I2S_EQU_BYP (1 << 6)
+
+/* DAC Offset Register (0xcb) */
+#define DAC_MUTE (1 << 7)
+#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 <haojian.zhuang@marvell.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:88pm860x-codec");
+
--- /dev/null
+/*
+ * 88pm860x-codec.h -- 88PM860x ALSA SoC Audio Driver
+ *
+ * Copyright 2010 Marvell International Ltd.
+ * Haojian Zhuang <haojian.zhuang@marvell.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __88PM860X_H
+#define __88PM860X_H
+
+/* The offset of these registers are 0xb0 */
+#define PM860X_PCM_IFACE_1 0x00
+#define PM860X_PCM_IFACE_2 0x01
+#define PM860X_PCM_IFACE_3 0x02
+#define PM860X_PCM_RATE 0x03
+#define PM860X_EC_PATH 0x04
+#define PM860X_SIDETONE_L_GAIN 0x05
+#define PM860X_SIDETONE_R_GAIN 0x06
+#define PM860X_SIDETONE_SHIFT 0x07
+#define PM860X_ADC_OFFSET_1 0x08
+#define PM860X_ADC_OFFSET_2 0x09
+#define PM860X_DMIC_DELAY 0x0a
+
+#define PM860X_I2S_IFACE_1 0x0b
+#define PM860X_I2S_IFACE_2 0x0c
+#define PM860X_I2S_IFACE_3 0x0d
+#define PM860X_I2S_IFACE_4 0x0e
+#define PM860X_EQUALIZER_N0_1 0x0f
+#define PM860X_EQUALIZER_N0_2 0x10
+#define PM860X_EQUALIZER_N1_1 0x11
+#define PM860X_EQUALIZER_N1_2 0x12
+#define PM860X_EQUALIZER_D1_1 0x13
+#define PM860X_EQUALIZER_D1_2 0x14
+#define PM860X_LOFI_GAIN_LEFT 0x15
+#define PM860X_LOFI_GAIN_RIGHT 0x16
+#define PM860X_HIFIL_GAIN_LEFT 0x17
+#define PM860X_HIFIL_GAIN_RIGHT 0x18
+#define PM860X_HIFIR_GAIN_LEFT 0x19
+#define PM860X_HIFIR_GAIN_RIGHT 0x1a
+#define PM860X_DAC_OFFSET 0x1b
+#define PM860X_OFFSET_LEFT_1 0x1c
+#define PM860X_OFFSET_LEFT_2 0x1d
+#define PM860X_OFFSET_RIGHT_1 0x1e
+#define PM860X_OFFSET_RIGHT_2 0x1f
+#define PM860X_ADC_ANA_1 0x20
+#define PM860X_ADC_ANA_2 0x21
+#define PM860X_ADC_ANA_3 0x22
+#define PM860X_ADC_ANA_4 0x23
+#define PM860X_ANA_TO_ANA 0x24
+#define PM860X_HS1_CTRL 0x25
+#define PM860X_HS2_CTRL 0x26
+#define PM860X_LO1_CTRL 0x27
+#define PM860X_LO2_CTRL 0x28
+#define PM860X_EAR_CTRL_1 0x29
+#define PM860X_EAR_CTRL_2 0x2a
+#define PM860X_AUDIO_SUPPLIES_1 0x2b
+#define PM860X_AUDIO_SUPPLIES_2 0x2c
+#define PM860X_ADC_EN_1 0x2d
+#define PM860X_ADC_EN_2 0x2e
+#define PM860X_DAC_EN_1 0x2f
+#define PM860X_DAC_EN_2 0x31
+#define PM860X_AUDIO_CAL_1 0x32
+#define PM860X_AUDIO_CAL_2 0x33
+#define PM860X_AUDIO_CAL_3 0x34
+#define PM860X_AUDIO_CAL_4 0x35
+#define PM860X_AUDIO_CAL_5 0x36
+#define PM860X_ANA_INPUT_SEL_1 0x37
+#define PM860X_ANA_INPUT_SEL_2 0x38
+
+#define PM860X_PCM_IFACE_4 0x39
+#define PM860X_I2S_IFACE_5 0x3a
+
+#define PM860X_SHORTS 0x3b
+#define PM860X_PLL_ADJ_1 0x3c
+#define PM860X_PLL_ADJ_2 0x3d
+
+/* bits definition */
+#define PM860X_CLK_DIR_IN 0
+#define PM860X_CLK_DIR_OUT 1
+
+#define PM860X_DET_HEADSET (1 << 0)
+#define PM860X_DET_MIC (1 << 1)
+#define PM860X_DET_HOOK (1 << 2)
+#define PM860X_SHORT_HEADSET (1 << 3)
+#define PM860X_SHORT_LINEOUT (1 << 4)
+#define PM860X_DET_MASK 0x1F
+
+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 */
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
select SND_SOC_TWL6040 if TWL4030_CORE
select SND_SOC_UDA134X
select SND_SOC_UDA1380 if I2C
+ select SND_SOC_WL1273 if WL1273_CORE
select SND_SOC_WM2000 if I2C
select SND_SOC_WM8350 if MFD_WM8350
select SND_SOC_WM8400 if MFD_WM8400
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
config SND_SOC_UDA1380
tristate
+config SND_SOC_WL1273
+ tristate
+
config SND_SOC_WM8350
tristate
+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
snd-soc-twl6040-objs := twl6040.o
snd-soc-uda134x-objs := uda134x.o
snd-soc-uda1380-objs := uda1380.o
+snd-soc-wl1273-objs := wl1273.o
snd-soc-wm8350-objs := wm8350.o
snd-soc-wm8400-objs := wm8400.o
snd-soc-wm8510-objs := wm8510.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
obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o
obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o
obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o
+obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o
obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o
obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o
obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o
*/
static struct snd_soc_dai_driver cx20442_dai = {
- .name = "cx20442-hifi",
+ .name = "cx20442-voice",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
*
* 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
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;
};
/*
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]);
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"},
{"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;
}
.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)
*/
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);
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);
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);
* 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
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) {
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];
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 = {
#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
--- /dev/null
+/*
+ * ALSA SoC WL1273 codec driver
+ *
+ * Author: Matti Aaltonen, <matti.j.aaltonen@nokia.com>
+ *
+ * Copyright: (C) 2010 Nokia Corporation
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/mfd/wl1273-core.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wl1273.h"
+
+enum wl1273_mode { WL1273_MODE_BT, WL1273_MODE_FM_RX, WL1273_MODE_FM_TX };
+
+/* codec private data */
+struct wl1273_priv {
+ enum wl1273_mode mode;
+ struct wl1273_core *core;
+ unsigned int channels;
+};
+
+static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core,
+ int rate, int width)
+{
+ struct device *dev = &core->i2c_dev->dev;
+ int r = 0;
+ u16 mode;
+
+ dev_dbg(dev, "rate: %d\n", rate);
+ dev_dbg(dev, "width: %d\n", width);
+
+ mutex_lock(&core->lock);
+
+ mode = core->i2s_mode & ~WL1273_IS2_WIDTH & ~WL1273_IS2_RATE;
+
+ switch (rate) {
+ case 48000:
+ mode |= WL1273_IS2_RATE_48K;
+ break;
+ case 44100:
+ mode |= WL1273_IS2_RATE_44_1K;
+ break;
+ case 32000:
+ mode |= WL1273_IS2_RATE_32K;
+ break;
+ case 22050:
+ mode |= WL1273_IS2_RATE_22_05K;
+ break;
+ case 16000:
+ mode |= WL1273_IS2_RATE_16K;
+ break;
+ case 12000:
+ mode |= WL1273_IS2_RATE_12K;
+ break;
+ case 11025:
+ mode |= WL1273_IS2_RATE_11_025;
+ break;
+ case 8000:
+ mode |= WL1273_IS2_RATE_8K;
+ break;
+ default:
+ dev_err(dev, "Sampling rate: %d not supported\n", rate);
+ r = -EINVAL;
+ goto out;
+ }
+
+ switch (width) {
+ case 16:
+ mode |= WL1273_IS2_WIDTH_32;
+ break;
+ case 20:
+ mode |= WL1273_IS2_WIDTH_40;
+ break;
+ case 24:
+ mode |= WL1273_IS2_WIDTH_48;
+ break;
+ case 25:
+ mode |= WL1273_IS2_WIDTH_50;
+ break;
+ case 30:
+ mode |= WL1273_IS2_WIDTH_60;
+ break;
+ case 32:
+ mode |= WL1273_IS2_WIDTH_64;
+ break;
+ case 40:
+ mode |= WL1273_IS2_WIDTH_80;
+ break;
+ case 48:
+ mode |= WL1273_IS2_WIDTH_96;
+ break;
+ case 64:
+ mode |= WL1273_IS2_WIDTH_128;
+ break;
+ default:
+ dev_err(dev, "Data width: %d not supported\n", width);
+ r = -EINVAL;
+ goto out;
+ }
+
+ dev_dbg(dev, "WL1273_I2S_DEF_MODE: 0x%04x\n", WL1273_I2S_DEF_MODE);
+ dev_dbg(dev, "core->i2s_mode: 0x%04x\n", core->i2s_mode);
+ dev_dbg(dev, "mode: 0x%04x\n", mode);
+
+ if (core->i2s_mode != mode) {
+ r = wl1273_fm_write_cmd(core, WL1273_I2S_MODE_CONFIG_SET, mode);
+ if (r)
+ goto out;
+
+ core->i2s_mode = mode;
+ r = wl1273_fm_write_cmd(core, WL1273_AUDIO_ENABLE,
+ WL1273_AUDIO_ENABLE_I2S);
+ if (r)
+ goto out;
+ }
+out:
+ mutex_unlock(&core->lock);
+
+ return r;
+}
+
+static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core,
+ int channel_number)
+{
+ struct i2c_client *client = core->i2c_dev;
+ struct device *dev = &client->dev;
+ int r = 0;
+
+ dev_dbg(dev, "%s\n", __func__);
+
+ mutex_lock(&core->lock);
+
+ if (core->channel_number == channel_number)
+ goto out;
+
+ if (channel_number == 1 && core->mode == WL1273_MODE_RX)
+ r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET,
+ WL1273_RX_MONO);
+ else if (channel_number == 1 && core->mode == WL1273_MODE_TX)
+ r = wl1273_fm_write_cmd(core, WL1273_MONO_SET,
+ WL1273_TX_MONO);
+ else if (channel_number == 2 && core->mode == WL1273_MODE_RX)
+ r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET,
+ WL1273_RX_STEREO);
+ else if (channel_number == 2 && core->mode == WL1273_MODE_TX)
+ r = wl1273_fm_write_cmd(core, WL1273_MONO_SET,
+ WL1273_TX_STEREO);
+ else
+ r = -EINVAL;
+out:
+ mutex_unlock(&core->lock);
+
+ return r;
+}
+
+static int snd_wl1273_get_audio_route(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.integer.value[0] = wl1273->mode;
+
+ return 0;
+}
+
+static const char *wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" };
+
+static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+ /* Do not allow changes while stream is running */
+ if (codec->active)
+ return -EPERM;
+
+ if (ucontrol->value.integer.value[0] < 0 ||
+ ucontrol->value.integer.value[0] >= ARRAY_SIZE(wl1273_audio_route))
+ return -EINVAL;
+
+ wl1273->mode = ucontrol->value.integer.value[0];
+
+ return 1;
+}
+
+static const struct soc_enum wl1273_enum =
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_route), wl1273_audio_route);
+
+static int snd_wl1273_fm_audio_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+ dev_dbg(codec->dev, "%s: enter.\n", __func__);
+
+ ucontrol->value.integer.value[0] = wl1273->core->audio_mode;
+
+ return 0;
+}
+
+static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+ int val, r = 0;
+
+ dev_dbg(codec->dev, "%s: enter.\n", __func__);
+
+ val = ucontrol->value.integer.value[0];
+ if (wl1273->core->audio_mode == val)
+ return 0;
+
+ r = wl1273_fm_set_audio(wl1273->core, val);
+ if (r < 0)
+ return r;
+
+ return 1;
+}
+
+static const char *wl1273_audio_strings[] = { "Digital", "Analog" };
+
+static const struct soc_enum wl1273_audio_enum =
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings),
+ wl1273_audio_strings);
+
+static int snd_wl1273_fm_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+ dev_dbg(codec->dev, "%s: enter.\n", __func__);
+
+ ucontrol->value.integer.value[0] = wl1273->core->volume;
+
+ return 0;
+}
+
+static int snd_wl1273_fm_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+ int r;
+
+ dev_dbg(codec->dev, "%s: enter.\n", __func__);
+
+ r = wl1273_fm_set_volume(wl1273->core,
+ ucontrol->value.integer.value[0]);
+ if (r)
+ return r;
+
+ return 1;
+}
+
+static const struct snd_kcontrol_new wl1273_controls[] = {
+ SOC_ENUM_EXT("Codec Mode", wl1273_enum,
+ snd_wl1273_get_audio_route, snd_wl1273_set_audio_route),
+ SOC_ENUM_EXT("Audio Switch", wl1273_audio_enum,
+ snd_wl1273_fm_audio_get, snd_wl1273_fm_audio_put),
+ SOC_SINGLE_EXT("Volume", 0, 0, WL1273_MAX_VOLUME, 0,
+ snd_wl1273_fm_volume_get, snd_wl1273_fm_volume_put),
+};
+
+static int wl1273_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+ switch (wl1273->mode) {
+ case WL1273_MODE_BT:
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ 8000, 8000);
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_CHANNELS, 1, 1);
+ break;
+ case WL1273_MODE_FM_RX:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ pr_err("Cannot play in RX mode.\n");
+ return -EINVAL;
+ }
+ break;
+ case WL1273_MODE_FM_TX:
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ pr_err("Cannot capture in TX mode.\n");
+ return -EINVAL;
+ }
+ break;
+ default:
+ return -EINVAL;
+ break;
+ }
+
+ return 0;
+}
+
+static int wl1273_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec);
+ struct wl1273_core *core = wl1273->core;
+ unsigned int rate, width, r;
+
+ if (params_format(params) != SNDRV_PCM_FORMAT_S16_LE) {
+ pr_err("Only SNDRV_PCM_FORMAT_S16_LE supported.\n");
+ return -EINVAL;
+ }
+
+ rate = params_rate(params);
+ width = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min;
+
+ if (wl1273->mode == WL1273_MODE_BT) {
+ if (rate != 8000) {
+ pr_err("Rate %d not supported.\n", params_rate(params));
+ return -EINVAL;
+ }
+
+ if (params_channels(params) != 1) {
+ pr_err("Only mono supported.\n");
+ return -EINVAL;
+ }
+
+ return 0;
+ }
+
+ if (wl1273->mode == WL1273_MODE_FM_TX &&
+ substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ pr_err("Only playback supported with TX.\n");
+ return -EINVAL;
+ }
+
+ if (wl1273->mode == WL1273_MODE_FM_RX &&
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ pr_err("Only capture supported with RX.\n");
+ return -EINVAL;
+ }
+
+ if (wl1273->mode != WL1273_MODE_FM_RX &&
+ wl1273->mode != WL1273_MODE_FM_TX) {
+ pr_err("Unexpected mode: %d.\n", wl1273->mode);
+ return -EINVAL;
+ }
+
+ r = snd_wl1273_fm_set_i2s_mode(core, rate, width);
+ if (r)
+ return r;
+
+ wl1273->channels = params_channels(params);
+ r = snd_wl1273_fm_set_channel_number(core, wl1273->channels);
+ if (r)
+ return r;
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops wl1273_dai_ops = {
+ .startup = wl1273_startup,
+ .hw_params = wl1273_hw_params,
+};
+
+static struct snd_soc_dai_driver wl1273_dai = {
+ .name = "wl1273-fm",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE},
+ .ops = &wl1273_dai_ops,
+};
+
+/* Audio interface format for the soc_card driver */
+int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt)
+{
+ struct wl1273_priv *wl1273;
+
+ if (codec == NULL || fmt == NULL)
+ return -EINVAL;
+
+ wl1273 = snd_soc_codec_get_drvdata(codec);
+
+ switch (wl1273->mode) {
+ case WL1273_MODE_FM_RX:
+ case WL1273_MODE_FM_TX:
+ *fmt = SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+
+ break;
+ case WL1273_MODE_BT:
+ *fmt = SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(wl1273_get_format);
+
+static int wl1273_probe(struct snd_soc_codec *codec)
+{
+ struct wl1273_core **core = codec->dev->platform_data;
+ struct wl1273_priv *wl1273;
+ int r;
+
+ dev_dbg(codec->dev, "%s.\n", __func__);
+
+ if (!core) {
+ dev_err(codec->dev, "Platform data is missing.\n");
+ return -EINVAL;
+ }
+
+ wl1273 = kzalloc(sizeof(struct wl1273_priv), GFP_KERNEL);
+ if (wl1273 == NULL) {
+ dev_err(codec->dev, "Cannot allocate memory.\n");
+ return -ENOMEM;
+ }
+
+ wl1273->mode = WL1273_MODE_BT;
+ wl1273->core = *core;
+
+ snd_soc_codec_set_drvdata(codec, wl1273);
+ mutex_init(&codec->mutex);
+
+ r = snd_soc_add_controls(codec, wl1273_controls,
+ ARRAY_SIZE(wl1273_controls));
+ if (r)
+ kfree(wl1273);
+
+ return r;
+}
+
+static int wl1273_remove(struct snd_soc_codec *codec)
+{
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+ dev_dbg(codec->dev, "%s\n", __func__);
+ kfree(wl1273);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_wl1273 = {
+ .probe = wl1273_probe,
+ .remove = wl1273_remove,
+};
+
+static int __devinit wl1273_platform_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wl1273,
+ &wl1273_dai, 1);
+}
+
+static int __devexit wl1273_platform_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+MODULE_ALIAS("platform:wl1273-codec");
+
+static struct platform_driver wl1273_platform_driver = {
+ .driver = {
+ .name = "wl1273-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wl1273_platform_probe,
+ .remove = __devexit_p(wl1273_platform_remove),
+};
+
+static int __init wl1273_init(void)
+{
+ return platform_driver_register(&wl1273_platform_driver);
+}
+module_init(wl1273_init);
+
+static void __exit wl1273_exit(void)
+{
+ platform_driver_unregister(&wl1273_platform_driver);
+}
+module_exit(wl1273_exit);
+
+MODULE_AUTHOR("Matti Aaltonen <matti.j.aaltonen@nokia.com>");
+MODULE_DESCRIPTION("ASoC WL1273 codec driver");
+MODULE_LICENSE("GPL");
--- /dev/null
+/*
+ * sound/soc/codec/wl1273.h
+ *
+ * ALSA SoC WL1273 codec driver
+ *
+ * Copyright (C) Nokia Corporation
+ * Author: Matti Aaltonen <matti.j.aaltonen@nokia.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __WL1273_CODEC_H__
+#define __WL1273_CODEC_H__
+
+/* I2S protocol, left channel first, data width 16 bits */
+#define WL1273_PCM_DEF_MODE 0x00
+
+/* Rx */
+#define WL1273_AUDIO_ENABLE_I2S (1 << 0)
+#define WL1273_AUDIO_ENABLE_ANALOG (1 << 1)
+
+/* Tx */
+#define WL1273_AUDIO_IO_SET_ANALOG 0
+#define WL1273_AUDIO_IO_SET_I2S 1
+
+#define WL1273_POWER_SET_OFF 0
+#define WL1273_POWER_SET_FM (1 << 0)
+#define WL1273_POWER_SET_RDS (1 << 1)
+#define WL1273_POWER_SET_RETENTION (1 << 4)
+
+#define WL1273_PUPD_SET_OFF 0x00
+#define WL1273_PUPD_SET_ON 0x01
+#define WL1273_PUPD_SET_RETENTION 0x10
+
+/* I2S mode */
+#define WL1273_IS2_WIDTH_32 0x0
+#define WL1273_IS2_WIDTH_40 0x1
+#define WL1273_IS2_WIDTH_22_23 0x2
+#define WL1273_IS2_WIDTH_23_22 0x3
+#define WL1273_IS2_WIDTH_48 0x4
+#define WL1273_IS2_WIDTH_50 0x5
+#define WL1273_IS2_WIDTH_60 0x6
+#define WL1273_IS2_WIDTH_64 0x7
+#define WL1273_IS2_WIDTH_80 0x8
+#define WL1273_IS2_WIDTH_96 0x9
+#define WL1273_IS2_WIDTH_128 0xa
+#define WL1273_IS2_WIDTH 0xf
+
+#define WL1273_IS2_FORMAT_STD (0x0 << 4)
+#define WL1273_IS2_FORMAT_LEFT (0x1 << 4)
+#define WL1273_IS2_FORMAT_RIGHT (0x2 << 4)
+#define WL1273_IS2_FORMAT_USER (0x3 << 4)
+
+#define WL1273_IS2_MASTER (0x0 << 6)
+#define WL1273_IS2_SLAVEW (0x1 << 6)
+
+#define WL1273_IS2_TRI_AFTER_SENDING (0x0 << 7)
+#define WL1273_IS2_TRI_ALWAYS_ACTIVE (0x1 << 7)
+
+#define WL1273_IS2_SDOWS_RR (0x0 << 8)
+#define WL1273_IS2_SDOWS_RF (0x1 << 8)
+#define WL1273_IS2_SDOWS_FR (0x2 << 8)
+#define WL1273_IS2_SDOWS_FF (0x3 << 8)
+
+#define WL1273_IS2_TRI_OPT (0x0 << 10)
+#define WL1273_IS2_TRI_ALWAYS (0x1 << 10)
+
+#define WL1273_IS2_RATE_48K (0x0 << 12)
+#define WL1273_IS2_RATE_44_1K (0x1 << 12)
+#define WL1273_IS2_RATE_32K (0x2 << 12)
+#define WL1273_IS2_RATE_22_05K (0x4 << 12)
+#define WL1273_IS2_RATE_16K (0x5 << 12)
+#define WL1273_IS2_RATE_12K (0x8 << 12)
+#define WL1273_IS2_RATE_11_025 (0x9 << 12)
+#define WL1273_IS2_RATE_8K (0xa << 12)
+#define WL1273_IS2_RATE (0xf << 12)
+
+#define WL1273_I2S_DEF_MODE (WL1273_IS2_WIDTH_32 | \
+ WL1273_IS2_FORMAT_STD | \
+ WL1273_IS2_MASTER | \
+ WL1273_IS2_TRI_AFTER_SENDING | \
+ WL1273_IS2_SDOWS_RR | \
+ WL1273_IS2_TRI_OPT | \
+ WL1273_IS2_RATE_48K)
+
+int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt);
+
+#endif /* End of __WL1273_CODEC_H__ */
};
static struct snd_soc_dai_driver wm8741_dai = {
- .name = "WM8741",
+ .name = "wm8741",
.playback = {
.stream_name = "Playback",
.channels_min = 2, /* Mono modes not yet supported */
bclk_reg = WM8994_AIF1_BCLK;
rate_reg = WM8994_AIF1_RATE;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
- wm8994->lrclk_shared[0])
+ wm8994->lrclk_shared[0]) {
lrclk_reg = WM8994_AIF1DAC_LRCLK;
- else
+ } else {
lrclk_reg = WM8994_AIF1ADC_LRCLK;
+ dev_dbg(codec->dev, "AIF1 using split LRCLK\n");
+ }
break;
case 2:
aif1_reg = WM8994_AIF2_CONTROL_1;
bclk_reg = WM8994_AIF2_BCLK;
rate_reg = WM8994_AIF2_RATE;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
- wm8994->lrclk_shared[1])
+ wm8994->lrclk_shared[1]) {
lrclk_reg = WM8994_AIF2DAC_LRCLK;
- else
+ } else {
lrclk_reg = WM8994_AIF2ADC_LRCLK;
+ dev_dbg(codec->dev, "AIF2 using split LRCLK\n");
+ }
break;
default:
return -EINVAL;
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
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
#include <linux/gfp.h>
#include <linux/of_platform.h>
#include <linux/list.h>
+#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
struct snd_soc_platform_driver dai;
dma_addr_t ssi_stx_phys;
dma_addr_t ssi_srx_phys;
+ unsigned int ssi_fifo_depth;
struct ccsr_dma_channel __iomem *channel;
unsigned int irq;
bool assigned;
unsigned int irq;
struct snd_pcm_substream *substream;
dma_addr_t ssi_sxx_phys;
+ unsigned int ssi_fifo_depth;
dma_addr_t ld_buf_phys;
unsigned int current_link;
dma_addr_t dma_buf_phys;
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;
else
dma_private->ssi_sxx_phys = dma->ssi_srx_phys;
+ dma_private->ssi_fifo_depth = dma->ssi_fifo_depth;
dma_private->dma_channel = dma->channel;
dma_private->irq = dma->irq;
dma_private->substream = substream;
struct device *dev = rtd->platform->dev;
/* Number of bits per sample */
- unsigned int sample_size =
+ unsigned int sample_bits =
snd_pcm_format_physical_width(params_format(hw_params));
/* Number of bytes per frame */
- unsigned int frame_size = 2 * (sample_size / 8);
+ unsigned int sample_bytes = sample_bits / 8;
/* Bus address of SSI STX register */
dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys;
* that offset here. While we're at it, also tell the DMA controller
* how much data to transfer per sample.
*/
- switch (sample_size) {
+ switch (sample_bits) {
case 8:
mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1;
ssi_sxx_phys += 3;
break;
default:
/* We should never get here */
- dev_err(dev, "unsupported sample size %u\n", sample_size);
+ dev_err(dev, "unsupported sample size %u\n", sample_bits);
return -EINVAL;
}
/*
- * BWC should always be a multiple of the frame size. BWC determines
- * how many bytes are sent/received before the DMA controller checks the
- * SSI to see if it needs to stop. For playback, the transmit FIFO can
- * hold three frames, so we want to send two frames at a time. For
- * capture, the receive FIFO is triggered when it contains one frame, so
- * we want to receive one frame at a time.
+ * BWC determines how many bytes are sent/received before the DMA
+ * controller checks the SSI to see if it needs to stop. BWC should
+ * always be a multiple of the frame size, so that we always transmit
+ * whole frames. Each frame occupies two slots in the FIFO. The
+ * parameter for CCSR_DMA_MR_BWC() is rounded down the next power of two
+ * (MR[BWC] can only represent even powers of two).
+ *
+ * To simplify the process, we set BWC to the largest value that is
+ * less than or equal to the FIFO watermark. For playback, this ensures
+ * that we transfer the maximum amount without overrunning the FIFO.
+ * For capture, this ensures that we transfer the maximum amount without
+ * underrunning the FIFO.
+ *
+ * f = SSI FIFO depth
+ * w = SSI watermark value (which equals f - 2)
+ * b = DMA bandwidth count (in bytes)
+ * s = sample size (in bytes, which equals frame_size * 2)
+ *
+ * For playback, we never transmit more than the transmit FIFO
+ * watermark, otherwise we might write more data than the FIFO can hold.
+ * The watermark is equal to the FIFO depth minus two.
+ *
+ * For capture, two equations must hold:
+ * w > f - (b / s)
+ * w >= b / s
+ *
+ * So, b > 2 * s, but b must also be <= s * w. To simplify, we set
+ * b = s * w, which is equal to
+ * (dma_private->ssi_fifo_depth - 2) * sample_bytes.
*/
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- mr |= CCSR_DMA_MR_BWC(2 * frame_size);
- else
- mr |= CCSR_DMA_MR_BWC(frame_size);
+ mr |= CCSR_DMA_MR_BWC((dma_private->ssi_fifo_depth - 2) * sample_bytes);
out_be32(&dma_channel->mr, mr);
.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;
int ret;
/* 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);
- of_node_put(ssi_np);
if (ret) {
- dev_err(&of_dev->dev, "could not determine device resources\n");
+ dev_err(&pdev->dev, "could not determine resources for %s\n",
+ ssi_np->full_name);
+ of_node_put(ssi_np);
return ret;
}
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;
}
dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0);
dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0);
- ret = snd_soc_register_platform(&of_dev->dev, &dma->dai);
+ iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL);
+ if (iprop)
+ dma->ssi_fifo_depth = *iprop;
+ else
+ /* Older 8610 DTs didn't have the fifo-depth property */
+ dma->ssi_fifo_depth = 8;
+
+ of_node_put(ssi_np);
+
+ 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;
}
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);
unsigned int playback;
unsigned int capture;
int asynchronous;
+ unsigned int fifo_depth;
struct snd_soc_dai_driver cpu_dai_drv;
struct device_attribute dev_attr;
struct platform_device *pdev;
/*
* Set the watermark for transmit FIFI 0 and receive FIFO 0. We
- * don't use FIFO 1. Since the SSI only supports stereo, the
- * watermark should never be an odd number.
+ * don't use FIFO 1. We program the transmit water to signal a
+ * DMA transfer if there are only two (or fewer) elements left
+ * in the FIFO. Two elements equals one frame (left channel,
+ * right channel). This value, however, depends on the depth of
+ * the transmit buffer.
+ *
+ * We program the receive FIFO to notify us if at least two
+ * elements (one frame) have been written to the FIFO. We could
+ * make this value larger (and maybe we should), but this way
+ * data will be written to memory as soon as it's available.
*/
out_be32(&ssi->sfcsr,
- CCSR_SSI_SFCSR_TFWM0(6) | CCSR_SSI_SFCSR_RFWM0(2));
+ CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) |
+ CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2));
/*
* We keep the SSI disabled because if we enable it, then the
}
}
-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;
char name[64];
/* 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;
}
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;
}
/* 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;
}
else
ssi_private->cpu_dai_drv.symmetric_rates = 1;
+ /* Determine the FIFO depth. */
+ iprop = of_get_property(np, "fsl,fifo-depth", NULL);
+ if (iprop)
+ ssi_private->fifo_depth = *iprop;
+ else
+ /* Older 8610 DTs didn't have the fifo-depth property */
+ ssi_private->fifo_depth = 8;
+
/* Initialize the the device_attribute structure */
dev_attr = &ssi_private->dev_attr;
dev_attr->attr.name = "statistics";
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;
}
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);
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;
}
#include <linux/module.h>
#include <linux/interrupt.h>
#include <linux/of_device.h>
+#include <linux/slab.h>
#include <sound/soc.h>
#include <asm/fsl_guts.h>
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;
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 */
--- /dev/null
+/**
+ * Freescale P1022DS ALSA SoC Machine driver
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * 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 <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/of_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <asm/fsl_guts.h>
+
+#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 <timur@freescale.com>");
+MODULE_DESCRIPTION("Freescale P1022 DS ALSA SoC machine driver");
+MODULE_LICENSE("GPL v2");
dma_data = &ssi->dma_params_rx;
}
+ if (ssi->flags & IMX_SSI_SYN)
+ reg = SSI_STCCR;
+
snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data);
sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK;
.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",
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
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
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
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);
static struct snd_soc_dai_link imote2_dai = {
.name = "WM8940",
.stream_name = "WM8940",
- .cpu_dai_name = "pxa-i2s",
+ .cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8940-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm8940-codec.0-0034",
{
.name = "uda1380",
.stream_name = "UDA1380 Capture",
- .cpu_dai_name = "pxa-i2s",
+ .cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "uda1380-hifi-capture",
.platform_name = "pxa-pcm-audio",
.codec_name = "uda1380-codec.0-0018",
static struct snd_soc_dai_link poodle_dai = {
.name = "WM8731",
.stream_name = "WM8731",
- .cpu_dai_name = "pxa-i2s",
+ .cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8731-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm8731-codec.0-001a",
struct ssp_priv *priv = snd_soc_dai_get_drvdata(dai);
pxa_ssp_free(priv->ssp);
+ kfree(priv);
return 0;
}
#include <mach/dma.h>
#include <mach/audio.h>
-#include "pxa2xx-pcm.h"
#include "pxa2xx-ac97.h"
static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:pxa2xx-i2s");
--- /dev/null
+/*
+ * saarb.c -- SoC audio for saarb
+ *
+ * Copyright (C) 2010 Marvell International Ltd.
+ * Haojian Zhuang <haojian.zhuang@marvell.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/88pm860x-codec.h"
+#include "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 <haojian.zhuang@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb");
+MODULE_LICENSE("GPL");
--- /dev/null
+/*
+ * tavorevb3.c -- SoC audio for Tavor EVB3
+ *
+ * Copyright (C) 2010 Marvell International Ltd.
+ * Haojian Zhuang <haojian.zhuang@marvell.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/88pm860x-codec.h"
+#include "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 <haojian.zhuang@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3");
+MODULE_LICENSE("GPL");
static struct snd_soc_dai_link z2_dai = {
.name = "wm8750",
.stream_name = "WM8750",
- .cpu_dai_name = "pxa-i2s",
+ .cpu_dai_name = "pxa2xx-i2s",
.codec_dai_name = "wm8750-hifi",
.platform_name = "pxa-pcm-audio",
.codec_name = "wm8750-codec.0-001a",
struct snd_soc_dai *dai;
int i, ret = 0;
- dev_dbg(dev, "dai register %s #%d\n", dev_name(dev), count);
+ dev_dbg(dev, "dai register %s #%Zu\n", dev_name(dev), count);
for (i = 0; i < count; i++) {
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);
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);
git://bbs.cooldavid.org / net-next-2.6.git / commitdiff