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Diffstat (limited to 'ANDROID_3.4.5/sound/soc/codecs/ak4642.c')
-rw-r--r--ANDROID_3.4.5/sound/soc/codecs/ak4642.c590
1 files changed, 0 insertions, 590 deletions
diff --git a/ANDROID_3.4.5/sound/soc/codecs/ak4642.c b/ANDROID_3.4.5/sound/soc/codecs/ak4642.c
deleted file mode 100644
index b3e24f28..00000000
--- a/ANDROID_3.4.5/sound/soc/codecs/ak4642.c
+++ /dev/null
@@ -1,590 +0,0 @@
-/*
- * ak4642.c -- AK4642/AK4643 ALSA Soc Audio driver
- *
- * Copyright (C) 2009 Renesas Solutions Corp.
- * Kuninori Morimoto <morimoto.kuninori@renesas.com>
- *
- * Based on wm8731.c by Richard Purdie
- * Based on ak4535.c by Richard Purdie
- * Based on wm8753.c by Liam Girdwood
- *
- * 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.
- */
-
-/* ** CAUTION **
- *
- * This is very simple driver.
- * It can use headphone output / stereo input only
- *
- * AK4642 is tested.
- * AK4643 is tested.
- * AK4648 is tested.
- */
-
-#include <linux/delay.h>
-#include <linux/i2c.h>
-#include <linux/slab.h>
-#include <linux/module.h>
-#include <sound/soc.h>
-#include <sound/initval.h>
-#include <sound/tlv.h>
-
-#define PW_MGMT1 0x00
-#define PW_MGMT2 0x01
-#define SG_SL1 0x02
-#define SG_SL2 0x03
-#define MD_CTL1 0x04
-#define MD_CTL2 0x05
-#define TIMER 0x06
-#define ALC_CTL1 0x07
-#define ALC_CTL2 0x08
-#define L_IVC 0x09
-#define L_DVC 0x0a
-#define ALC_CTL3 0x0b
-#define R_IVC 0x0c
-#define R_DVC 0x0d
-#define MD_CTL3 0x0e
-#define MD_CTL4 0x0f
-#define PW_MGMT3 0x10
-#define DF_S 0x11
-#define FIL3_0 0x12
-#define FIL3_1 0x13
-#define FIL3_2 0x14
-#define FIL3_3 0x15
-#define EQ_0 0x16
-#define EQ_1 0x17
-#define EQ_2 0x18
-#define EQ_3 0x19
-#define EQ_4 0x1a
-#define EQ_5 0x1b
-#define FIL1_0 0x1c
-#define FIL1_1 0x1d
-#define FIL1_2 0x1e
-#define FIL1_3 0x1f
-#define PW_MGMT4 0x20
-#define MD_CTL5 0x21
-#define LO_MS 0x22
-#define HP_MS 0x23
-#define SPK_MS 0x24
-
-/* PW_MGMT1*/
-#define PMVCM (1 << 6) /* VCOM Power Management */
-#define PMMIN (1 << 5) /* MIN Input Power Management */
-#define PMDAC (1 << 2) /* DAC Power Management */
-#define PMADL (1 << 0) /* MIC Amp Lch and ADC Lch Power Management */
-
-/* PW_MGMT2 */
-#define HPMTN (1 << 6)
-#define PMHPL (1 << 5)
-#define PMHPR (1 << 4)
-#define MS (1 << 3) /* master/slave select */
-#define MCKO (1 << 1)
-#define PMPLL (1 << 0)
-
-#define PMHP_MASK (PMHPL | PMHPR)
-#define PMHP PMHP_MASK
-
-/* PW_MGMT3 */
-#define PMADR (1 << 0) /* MIC L / ADC R Power Management */
-
-/* SG_SL1 */
-#define MINS (1 << 6) /* Switch from MIN to Speaker */
-#define DACL (1 << 4) /* Switch from DAC to Stereo or Receiver */
-#define PMMP (1 << 2) /* MPWR pin Power Management */
-#define MGAIN0 (1 << 0) /* MIC amp gain*/
-
-/* TIMER */
-#define ZTM(param) ((param & 0x3) << 4) /* ALC Zoro Crossing TimeOut */
-#define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2))
-
-/* ALC_CTL1 */
-#define ALC (1 << 5) /* ALC Enable */
-#define LMTH0 (1 << 0) /* ALC Limiter / Recovery Level */
-
-/* MD_CTL1 */
-#define PLL3 (1 << 7)
-#define PLL2 (1 << 6)
-#define PLL1 (1 << 5)
-#define PLL0 (1 << 4)
-#define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0)
-
-#define BCKO_MASK (1 << 3)
-#define BCKO_64 BCKO_MASK
-
-#define DIF_MASK (3 << 0)
-#define DSP (0 << 0)
-#define RIGHT_J (1 << 0)
-#define LEFT_J (2 << 0)
-#define I2S (3 << 0)
-
-/* MD_CTL2 */
-#define FS0 (1 << 0)
-#define FS1 (1 << 1)
-#define FS2 (1 << 2)
-#define FS3 (1 << 5)
-#define FS_MASK (FS0 | FS1 | FS2 | FS3)
-
-/* MD_CTL3 */
-#define BST1 (1 << 3)
-
-/* MD_CTL4 */
-#define DACH (1 << 0)
-
-/*
- * Playback Volume (table 39)
- *
- * max : 0x00 : +12.0 dB
- * ( 0.5 dB step )
- * min : 0xFE : -115.0 dB
- * mute: 0xFF
- */
-static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1);
-
-static const struct snd_kcontrol_new ak4642_snd_controls[] = {
-
- SOC_DOUBLE_R_TLV("Digital Playback Volume", L_DVC, R_DVC,
- 0, 0xFF, 1, out_tlv),
-};
-
-static const struct snd_kcontrol_new ak4642_headphone_control =
- SOC_DAPM_SINGLE("Switch", PW_MGMT2, 6, 1, 0);
-
-static const struct snd_kcontrol_new ak4642_lout_mixer_controls[] = {
- SOC_DAPM_SINGLE("DACL", SG_SL1, 4, 1, 0),
-};
-
-static const struct snd_soc_dapm_widget ak4642_dapm_widgets[] = {
-
- /* Outputs */
- SND_SOC_DAPM_OUTPUT("HPOUTL"),
- SND_SOC_DAPM_OUTPUT("HPOUTR"),
- SND_SOC_DAPM_OUTPUT("LINEOUT"),
-
- SND_SOC_DAPM_PGA("HPL Out", PW_MGMT2, 5, 0, NULL, 0),
- SND_SOC_DAPM_PGA("HPR Out", PW_MGMT2, 4, 0, NULL, 0),
- SND_SOC_DAPM_SWITCH("Headphone Enable", SND_SOC_NOPM, 0, 0,
- &ak4642_headphone_control),
-
- SND_SOC_DAPM_PGA("DACH", MD_CTL4, 0, 0, NULL, 0),
-
- SND_SOC_DAPM_MIXER("LINEOUT Mixer", PW_MGMT1, 3, 0,
- &ak4642_lout_mixer_controls[0],
- ARRAY_SIZE(ak4642_lout_mixer_controls)),
-
- /* DAC */
- SND_SOC_DAPM_DAC("DAC", "HiFi Playback", PW_MGMT1, 2, 0),
-};
-
-static const struct snd_soc_dapm_route ak4642_intercon[] = {
-
- /* Outputs */
- {"HPOUTL", NULL, "HPL Out"},
- {"HPOUTR", NULL, "HPR Out"},
- {"LINEOUT", NULL, "LINEOUT Mixer"},
-
- {"HPL Out", NULL, "Headphone Enable"},
- {"HPR Out", NULL, "Headphone Enable"},
-
- {"Headphone Enable", "Switch", "DACH"},
-
- {"DACH", NULL, "DAC"},
-
- {"LINEOUT Mixer", "DACL", "DAC"},
-};
-
-/* codec private data */
-struct ak4642_priv {
- unsigned int sysclk;
- enum snd_soc_control_type control_type;
-};
-
-/*
- * ak4642 register cache
- */
-static const u8 ak4642_reg[] = {
- 0x00, 0x00, 0x01, 0x00,
- 0x02, 0x00, 0x00, 0x00,
- 0xe1, 0xe1, 0x18, 0x00,
- 0xe1, 0x18, 0x11, 0x08,
- 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
- 0x00,
-};
-
-static const u8 ak4648_reg[] = {
- 0x00, 0x00, 0x01, 0x00,
- 0x02, 0x00, 0x00, 0x00,
- 0xe1, 0xe1, 0x18, 0x00,
- 0xe1, 0x18, 0x11, 0xb8,
- 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x00, 0x00, 0x00,
- 0x00, 0x88, 0x88, 0x08,
-};
-
-static int ak4642_dai_startup(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
- struct snd_soc_codec *codec = dai->codec;
-
- if (is_play) {
- /*
- * start headphone output
- *
- * PLL, Master Mode
- * Audio I/F Format :MSB justified (ADC & DAC)
- * Bass Boost Level : Middle
- *
- * This operation came from example code of
- * "ASAHI KASEI AK4642" (japanese) manual p97.
- */
- snd_soc_write(codec, L_IVC, 0x91); /* volume */
- snd_soc_write(codec, R_IVC, 0x91); /* volume */
- } else {
- /*
- * start stereo input
- *
- * PLL Master Mode
- * Audio I/F Format:MSB justified (ADC & DAC)
- * Pre MIC AMP:+20dB
- * MIC Power On
- * ALC setting:Refer to Table 35
- * ALC bit=“1”
- *
- * This operation came from example code of
- * "ASAHI KASEI AK4642" (japanese) manual p94.
- */
- snd_soc_write(codec, SG_SL1, PMMP | MGAIN0);
- snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3));
- snd_soc_write(codec, ALC_CTL1, ALC | LMTH0);
- snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL);
- snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR);
- }
-
- return 0;
-}
-
-static void ak4642_dai_shutdown(struct snd_pcm_substream *substream,
- struct snd_soc_dai *dai)
-{
- int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
- struct snd_soc_codec *codec = dai->codec;
-
- if (is_play) {
- } else {
- /* stop stereo input */
- snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0);
- snd_soc_update_bits(codec, PW_MGMT3, PMADR, 0);
- snd_soc_update_bits(codec, ALC_CTL1, ALC, 0);
- }
-}
-
-static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
- int clk_id, unsigned int freq, int dir)
-{
- struct snd_soc_codec *codec = codec_dai->codec;
- u8 pll;
-
- switch (freq) {
- case 11289600:
- pll = PLL2;
- break;
- case 12288000:
- pll = PLL2 | PLL0;
- break;
- case 12000000:
- pll = PLL2 | PLL1;
- break;
- case 24000000:
- pll = PLL2 | PLL1 | PLL0;
- break;
- case 13500000:
- pll = PLL3 | PLL2;
- break;
- case 27000000:
- pll = PLL3 | PLL2 | PLL0;
- break;
- default:
- return -EINVAL;
- }
- snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll);
-
- return 0;
-}
-
-static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
-{
- struct snd_soc_codec *codec = dai->codec;
- u8 data;
- u8 bcko;
-
- data = MCKO | PMPLL; /* use MCKO */
- bcko = 0;
-
- /* set master/slave audio interface */
- switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
- case SND_SOC_DAIFMT_CBM_CFM:
- data |= MS;
- bcko = BCKO_64;
- break;
- case SND_SOC_DAIFMT_CBS_CFS:
- break;
- default:
- return -EINVAL;
- }
- snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data);
- snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
-
- /* format type */
- data = 0;
- switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
- case SND_SOC_DAIFMT_LEFT_J:
- data = LEFT_J;
- break;
- case SND_SOC_DAIFMT_I2S:
- data = I2S;
- break;
- /* FIXME
- * Please add RIGHT_J / DSP support here
- */
- default:
- return -EINVAL;
- break;
- }
- snd_soc_update_bits(codec, MD_CTL1, DIF_MASK, data);
-
- return 0;
-}
-
-static int ak4642_dai_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;
- u8 rate;
-
- switch (params_rate(params)) {
- case 7350:
- rate = FS2;
- break;
- case 8000:
- rate = 0;
- break;
- case 11025:
- rate = FS2 | FS0;
- break;
- case 12000:
- rate = FS0;
- break;
- case 14700:
- rate = FS2 | FS1;
- break;
- case 16000:
- rate = FS1;
- break;
- case 22050:
- rate = FS2 | FS1 | FS0;
- break;
- case 24000:
- rate = FS1 | FS0;
- break;
- case 29400:
- rate = FS3 | FS2 | FS1;
- break;
- case 32000:
- rate = FS3 | FS1;
- break;
- case 44100:
- rate = FS3 | FS2 | FS1 | FS0;
- break;
- case 48000:
- rate = FS3 | FS1 | FS0;
- break;
- default:
- return -EINVAL;
- break;
- }
- snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
-
- return 0;
-}
-
-static int ak4642_set_bias_level(struct snd_soc_codec *codec,
- enum snd_soc_bias_level level)
-{
- switch (level) {
- case SND_SOC_BIAS_OFF:
- snd_soc_write(codec, PW_MGMT1, 0x00);
- break;
- default:
- snd_soc_update_bits(codec, PW_MGMT1, PMVCM, PMVCM);
- break;
- }
- codec->dapm.bias_level = level;
-
- return 0;
-}
-
-static const struct snd_soc_dai_ops ak4642_dai_ops = {
- .startup = ak4642_dai_startup,
- .shutdown = ak4642_dai_shutdown,
- .set_sysclk = ak4642_dai_set_sysclk,
- .set_fmt = ak4642_dai_set_fmt,
- .hw_params = ak4642_dai_hw_params,
-};
-
-static struct snd_soc_dai_driver ak4642_dai = {
- .name = "ak4642-hifi",
- .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 = &ak4642_dai_ops,
- .symmetric_rates = 1,
-};
-
-static int ak4642_resume(struct snd_soc_codec *codec)
-{
- snd_soc_cache_sync(codec);
- return 0;
-}
-
-
-static int ak4642_probe(struct snd_soc_codec *codec)
-{
- struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec);
- int ret;
-
- ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4642->control_type);
- if (ret < 0) {
- dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
- return ret;
- }
-
- snd_soc_add_codec_controls(codec, ak4642_snd_controls,
- ARRAY_SIZE(ak4642_snd_controls));
-
- ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
- return 0;
-}
-
-static int ak4642_remove(struct snd_soc_codec *codec)
-{
- ak4642_set_bias_level(codec, SND_SOC_BIAS_OFF);
- return 0;
-}
-
-static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
- .probe = ak4642_probe,
- .remove = ak4642_remove,
- .resume = ak4642_resume,
- .set_bias_level = ak4642_set_bias_level,
- .reg_cache_default = ak4642_reg, /* ak4642 reg */
- .reg_cache_size = ARRAY_SIZE(ak4642_reg), /* ak4642 reg */
- .reg_word_size = sizeof(u8),
- .dapm_widgets = ak4642_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets),
- .dapm_routes = ak4642_intercon,
- .num_dapm_routes = ARRAY_SIZE(ak4642_intercon),
-};
-
-static struct snd_soc_codec_driver soc_codec_dev_ak4648 = {
- .probe = ak4642_probe,
- .remove = ak4642_remove,
- .resume = ak4642_resume,
- .set_bias_level = ak4642_set_bias_level,
- .reg_cache_default = ak4648_reg, /* ak4648 reg */
- .reg_cache_size = ARRAY_SIZE(ak4648_reg), /* ak4648 reg */
- .reg_word_size = sizeof(u8),
- .dapm_widgets = ak4642_dapm_widgets,
- .num_dapm_widgets = ARRAY_SIZE(ak4642_dapm_widgets),
- .dapm_routes = ak4642_intercon,
- .num_dapm_routes = ARRAY_SIZE(ak4642_intercon),
-};
-
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-static __devinit int ak4642_i2c_probe(struct i2c_client *i2c,
- const struct i2c_device_id *id)
-{
- struct ak4642_priv *ak4642;
- int ret;
-
- ak4642 = devm_kzalloc(&i2c->dev, sizeof(struct ak4642_priv),
- GFP_KERNEL);
- if (!ak4642)
- return -ENOMEM;
-
- i2c_set_clientdata(i2c, ak4642);
- ak4642->control_type = SND_SOC_I2C;
-
- ret = snd_soc_register_codec(&i2c->dev,
- (struct snd_soc_codec_driver *)id->driver_data,
- &ak4642_dai, 1);
- return ret;
-}
-
-static __devexit int ak4642_i2c_remove(struct i2c_client *client)
-{
- snd_soc_unregister_codec(&client->dev);
- return 0;
-}
-
-static const struct i2c_device_id ak4642_i2c_id[] = {
- { "ak4642", (kernel_ulong_t)&soc_codec_dev_ak4642 },
- { "ak4643", (kernel_ulong_t)&soc_codec_dev_ak4642 },
- { "ak4648", (kernel_ulong_t)&soc_codec_dev_ak4648 },
- { }
-};
-MODULE_DEVICE_TABLE(i2c, ak4642_i2c_id);
-
-static struct i2c_driver ak4642_i2c_driver = {
- .driver = {
- .name = "ak4642-codec",
- .owner = THIS_MODULE,
- },
- .probe = ak4642_i2c_probe,
- .remove = __devexit_p(ak4642_i2c_remove),
- .id_table = ak4642_i2c_id,
-};
-#endif
-
-static int __init ak4642_modinit(void)
-{
- int ret = 0;
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- ret = i2c_add_driver(&ak4642_i2c_driver);
-#endif
- return ret;
-
-}
-module_init(ak4642_modinit);
-
-static void __exit ak4642_exit(void)
-{
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
- i2c_del_driver(&ak4642_i2c_driver);
-#endif
-
-}
-module_exit(ak4642_exit);
-
-MODULE_DESCRIPTION("Soc AK4642 driver");
-MODULE_AUTHOR("Kuninori Morimoto <morimoto.kuninori@renesas.com>");
-MODULE_LICENSE("GPL");
generated by cgit (git 2.34.1) at 2024-12-21 17:46:46 +0000