diff options
Diffstat (limited to 'sound/soc')
33 files changed, 2590 insertions, 110 deletions
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 27658521516..a3b51df2bea 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -29,6 +29,7 @@ source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" source "sound/soc/sh/Kconfig" source "sound/soc/fsl/Kconfig" +source "sound/soc/davinci/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 4869c9ae7a0..e489dbdde45 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,4 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 898a7d36328..3903ab7dfa4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -18,6 +18,10 @@ config SND_SOC_WM9712 tristate depends on SND_SOC +config SND_SOC_WM9713 + tristate + depends on SND_SOC + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index c6e5338c266..4e1314c9d3e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,7 @@ snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm9712-objs := wm9712.o +snd-soc-wm9713-objs := wm9713.o snd-soc-cs4270-objs := cs4270.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -11,5 +12,6 @@ obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o +obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 242130cf1ab..2a1ffe39690 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -40,7 +40,8 @@ static int ac97_prepare(struct snd_pcm_substream *substream) } #define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) struct snd_soc_codec_dai ac97_dai = { .name = "AC97 HiFi", @@ -86,7 +87,7 @@ static int ac97_soc_probe(struct platform_device *pdev) printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION); socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); - if (socdev->codec == NULL) + if (!socdev->codec) return -ENOMEM; codec = socdev->codec; mutex_init(&codec->mutex); @@ -102,17 +103,17 @@ static int ac97_soc_probe(struct platform_device *pdev) /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); - if(ret < 0) + if (ret < 0) goto err; /* add codec as bus device for standard ac97 */ ret = snd_ac97_bus(codec->card, 0, &soc_ac97_ops, NULL, &ac97_bus); - if(ret < 0) + if (ret < 0) goto bus_err; memset(&ac97_template, 0, sizeof(struct snd_ac97_template)); ret = snd_ac97_mixer(ac97_bus, &ac97_template, &codec->ac97); - if(ret < 0) + if (ret < 0) goto bus_err; ret = snd_soc_register_card(socdev); @@ -135,7 +136,7 @@ static int ac97_soc_remove(struct platform_device *pdev) struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->codec; - if(codec == NULL) + if (!codec) return 0; snd_soc_free_pcms(socdev); @@ -145,11 +146,10 @@ static int ac97_soc_remove(struct platform_device *pdev) return 0; } -struct snd_soc_codec_device soc_codec_dev_ac97= { +struct snd_soc_codec_device soc_codec_dev_ac97 = { .probe = ac97_soc_probe, .remove = ac97_soc_remove, }; - EXPORT_SYMBOL_GPL(soc_codec_dev_ac97); MODULE_DESCRIPTION("Soc Generic AC97 driver"); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index bf2ab72d49b..e73fcfd9f5c 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -372,7 +372,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, struct snd_soc_device *socdev = rtd->socdev; struct snd_soc_codec *codec = socdev->codec; struct cs4270_private *cs4270 = codec->private_data; - unsigned int ret = 0; + int ret; unsigned int i; unsigned int rate; unsigned int ratio; diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 889a897d41a..630684f4a0b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -660,33 +660,53 @@ struct aic3x_rate_divs { /* AIC3X codec mclk clock divider coefficients */ static const struct aic3x_rate_divs aic3x_divs[] = { /* 8k */ + {12000000, 8000, 48000, 0xa, 16, 3840}, + {19200000, 8000, 48000, 0xa, 10, 2400}, {22579200, 8000, 48000, 0xa, 8, 7075}, {33868800, 8000, 48000, 0xa, 5, 8049}, /* 11.025k */ + {12000000, 11025, 44100, 0x6, 15, 528}, + {19200000, 11025, 44100, 0x6, 9, 4080}, {22579200, 11025, 44100, 0x6, 8, 0}, {33868800, 11025, 44100, 0x6, 5, 3333}, /* 16k */ + {12000000, 16000, 48000, 0x4, 16, 3840}, + {19200000, 16000, 48000, 0x4, 10, 2400}, {22579200, 16000, 48000, 0x4, 8, 7075}, {33868800, 16000, 48000, 0x4, 5, 8049}, /* 22.05k */ + {12000000, 22050, 44100, 0x2, 15, 528}, + {19200000, 22050, 44100, 0x2, 9, 4080}, {22579200, 22050, 44100, 0x2, 8, 0}, {33868800, 22050, 44100, 0x2, 5, 3333}, /* 32k */ + {12000000, 32000, 48000, 0x1, 16, 3840}, + {19200000, 32000, 48000, 0x1, 10, 2400}, {22579200, 32000, 48000, 0x1, 8, 7075}, {33868800, 32000, 48000, 0x1, 5, 8049}, /* 44.1k */ + {12000000, 44100, 44100, 0x0, 15, 528}, + {19200000, 44100, 44100, 0x0, 9, 4080}, {22579200, 44100, 44100, 0x0, 8, 0}, {33868800, 44100, 44100, 0x0, 5, 3333}, /* 48k */ + {12000000, 48000, 48000, 0x0, 16, 3840}, + {19200000, 48000, 48000, 0x0, 10, 2400}, {22579200, 48000, 48000, 0x0, 8, 7075}, {33868800, 48000, 48000, 0x0, 5, 8049}, /* 64k */ + {12000000, 64000, 96000, 0x1, 16, 3840}, + {19200000, 64000, 96000, 0x1, 10, 2400}, {22579200, 64000, 96000, 0x1, 8, 7075}, {33868800, 64000, 96000, 0x1, 5, 8049}, /* 88.2k */ + {12000000, 88200, 88200, 0x0, 15, 528}, + {19200000, 88200, 88200, 0x0, 9, 4080}, {22579200, 88200, 88200, 0x0, 8, 0}, {33868800, 88200, 88200, 0x0, 5, 3333}, /* 96k */ + {12000000, 96000, 96000, 0x0, 16, 3840}, + {19200000, 96000, 96000, 0x0, 10, 2400}, {22579200, 96000, 96000, 0x0, 8, 7075}, {33868800, 96000, 96000, 0x0, 5, 8049}, }; @@ -807,6 +827,8 @@ static int aic3x_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, struct aic3x_priv *aic3x = codec->private_data; switch (freq) { + case 12000000: + case 19200000: case 22579200: case 33868800: aic3x->sysclk = freq; diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 9c33fe87492..0cf9265fca8 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -110,7 +110,7 @@ static int wm8731_write(struct snd_soc_codec *codec, unsigned int reg, data[0] = (reg << 1) | ((value >> 8) & 0x0001); data[1] = value & 0x00ff; - wm8731_write_reg_cache (codec, reg, value); + wm8731_write_reg_cache(codec, reg, value); if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; else @@ -154,8 +154,10 @@ static int wm8731_add_controls(struct snd_soc_codec *codec) int err, i; for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) { - if ((err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8731_snd_controls[i],codec, NULL))) < 0) + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm8731_snd_controls[i], + codec, NULL)); + if (err < 0) return err; } @@ -221,15 +223,13 @@ static int wm8731_add_widgets(struct snd_soc_codec *codec) { int i; - for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) { + for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); - } /* set up audio path interconnects */ - for(i = 0; intercon[i][0] != NULL; i++) { + for (i = 0; intercon[i][0] != NULL; i++) snd_soc_dapm_connect_input(codec, intercon[i][0], intercon[i][1], intercon[i][2]); - } snd_soc_dapm_new_widgets(codec); return 0; @@ -589,7 +589,7 @@ pcm_err: static struct snd_soc_device *wm8731_socdev; -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * WM8731 2 wire address is determined by GPIO5 @@ -651,7 +651,7 @@ err: static int wm8731_i2c_detach(struct i2c_client *client) { - struct snd_soc_codec* codec = i2c_get_clientdata(client); + struct snd_soc_codec *codec = i2c_get_clientdata(client); i2c_detach_client(client); kfree(codec->reg_cache); kfree(client); @@ -709,7 +709,7 @@ static int wm8731_probe(struct platform_device *pdev) INIT_LIST_HEAD(&codec->dapm_paths); wm8731_socdev = socdev; -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; @@ -734,7 +734,7 @@ static int wm8731_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8731_i2c_driver); #endif kfree(codec->private_data); @@ -749,7 +749,6 @@ struct snd_soc_codec_device soc_codec_dev_wm8731 = { .suspend = wm8731_suspend, .resume = wm8731_resume, }; - EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731); MODULE_DESCRIPTION("ASoC WM8731 driver"); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 77a857b997a..16cd5d4d5ad 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -110,7 +110,7 @@ static int wm8750_write(struct snd_soc_codec *codec, unsigned int reg, data[0] = (reg << 1) | ((value >> 8) & 0x0001); data[1] = value & 0x00ff; - wm8750_write_reg_cache (codec, reg, value); + wm8750_write_reg_cache(codec, reg, value); if (codec->hw_write(codec->control_data, data, 2) == 2) return 0; else @@ -257,7 +257,8 @@ static int wm8750_add_controls(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) { err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8750_snd_controls[i],codec, NULL)); + snd_soc_cnew(&wm8750_snd_controls[i], + codec, NULL)); if (err < 0) return err; } @@ -478,15 +479,13 @@ static int wm8750_add_widgets(struct snd_soc_codec *codec) { int i; - for(i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) { + for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); - } /* set up audio path audio_mapnects */ - for(i = 0; audio_map[i][0] != NULL; i++) { + for (i = 0; audio_map[i][0] != NULL; i++) snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); - } snd_soc_dapm_new_widgets(codec); return 0; @@ -714,8 +713,8 @@ static int wm8750_dapm_event(struct snd_soc_codec *codec, int event) } #define WM8750_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) @@ -784,7 +783,8 @@ static int wm8750_resume(struct platform_device *pdev) if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) { wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2); codec->dapm_state = SNDRV_CTL_POWER_D0; - schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000)); + schedule_delayed_work(&codec->delayed_work, + msecs_to_jiffies(1000)); } return 0; @@ -864,7 +864,7 @@ pcm_err: around */ static struct snd_soc_device *wm8750_socdev; -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) /* * WM8731 2 wire address is determined by GPIO5 @@ -979,8 +979,8 @@ static int wm8750_probe(struct platform_device *pdev) INIT_LIST_HEAD(&codec->dapm_paths); wm8750_socdev = socdev; INIT_DELAYED_WORK(&codec->delayed_work, wm8750_work); - -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) if (setup->i2c_address) { normal_i2c[0] = setup->i2c_address; codec->hw_write = (hw_write_t)i2c_master_send; @@ -1025,7 +1025,7 @@ static int wm8750_remove(struct platform_device *pdev) run_delayed_work(&codec->delayed_work); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); -#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE) +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) i2c_del_driver(&wm8750_i2c_driver); #endif kfree(codec->private_data); @@ -1040,7 +1040,6 @@ struct snd_soc_codec_device soc_codec_dev_wm8750 = { .suspend = wm8750_suspend, .resume = wm8750_resume, }; - EXPORT_SYMBOL_GPL(soc_codec_dev_wm8750); MODULE_DESCRIPTION("ASoC WM8750 driver"); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index ddd9c71b3fd..76a5c7b05df 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -198,6 +198,7 @@ static const char *wm8753_mic_sel[] = {"Mic 1", "Mic 2", "Mic 3"}; static const char *wm8753_dai_mode[] = {"DAI 0", "DAI 1", "DAI 2", "DAI 3"}; static const char *wm8753_dat_sel[] = {"Stereo", "Left ADC", "Right ADC", "Channel Swap"}; +static const char *wm8753_rout2_phase[] = {"Non Inverted", "Inverted"}; static const struct soc_enum wm8753_enum[] = { SOC_ENUM_SINGLE(WM8753_BASS, 7, 2, wm8753_base), @@ -228,6 +229,7 @@ SOC_ENUM_SINGLE(WM8753_ADC, 4, 2, wm8753_adc_filter), SOC_ENUM_SINGLE(WM8753_MICBIAS, 6, 3, wm8753_mic_sel), SOC_ENUM_SINGLE(WM8753_IOCTL, 2, 4, wm8753_dai_mode), SOC_ENUM_SINGLE(WM8753_ADC, 7, 4, wm8753_dat_sel), +SOC_ENUM_SINGLE(WM8753_OUTCTL, 2, 2, wm8753_rout2_phase), }; @@ -279,7 +281,7 @@ SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7, 1, 0 SOC_SINGLE("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1), SOC_SINGLE("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1), -SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 4, 7, 1), +SOC_SINGLE("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1), SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0), SOC_ENUM("Bass Boost", wm8753_enum[0]), @@ -330,6 +332,7 @@ SOC_SINGLE("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0), SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai), SOC_ENUM("ADC Data Select", wm8753_enum[27]), +SOC_ENUM("ROUT2 Phase", wm8753_enum[28]), }; /* add non dapm controls */ diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 524f7450804..d2d79e182a4 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -581,22 +581,14 @@ static int wm9712_dapm_event(struct snd_soc_codec *codec, int event) switch (event) { case SNDRV_CTL_POWER_D0: /* full On */ - /* liam - maybe enable thermal shutdown */ - reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xdfff; - ac97_write(codec, AC97_EXTENDED_MID, reg); - break; case SNDRV_CTL_POWER_D1: /* partial On */ case SNDRV_CTL_POWER_D2: /* partial On */ break; case SNDRV_CTL_POWER_D3hot: /* Off, with power */ - /* enable master bias and vmid */ - reg = ac97_read(codec, AC97_EXTENDED_MID) & 0xbbff; - ac97_write(codec, AC97_EXTENDED_MID, reg); ac97_write(codec, AC97_POWERDOWN, 0x0000); break; case SNDRV_CTL_POWER_D3cold: /* Off, without power */ /* disable everything including AC link */ - ac97_write(codec, AC97_EXTENDED_MID, 0xffff); ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); ac97_write(codec, AC97_POWERDOWN, 0xffff); break; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c new file mode 100644 index 00000000000..1f241161445 --- /dev/null +++ b/sound/soc/codecs/wm9713.c @@ -0,0 +1,1300 @@ +/* + * wm9713.c -- ALSA Soc WM9713 codec support + * + * Copyright 2006 Wolfson Microelectronics PLC. + * Author: Liam Girdwood + * liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + * Revision history + * 4th Feb 2006 Initial version. + * + * Features:- + * + * o Support for AC97 Codec, Voice DAC and Aux DAC + * o Support for DAPM + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "wm9713.h" + +#define WM9713_VERSION "0.15" + +struct wm9713_priv { + u32 pll_in; /* PLL input frequency */ + u32 pll_out; /* PLL output frequency */ +}; + +static unsigned int ac97_read(struct snd_soc_codec *codec, + unsigned int reg); +static int ac97_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int val); + +/* + * WM9713 register cache + * Reg 0x3c bit 15 is used by touch driver. + */ +static const u16 wm9713_reg[] = { + 0x6174, 0x8080, 0x8080, 0x8080, + 0xc880, 0xe808, 0xe808, 0x0808, + 0x00da, 0x8000, 0xd600, 0xaaa0, + 0xaaa0, 0xaaa0, 0x0000, 0x0000, + 0x0f0f, 0x0040, 0x0000, 0x7f00, + 0x0405, 0x0410, 0xbb80, 0xbb80, + 0x0000, 0xbb80, 0x0000, 0x4523, + 0x0000, 0x2000, 0x7eff, 0xffff, + 0x0000, 0x0000, 0x0080, 0x0000, + 0x0000, 0x0000, 0xfffe, 0xffff, + 0x0000, 0x0000, 0x0000, 0xfffe, + 0x4000, 0x0000, 0x0000, 0x0000, + 0xb032, 0x3e00, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0006, + 0x0001, 0x0000, 0x574d, 0x4c13, + 0x0000, 0x0000, 0x0000 +}; + +/* virtual HP mixers regs */ +#define HPL_MIXER 0x80 +#define HPR_MIXER 0x82 +#define MICB_MUX 0x82 + +static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"}; +static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"}; +static const char *wm9713_rec_src[] = + {"Mic 1", "Mic 2", "Line", "Mono In", "Headphone", "Speaker", + "Mono Out", "Zh"}; +static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"}; +static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"}; +static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv", + "Mono Vmid", "Inv Vmid"}; +static const char *wm9713_spk_pga[] = + {"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid", + "Speaker Vmid", "Inv Vmid"}; +static const char *wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone", + "Headphone Vmid"}; +static const char *wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "Inv 1 Vmid"}; +static const char *wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "Inv 2 Vmid"}; +static const char *wm9713_dac_inv[] = + {"Off", "Mono", "Speaker", "Left Headphone", "Right Headphone", + "Headphone Mono", "NC", "Vmid"}; +static const char *wm9713_bass[] = {"Linear Control", "Adaptive Boost"}; +static const char *wm9713_ng_type[] = {"Constant Gain", "Mute"}; +static const char *wm9713_mic_select[] = {"Mic 1", "Mic 2 A", "Mic 2 B"}; +static const char *wm9713_micb_select[] = {"MPB", "MPA"}; + +static const struct soc_enum wm9713_enum[] = { +SOC_ENUM_SINGLE(AC97_LINE, 3, 4, wm9713_mic_mixer), /* record mic mixer 0 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 14, 4, wm9713_rec_mux), /* record mux hp 1 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 9, 4, wm9713_rec_mux), /* record mux mono 2 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 3, 8, wm9713_rec_src), /* record mux left 3 */ +SOC_ENUM_SINGLE(AC97_VIDEO, 0, 8, wm9713_rec_src), /* record mux right 4*/ +SOC_ENUM_DOUBLE(AC97_CD, 14, 6, 2, wm9713_rec_gain), /* record step size 5 */ +SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9713_alc_select), /* alc source select 6*/ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 14, 4, wm9713_mono_pga), /* mono input select 7 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 11, 8, wm9713_spk_pga), /* speaker left input select 8 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 8, 8, wm9713_spk_pga), /* speaker right input select 9 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 6, 3, wm9713_hp_pga), /* headphone left input 10 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 4, 3, wm9713_hp_pga), /* headphone right input 11 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 2, 4, wm9713_out3_pga), /* out 3 source 12 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN, 0, 4, wm9713_out4_pga), /* out 4 source 13 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 13, 8, wm9713_dac_inv), /* dac invert 1 14 */ +SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 10, 8, wm9713_dac_inv), /* dac invert 2 15 */ +SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_bass), /* bass control 16 */ +SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), /* noise gate type 17 */ +SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 */ +SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */ +}; + +static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = { +SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), +SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1), +SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), +SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 7, 1, 1), +SOC_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1), +SOC_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1), +SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1), +SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1), + +SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0), +SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1), + +SOC_SINGLE("Capture Switch", AC97_CD, 15, 1, 1), +SOC_ENUM("Capture Volume Steps", wm9713_enum[5]), +SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 31, 0), +SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0), + +SOC_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1), +SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0), +SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0), + +SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), +SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), +SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0), +SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0), +SOC_ENUM("ALC Function", wm9713_enum[6]), +SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0), +SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 0), +SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0), +SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0), +SOC_ENUM("ALC NG Type", wm9713_enum[17]), +SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 0), + +SOC_DOUBLE("Speaker Playback ZC Switch", AC97_MASTER, 14, 6, 1, 0), +SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0), + +SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1), +SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0), +SOC_SINGLE("Out4 Playback Volume", AC97_MASTER_MONO, 8, 63, 1), + +SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1), +SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0), +SOC_SINGLE("Out3 Playback Volume", AC97_MASTER_MONO, 0, 63, 1), + +SOC_SINGLE("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1), +SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1), +SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0), +SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1), + +SOC_SINGLE("PC Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1), +SOC_SINGLE("PC Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1), +SOC_SINGLE("PC Beep Playback Mono Volume", AC97_AUX, 4, 7, 1), + +SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1), +SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1), +SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1), + +SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1), +SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1), +SOC_SINGLE("Aux Playback Mono Volume", AC97_REC_SEL, 4, 7, 1), + +SOC_ENUM("Bass Control", wm9713_enum[16]), +SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1), +SOC_SINGLE("Tone Cut-off Switch", AC97_GENERAL_PURPOSE, 4, 1, 1), +SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_GENERAL_PURPOSE, 6, 1, 0), +SOC_SINGLE("Bass Volume", AC97_GENERAL_PURPOSE, 8, 15, 1), +SOC_SINGLE("Tone Volume", AC97_GENERAL_PURPOSE, 0, 15, 1), + +SOC_SINGLE("3D Upper Cut-off Switch", AC97_REC_GAIN_MIC, 5, 1, 0), +SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0), +SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), +}; + +/* add non dapm controls */ +static int wm9713_add_controls(struct snd_soc_codec *codec) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) { + err = snd_ctl_add(codec->card, + snd_soc_cnew(&wm9713_snd_ac97_controls[i], + codec, NULL)); + if (err < 0) + return err; + } + return 0; +} + +/* We have to create a fake left and right HP mixers because + * the codec only has a single control that is shared by both channels. + * This makes it impossible to determine the audio path using the current + * register map, thus we add a new (virtual) register to help determine the + * audio route within the device. + */ +static int mixer_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + u16 l, r, beep, tone, phone, rec, pcm, aux; + + l = ac97_read(w->codec, HPL_MIXER); + r = ac97_read(w->codec, HPR_MIXER); + beep = ac97_read(w->codec, AC97_PC_BEEP); + tone = ac97_read(w->codec, AC97_MASTER_TONE); + phone = ac97_read(w->codec, AC97_PHONE); + rec = ac97_read(w->codec, AC97_REC_SEL); + pcm = ac97_read(w->codec, AC97_PCM); + aux = ac97_read(w->codec, AC97_AUX); + + if (event & SND_SOC_DAPM_PRE_REG) + return 0; + if ((l & 0x1) || (r & 0x1)) + ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff); + else + ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000); + + if ((l & 0x2) || (r & 0x2)) + ac97_write(w->codec, AC97_MASTER_TONE, tone & 0x7fff); + else + ac97_write(w->codec, AC97_MASTER_TONE, tone | 0x8000); + + if ((l & 0x4) || (r & 0x4)) + ac97_write(w->codec, AC97_PHONE, phone & 0x7fff); + else + ac97_write(w->codec, AC97_PHONE, phone | 0x8000); + + if ((l & 0x8) || (r & 0x8)) + ac97_write(w->codec, AC97_REC_SEL, rec & 0x7fff); + else + ac97_write(w->codec, AC97_REC_SEL, rec | 0x8000); + + if ((l & 0x10) || (r & 0x10)) + ac97_write(w->codec, AC97_PCM, pcm & 0x7fff); + else + ac97_write(w->codec, AC97_PCM, pcm | 0x8000); + + if ((l & 0x20) || (r & 0x20)) + ac97_write(w->codec, AC97_AUX, aux & 0x7fff); + else + ac97_write(w->codec, AC97_AUX, aux | 0x8000); + + return 0; +} + +/* Left Headphone Mixers */ +static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", HPL_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Voice Playback Switch", HPL_MIXER, 4, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 3, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 2, 1, 0), +SOC_DAPM_SINGLE("MonoIn Playback Switch", HPL_MIXER, 1, 1, 0), +SOC_DAPM_SINGLE("Bypass Playback Switch", HPL_MIXER, 0, 1, 0), +}; + +/* Right Headphone Mixers */ +static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", HPR_MIXER, 5, 1, 0), +SOC_DAPM_SINGLE("Voice Playback Switch", HPR_MIXER, 4, 1, 0), +SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 3, 1, 0), +SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 2, 1, 0), +SOC_DAPM_SINGLE("MonoIn Playback Switch", HPR_MIXER, 1, 1, 0), +SOC_DAPM_SINGLE("Bypass Playback Switch", HPR_MIXER, 0, 1, 0), +}; + +/* headphone capture mux */ +static const struct snd_kcontrol_new wm9713_hp_rec_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[1]); + +/* headphone mic mux */ +static const struct snd_kcontrol_new wm9713_hp_mic_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[0]); + +/* Speaker Mixer */ +static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 11, 1, 1), +SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1), +SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1), +SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1), +SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 14, 1, 1), +SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1), +}; + +/* Mono Mixer */ +static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = { +SOC_DAPM_SINGLE("PC Beep Playback Switch", AC97_AUX, 7, 1, 1), +SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1), +SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1), +SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1), +SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 13, 1, 1), +SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 13, 1, 1), +SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_LINE, 7, 1, 1), +SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_LINE, 6, 1, 1), +}; + +/* mono mic mux */ +static const struct snd_kcontrol_new wm9713_mono_mic_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[2]); + +/* mono output mux */ +static const struct snd_kcontrol_new wm9713_mono_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[7]); + +/* speaker left output mux */ +static const struct snd_kcontrol_new wm9713_hp_spkl_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[8]); + +/* speaker right output mux */ +static const struct snd_kcontrol_new wm9713_hp_spkr_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[9]); + +/* headphone left output mux */ +static const struct snd_kcontrol_new wm9713_hpl_out_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[10]); + +/* headphone right output mux */ +static const struct snd_kcontrol_new wm9713_hpr_out_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[11]); + +/* Out3 mux */ +static const struct snd_kcontrol_new wm9713_out3_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[12]); + +/* Out4 mux */ +static const struct snd_kcontrol_new wm9713_out4_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[13]); + +/* DAC inv mux 1 */ +static const struct snd_kcontrol_new wm9713_dac_inv1_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[14]); + +/* DAC inv mux 2 */ +static const struct snd_kcontrol_new wm9713_dac_inv2_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[15]); + +/* Capture source left */ +static const struct snd_kcontrol_new wm9713_rec_srcl_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[3]); + +/* Capture source right */ +static const struct snd_kcontrol_new wm9713_rec_srcr_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[4]); + +/* mic source */ +static const struct snd_kcontrol_new wm9713_mic_sel_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[18]); + +/* mic source B virtual control */ +static const struct snd_kcontrol_new wm9713_micb_sel_mux_controls = +SOC_DAPM_ENUM("Route", wm9713_enum[19]); + +static const struct snd_soc_dapm_widget wm9713_dapm_widgets[] = { +SND_SOC_DAPM_MUX("Capture Headphone Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_rec_mux_controls), +SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_mic_mux_controls), +SND_SOC_DAPM_MUX("Capture Mono Mux", SND_SOC_NOPM, 0, 0, + &wm9713_mono_mic_mux_controls), +SND_SOC_DAPM_MUX("Mono Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_mono_mux_controls), +SND_SOC_DAPM_MUX("Left Speaker Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_spkl_mux_controls), +SND_SOC_DAPM_MUX("Right Speaker Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hp_spkr_mux_controls), +SND_SOC_DAPM_MUX("Left Headphone Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hpl_out_mux_controls), +SND_SOC_DAPM_MUX("Right Headphone Out Mux", SND_SOC_NOPM, 0, 0, + &wm9713_hpr_out_mux_controls), +SND_SOC_DAPM_MUX("Out 3 Mux", SND_SOC_NOPM, 0, 0, + &wm9713_out3_mux_controls), +SND_SOC_DAPM_MUX("Out 4 Mux", SND_SOC_NOPM, 0, 0, + &wm9713_out4_mux_controls), +SND_SOC_DAPM_MUX("DAC Inv Mux 1", SND_SOC_NOPM, 0, 0, + &wm9713_dac_inv1_mux_controls), +SND_SOC_DAPM_MUX("DAC Inv Mux 2", SND_SOC_NOPM, 0, 0, + &wm9713_dac_inv2_mux_controls), +SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0, + &wm9713_rec_srcl_mux_controls), +SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0, + &wm9713_rec_srcr_mux_controls), +SND_SOC_DAPM_MUX("Mic A Source", SND_SOC_NOPM, 0, 0, + &wm9713_mic_sel_mux_controls), +SND_SOC_DAPM_MUX("Mic B Source", SND_SOC_NOPM, 0, 0, + &wm9713_micb_sel_mux_controls), +SND_SOC_DAPM_MIXER_E("Left HP Mixer", AC97_EXTENDED_MID, 3, 1, + &wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls), + mixer_event, SND_SOC_DAPM_POST_REG), +SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_EXTENDED_MID, 2, 1, + &wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls), + mixer_event, SND_SOC_DAPM_POST_REG), +SND_SOC_DAPM_MIXER("Mono Mixer", AC97_EXTENDED_MID, 0, 1, + &wm9713_mono_mixer_controls[0], ARRAY_SIZE(wm9713_mono_mixer_controls)), +SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_EXTENDED_MID, 1, 1, + &wm9713_speaker_mixer_controls[0], + ARRAY_SIZE(wm9713_speaker_mixer_controls)), +SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_EXTENDED_MID, 7, 1), +SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_EXTENDED_MID, 6, 1), +SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1), +SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1), +SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_EXTENDED_MID, 5, 1), +SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_EXTENDED_MID, 4, 1), +SND_SOC_DAPM_PGA("Left Headphone", AC97_EXTENDED_MSTATUS, 10, 1, NULL, 0), +SND_SOC_DAPM_PGA("Right Headphone", AC97_EXTENDED_MSTATUS, 9, 1, NULL, 0), +SND_SOC_DAPM_PGA("Left Speaker", AC97_EXTENDED_MSTATUS, 8, 1, NULL, 0), +SND_SOC_DAPM_PGA("Right Speaker", AC97_EXTENDED_MSTATUS, 7, 1, NULL, 0), +SND_SOC_DAPM_PGA("Out 3", AC97_EXTENDED_MSTATUS, 11, 1, NULL, 0), +SND_SOC_DAPM_PGA("Out 4", AC97_EXTENDED_MSTATUS, 12, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mono Out", AC97_EXTENDED_MSTATUS, 13, 1, NULL, 0), +SND_SOC_DAPM_PGA("Left Line In", AC97_EXTENDED_MSTATUS, 6, 1, NULL, 0), +SND_SOC_DAPM_PGA("Right Line In", AC97_EXTENDED_MSTATUS, 5, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mono In", AC97_EXTENDED_MSTATUS, 4, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic A PGA", AC97_EXTENDED_MSTATUS, 3, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic B PGA", AC97_EXTENDED_MSTATUS, 2, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic A Pre Amp", AC97_EXTENDED_MSTATUS, 1, 1, NULL, 0), +SND_SOC_DAPM_PGA("Mic B Pre Amp", AC97_EXTENDED_MSTATUS, 0, 1, NULL, 0), +SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_EXTENDED_MSTATUS, 14, 1), +SND_SOC_DAPM_OUTPUT("MONO"), +SND_SOC_DAPM_OUTPUT("HPL"), +SND_SOC_DAPM_OUTPUT("HPR"), +SND_SOC_DAPM_OUTPUT("SPKL"), +SND_SOC_DAPM_OUTPUT("SPKR"), +SND_SOC_DAPM_OUTPUT("OUT3"), +SND_SOC_DAPM_OUTPUT("OUT4"), +SND_SOC_DAPM_INPUT("LINEL"), +SND_SOC_DAPM_INPUT("LINER"), +SND_SOC_DAPM_INPUT("MONOIN"), +SND_SOC_DAPM_INPUT("PCBEEP"), +SND_SOC_DAPM_INPUT("MIC1"), +SND_SOC_DAPM_INPUT("MIC2A"), +SND_SOC_DAPM_INPUT("MIC2B"), +SND_SOC_DAPM_VMID("VMID"), +}; + +static const char *audio_map[][3] = { + /* left HP mixer */ + {"Left HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Left HP Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Left HP Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Left HP Mixer", "Bypass Playback Switch", "Left Line In"}, + {"Left HP Mixer", "PCM Playback Switch", "Left DAC"}, + {"Left HP Mixer", "MonoIn Playback Switch", "Mono In"}, + {"Left HP Mixer", NULL, "Capture Headphone Mux"}, + + /* right HP mixer */ + {"Right HP Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Right HP Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Right HP Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Right HP Mixer", "Bypass Playback Switch", "Right Line In"}, + {"Right HP Mixer", "PCM Playback Switch", "Right DAC"}, + {"Right HP Mixer", "MonoIn Playback Switch", "Mono In"}, + {"Right HP Mixer", NULL, "Capture Headphone Mux"}, + + /* virtual mixer - mixes left & right channels for spk and mono */ + {"AC97 Mixer", NULL, "Left DAC"}, + {"AC97 Mixer", NULL, "Right DAC"}, + {"Line Mixer", NULL, "Right Line In"}, + {"Line Mixer", NULL, "Left Line In"}, + {"HP Mixer", NULL, "Left HP Mixer"}, + {"HP Mixer", NULL, "Right HP Mixer"}, + {"Capture Mixer", NULL, "Left Capture Source"}, + {"Capture Mixer", NULL, "Right Capture Source"}, + + /* speaker mixer */ + {"Speaker Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Speaker Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Speaker Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"}, + {"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"}, + {"Speaker Mixer", "MonoIn Playback Switch", "Mono In"}, + + /* mono mixer */ + {"Mono Mixer", "PC Beep Playback Switch", "PCBEEP"}, + {"Mono Mixer", "Voice Playback Switch", "Voice DAC"}, + {"Mono Mixer", "Aux Playback Switch", "Aux DAC"}, + {"Mono Mixer", "Bypass Playback Switch", "Line Mixer"}, + {"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"}, + {"Mono Mixer", "Mic 1 Sidetone Switch", "Mic A PGA"}, + {"Mono Mixer", "Mic 2 Sidetone Switch", "Mic B PGA"}, + {"Mono Mixer", NULL, "Capture Mono Mux"}, + + /* DAC inv mux 1 */ + {"DAC Inv Mux 1", "Mono", "Mono Mixer"}, + {"DAC Inv Mux 1", "Speaker", "Speaker Mixer"}, + {"DAC Inv Mux 1", "Left Headphone", "Left HP Mixer"}, + {"DAC Inv Mux 1", "Right Headphone", "Right HP Mixer"}, + {"DAC Inv Mux 1", "Headphone Mono", "HP Mixer"}, + + /* DAC inv mux 2 */ + {"DAC Inv Mux 2", "Mono", "Mono Mixer"}, + {"DAC Inv Mux 2", "Speaker", "Speaker Mixer"}, + {"DAC Inv Mux 2", "Left Headphone", "Left HP Mixer"}, + {"DAC Inv Mux 2", "Right Headphone", "Right HP Mixer"}, + {"DAC Inv Mux 2", "Headphone Mono", "HP Mixer"}, + + /* headphone left mux */ + {"Left Headphone Out Mux", "Headphone", "Left HP Mixer"}, + + /* headphone right mux */ + {"Right Headphone Out Mux", "Headphone", "Right HP Mixer"}, + + /* speaker left mux */ + {"Left Speaker Out Mux", "Headphone", "Left HP Mixer"}, + {"Left Speaker Out Mux", "Speaker", "Speaker Mixer"}, + {"Left Speaker Out Mux", "Inv", "DAC Inv Mux 1"}, + + /* speaker right mux */ + {"Right Speaker Out Mux", "Headphone", "Right HP Mixer"}, + {"Right Speaker Out Mux", "Speaker", "Speaker Mixer"}, + {"Right Speaker Out Mux", "Inv", "DAC Inv Mux 2"}, + + /* mono mux */ + {"Mono Out Mux", "Mono", "Mono Mixer"}, + {"Mono Out Mux", "Inv", "DAC Inv Mux 1"}, + + /* out 3 mux */ + {"Out 3 Mux", "Inv 1", "DAC Inv Mux 1"}, + + /* out 4 mux */ + {"Out 4 Mux", "Inv 2", "DAC Inv Mux 2"}, + + /* output pga */ + {"HPL", NULL, "Left Headphone"}, + {"Left Headphone", NULL, "Left Headphone Out Mux"}, + {"HPR", NULL, "Right Headphone"}, + {"Right Headphone", NULL, "Right Headphone Out Mux"}, + {"OUT3", NULL, "Out 3"}, + {"Out 3", NULL, "Out 3 Mux"}, + {"OUT4", NULL, "Out 4"}, + {"Out 4", NULL, "Out 4 Mux"}, + {"SPKL", NULL, "Left Speaker"}, + {"Left Speaker", NULL, "Left Speaker Out Mux"}, + {"SPKR", NULL, "Right Speaker"}, + {"Right Speaker", NULL, "Right Speaker Out Mux"}, + {"MONO", NULL, "Mono Out"}, + {"Mono Out", NULL, "Mono Out Mux"}, + + /* input pga */ + {"Left Line In", NULL, "LINEL"}, + {"Right Line In", NULL, "LINER"}, + {"Mono In", NULL, "MONOIN"}, + {"Mic A PGA", NULL, "Mic A Pre Amp"}, + {"Mic B PGA", NULL, "Mic B Pre Amp"}, + + /* left capture select */ + {"Left Capture Source", "Mic 1", "Mic A Pre Amp"}, + {"Left Capture Source", "Mic 2", "Mic B Pre Amp"}, + {"Left Capture Source", "Line", "LINEL"}, + {"Left Capture Source", "Mono In", "MONOIN"}, + {"Left Capture Source", "Headphone", "Left HP Mixer"}, + {"Left Capture Source", "Speaker", "Speaker Mixer"}, + {"Left Capture Source", "Mono Out", "Mono Mixer"}, + + /* right capture select */ + {"Right Capture Source", "Mic 1", "Mic A Pre Amp"}, + {"Right Capture Source", "Mic 2", "Mic B Pre Amp"}, + {"Right Capture Source", "Line", "LINER"}, + {"Right Capture Source", "Mono In", "MONOIN"}, + {"Right Capture Source", "Headphone", "Right HP Mixer"}, + {"Right Capture Source", "Speaker", "Speaker Mixer"}, + {"Right Capture Source", "Mono Out", "Mono Mixer"}, + + /* left ADC */ + {"Left ADC", NULL, "Left Capture Source"}, + + /* right ADC */ + {"Right ADC", NULL, "Right Capture Source"}, + + /* mic */ + {"Mic A Pre Amp", NULL, "Mic A Source"}, + {"Mic A Source", "Mic 1", "MIC1"}, + {"Mic A Source", "Mic 2 A", "MIC2A"}, + {"Mic A Source", "Mic 2 B", "Mic B Source"}, + {"Mic B Pre Amp", "MPB", "Mic B Source"}, + {"Mic B Source", NULL, "MIC2B"}, + + /* headphone capture */ + {"Capture Headphone Mux", "Stereo", "Capture Mixer"}, + {"Capture Headphone Mux", "Left", "Left Capture Source"}, + {"Capture Headphone Mux", "Right", "Right Capture Source"}, + + /* mono capture */ + {"Capture Mono Mux", "Stereo", "Capture Mixer"}, + {"Capture Mono Mux", "Left", "Left Capture Source"}, + {"Capture Mono Mux", "Right", "Right Capture Source"}, + + {NULL, NULL, NULL}, +}; + +static int wm9713_add_widgets(struct snd_soc_codec *codec) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(wm9713_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &wm9713_dapm_widgets[i]); + + /* set up audio path audio_mapnects */ + for (i = 0; audio_map[i][0] != NULL; i++) + snd_soc_dapm_connect_input(codec, audio_map[i][0], + audio_map[i][1], audio_map[i][2]); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static unsigned int ac97_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + + if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || + reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 || + reg == AC97_CD) + return soc_ac97_ops.read(codec->ac97, reg); + else { + reg = reg >> 1; + + if (reg > (ARRAY_SIZE(wm9713_reg))) + return -EIO; + + return cache[reg]; + } +} + +static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + u16 *cache = codec->reg_cache; + if (reg < 0x7c) + soc_ac97_ops.write(codec->ac97, reg, val); + reg = reg >> 1; + if (reg <= (ARRAY_SIZE(wm9713_reg))) + cache[reg] = val; + + return 0; +} + +/* PLL divisors */ +struct _pll_div { + u32 divsel:1; + u32 divctl:1; + u32 lf:1; + u32 n:4; + u32 k:24; +}; + +/* The size in bits of the PLL divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 22) * 10) + +static void pll_factors(struct _pll_div *pll_div, unsigned int source) +{ + u64 Kpart; + unsigned int K, Ndiv, Nmod, target; + + /* The the PLL output is always 98.304MHz. */ + target = 98304000; + + /* If the input frequency is over 14.4MHz then scale it down. */ + if (source > 14400000) { + source >>= 1; + pll_div->divsel = 1; + + if (source > 14400000) { + source >>= 1; + pll_div->divctl = 1; + } else + pll_div->divctl = 0; + + } else { + pll_div->divsel = 0; + pll_div->divctl = 0; + } + + /* Low frequency sources require an additional divide in the + * loop. + */ + if (source < 8192000) { + pll_div->lf = 1; + target >>= 2; + } else + pll_div->lf = 0; + + Ndiv = target / source; + if ((Ndiv < 5) || (Ndiv > 12)) + printk(KERN_WARNING + "WM9713 PLL N value %d out of recommended range!\n", + Ndiv); + + pll_div->n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div->k = K; +} + +/** + * Please note that changing the PLL input frequency may require + * resynchronisation with the AC97 controller. + */ +static int wm9713_set_pll(struct snd_soc_codec *codec, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct wm9713_priv *wm9713 = codec->private_data; + u16 reg, reg2; + struct _pll_div pll_div; + + /* turn PLL off ? */ + if (freq_in == 0 || freq_out == 0) { + /* disable PLL power and select ext source */ + reg = ac97_read(codec, AC97_HANDSET_RATE); + ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080); + reg = ac97_read(codec, AC97_EXTENDED_MID); + ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200); + wm9713->pll_out = 0; + return 0; + } + + pll_factors(&pll_div, freq_in); + + if (pll_div.k == 0) { + reg = (pll_div.n << 12) | (pll_div.lf << 11) | + (pll_div.divsel << 9) | (pll_div.divctl << 8); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + } else { + /* write the fractional k to the reg 0x46 pages */ + reg2 = (pll_div.n << 12) | (pll_div.lf << 11) | (1 << 10) | + (pll_div.divsel << 9) | (pll_div.divctl << 8); + + /* K [21:20] */ + reg = reg2 | (0x5 << 4) | (pll_div.k >> 20); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [19:16] */ + reg = reg2 | (0x4 << 4) | ((pll_div.k >> 16) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [15:12] */ + reg = reg2 | (0x3 << 4) | ((pll_div.k >> 12) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [11:8] */ + reg = reg2 | (0x2 << 4) | ((pll_div.k >> 8) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + /* K [7:4] */ + reg = reg2 | (0x1 << 4) | ((pll_div.k >> 4) & 0xf); + ac97_write(codec, AC97_LINE1_LEVEL, reg); + + reg = reg2 | (0x0 << 4) | (pll_div.k & 0xf); /* K [3:0] */ + ac97_write(codec, AC97_LINE1_LEVEL, reg); + } + + /* turn PLL on and select as source */ + reg = ac97_read(codec, AC97_EXTENDED_MID); + ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff); + reg = ac97_read(codec, AC97_HANDSET_RATE); + ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f); + wm9713->pll_out = freq_out; + wm9713->pll_in = freq_in; + + /* wait 10ms AC97 link frames for the link to stabilise */ + schedule_timeout_interruptible(msecs_to_jiffies(10)); + return 0; +} + +static int wm9713_set_dai_pll(struct snd_soc_codec_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + return wm9713_set_pll(codec, pll_id, freq_in, freq_out); +} + +/* + * Tristate the PCM DAI lines, tristate can be disabled by calling + * wm9713_set_dai_fmt() + */ +static int wm9713_set_dai_tristate(struct snd_soc_codec_dai *codec_dai, + int tristate) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0x9fff; + + if (tristate) + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + + return 0; +} + +/* + * Configure WM9713 clock dividers. + * Voice DAC needs 256 FS + */ +static int wm9713_set_dai_clkdiv(struct snd_soc_codec_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM9713_PCMCLK_DIV: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xf0ff; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_CLKA_MULT: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffd; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_CLKB_MULT: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0xfffb; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_HIFI_DIV: + reg = ac97_read(codec, AC97_HANDSET_RATE) & 0x8fff; + ac97_write(codec, AC97_HANDSET_RATE, reg | div); + break; + case WM9713_PCMBCLK_DIV: + reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xf1ff; + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg | div); + break; + case WM9713_PCMCLK_PLL_DIV: + reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80; + ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x60 | div); + break; + case WM9713_HIFI_PLL_DIV: + reg = ac97_read(codec, AC97_LINE1_LEVEL) & 0xff80; + ac97_write(codec, AC97_LINE1_LEVEL, reg | 0x70 | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int wm9713_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 gpio = ac97_read(codec, AC97_GPIO_CFG) & 0xffc5; + u16 reg = 0x8000; + + /* clock masters */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + reg |= 0x4000; + gpio |= 0x0010; + break; + case SND_SOC_DAIFMT_CBM_CFS: + reg |= 0x6000; + gpio |= 0x0018; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg |= 0x0200; + gpio |= 0x001a; + break; + case SND_SOC_DAIFMT_CBS_CFM: + gpio |= 0x0012; + break; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + reg |= 0x00c0; + break; + case SND_SOC_DAIFMT_IB_NF: + reg |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + reg |= 0x0040; + break; + } + + /* DAI format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + reg |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + reg |= 0x0043; + break; + } + + ac97_write(codec, AC97_GPIO_CFG, gpio); + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + return 0; +} + +static int wm9713_pcm_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_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 reg = ac97_read(codec, AC97_CENTER_LFE_MASTER) & 0xfff3; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + reg |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + reg |= 0x0008; + break; + case SNDRV_PCM_FORMAT_S32_LE: + reg |= 0x000c; + break; + } + + /* enable PCM interface in master mode */ + ac97_write(codec, AC97_CENTER_LFE_MASTER, reg); + return 0; +} + +static void wm9713_voiceshutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 status; + + /* Gracefully shut down the voice interface. */ + status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000; + ac97_write(codec, AC97_HANDSET_RATE, 0x0280); + schedule_timeout_interruptible(msecs_to_jiffies(1)); + ac97_write(codec, AC97_HANDSET_RATE, 0x0F80); + ac97_write(codec, AC97_EXTENDED_MID, status); +} + +static int ac97_hifi_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + int reg; + u16 vra; + + vra = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = AC97_PCM_FRONT_DAC_RATE; + else + reg = AC97_PCM_LR_ADC_RATE; + + return ac97_write(codec, reg, runtime->rate); +} + +static int ac97_aux_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + u16 vra, xsle; + + vra = ac97_read(codec, AC97_EXTENDED_STATUS); + ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); + xsle = ac97_read(codec, AC97_PCI_SID); + ac97_write(codec, AC97_PCI_SID, xsle | 0x8000); + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -ENODEV; + + return ac97_write(codec, AC97_PCM_SURR_DAC_RATE, runtime->rate); +} + +#define WM9713_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) + +#define WM9713_PCM_FORMATS \ + (SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \ + SNDRV_PCM_FORMAT_S24_LE) + +struct snd_soc_codec_dai wm9713_dai[] = { +{ + .name = "AC97 HiFi", + .type = SND_SOC_DAI_AC97_BUS, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM9713_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM9713_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .prepare = ac97_hifi_prepare,}, + .dai_ops = { + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll,}, + }, + { + .name = "AC97 Aux", + .playback = { + .stream_name = "Aux Playback", + .channels_min = 1, + .channels_max = 1, + .rates = WM9713_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .prepare = ac97_aux_prepare,}, + .dai_ops = { + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll,}, + }, + { + .name = "WM9713 Voice", + .playback = { + .stream_name = "Voice Playback", + .channels_min = 1, + .channels_max = 1, + .rates = WM9713_RATES, + .formats = WM9713_PCM_FORMATS,}, + .capture = { + .stream_name = "Voice Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM9713_RATES, + .formats = WM9713_PCM_FORMATS,}, + .ops = { + .hw_params = wm9713_pcm_hw_params, + .shutdown = wm9713_voiceshutdown,}, + .dai_ops = { + .set_clkdiv = wm9713_set_dai_clkdiv, + .set_pll = wm9713_set_dai_pll, + .set_fmt = wm9713_set_dai_fmt, + .set_tristate = wm9713_set_dai_tristate, + }, + }, +}; +EXPORT_SYMBOL_GPL(wm9713_dai); + +int wm9713_reset(struct snd_soc_codec *codec, int try_warm) +{ + if (try_warm && soc_ac97_ops.warm_reset) { + soc_ac97_ops.warm_reset(codec->ac97); + if (!(ac97_read(codec, 0) & 0x8000)) + return 1; + } + + soc_ac97_ops.reset(codec->ac97); + if (ac97_read(codec, 0) & 0x8000) + return -EIO; + return 0; +} +EXPORT_SYMBOL_GPL(wm9713_reset); + +static int wm9713_dapm_event(struct snd_soc_codec *codec, int event) +{ + u16 reg; + + switch (event) { + case SNDRV_CTL_POWER_D0: /* full On */ + /* enable thermal shutdown */ + reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x1bff; + ac97_write(codec, AC97_EXTENDED_MID, reg); + break; + case SNDRV_CTL_POWER_D1: /* partial On */ + case SNDRV_CTL_POWER_D2: /* partial On */ + break; + case SNDRV_CTL_POWER_D3hot: /* Off, with power */ + /* enable master bias and vmid */ + reg = ac97_read(codec, AC97_EXTENDED_MID) & 0x3bff; + ac97_write(codec, AC97_EXTENDED_MID, reg); + ac97_write(codec, AC97_POWERDOWN, 0x0000); + break; + case SNDRV_CTL_POWER_D3cold: /* Off, without power */ + /* disable everything including AC link */ + ac97_write(codec, AC97_EXTENDED_MID, 0xffff); + ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); + ac97_write(codec, AC97_POWERDOWN, 0xffff); + break; + } + codec->dapm_state = event; + return 0; +} + +static int wm9713_soc_suspend(struct platform_device *pdev, + pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + u16 reg; + + /* Disable everything except touchpanel - that will be handled + * by the touch driver and left disabled if touch is not in + * use. */ + reg = ac97_read(codec, AC97_EXTENDED_MID); + ac97_write(codec, AC97_EXTENDED_MID, reg | 0x7fff); + ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff); + ac97_write(codec, AC97_POWERDOWN, 0x6f00); + ac97_write(codec, AC97_POWERDOWN, 0xffff); + + return 0; +} + +static int wm9713_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + struct wm9713_priv *wm9713 = codec->private_data; + int i, ret; + u16 *cache = codec->reg_cache; + + ret = wm9713_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "could not reset AC97 codec\n"); + return ret; + } + + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + + /* do we need to re-start the PLL ? */ + if (wm9713->pll_out) + wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out); + + /* only synchronise the codec if warm reset failed */ + if (ret == 0) { + for (i = 2; i < ARRAY_SIZE(wm9713_reg) << 1; i += 2) { + if (i == AC97_POWERDOWN || i == AC97_EXTENDED_MID || + i == AC97_EXTENDED_MSTATUS || i > 0x66) + continue; + soc_ac97_ops.write(codec->ac97, i, cache[i>>1]); + } + } + + if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D0); + + return ret; +} + +static int wm9713_soc_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0, reg; + + printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION); + + socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (socdev->codec == NULL) + return -ENOMEM; + codec = socdev->codec; + mutex_init(&codec->mutex); + + codec->reg_cache = kmemdup(wm9713_reg, sizeof(wm9713_reg), GFP_KERNEL); + if (codec->reg_cache == NULL) { + ret = -ENOMEM; + goto cache_err; + } + codec->reg_cache_size = sizeof(wm9713_reg); + codec->reg_cache_step = 2; + + codec->private_data = kzalloc(sizeof(struct wm9713_priv), GFP_KERNEL); + if (codec->private_data == NULL) { + ret = -ENOMEM; + goto priv_err; + } + + codec->name = "WM9713"; + codec->owner = THIS_MODULE; + codec->dai = wm9713_dai; + codec->num_dai = ARRAY_SIZE(wm9713_dai); + codec->write = ac97_write; + codec->read = ac97_read; + codec->dapm_event = wm9713_dapm_event; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) + goto codec_err; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) + goto pcm_err; + + /* do a cold reset for the controller and then try + * a warm reset followed by an optional cold reset for codec */ + wm9713_reset(codec, 0); + ret = wm9713_reset(codec, 1); + if (ret < 0) { + printk(KERN_ERR "AC97 link error\n"); + goto reset_err; + } + + wm9713_dapm_event(codec, SNDRV_CTL_POWER_D3hot); + + /* unmute the adc - move to kcontrol */ + reg = ac97_read(codec, AC97_CD) & 0x7fff; + ac97_write(codec, AC97_CD, reg); + + wm9713_add_controls(codec); + wm9713_add_widgets(codec); + ret = snd_soc_register_card(socdev); + if (ret < 0) + goto reset_err; + return 0; + +reset_err: + snd_soc_free_pcms(socdev); + +pcm_err: + snd_soc_free_ac97_codec(codec); + +codec_err: + kfree(codec->private_data); + +priv_err: + kfree(codec->reg_cache); + +cache_err: + kfree(socdev->codec); + socdev->codec = NULL; + return ret; +} + +static int wm9713_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->codec; + + if (codec == NULL) + return 0; + + snd_soc_dapm_free(socdev); + snd_soc_free_pcms(socdev); + snd_soc_free_ac97_codec(codec); + kfree(codec->private_data); + kfree(codec->reg_cache); + kfree(codec->dai); + kfree(codec); + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm9713 = { + .probe = wm9713_soc_probe, + .remove = wm9713_soc_remove, + .suspend = wm9713_soc_suspend, + .resume = wm9713_soc_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm9713); + +MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm9713.h b/sound/soc/codecs/wm9713.h new file mode 100644 index 00000000000..d357b6c8134 --- /dev/null +++ b/sound/soc/codecs/wm9713.h @@ -0,0 +1,53 @@ +/* + * wm9713.h -- WM9713 Soc Audio driver + */ + +#ifndef _WM9713_H +#define _WM9713_H + +/* clock inputs */ +#define WM9713_CLKA_PIN 0 +#define WM9713_CLKB_PIN 1 + +/* clock divider ID's */ +#define WM9713_PCMCLK_DIV 0 +#define WM9713_CLKA_MULT 1 +#define WM9713_CLKB_MULT 2 +#define WM9713_HIFI_DIV 3 +#define WM9713_PCMBCLK_DIV 4 +#define WM9713_PCMCLK_PLL_DIV 5 +#define WM9713_HIFI_PLL_DIV 6 + +/* Calculate the appropriate bit mask for the external PCM clock divider */ +#define WM9713_PCMDIV(x) ((x - 1) << 8) + +/* Calculate the appropriate bit mask for the external HiFi clock divider */ +#define WM9713_HIFIDIV(x) ((x - 1) << 12) + +/* MCLK clock mulitipliers */ +#define WM9713_CLKA_X1 (0 << 1) +#define WM9713_CLKA_X2 (1 << 1) +#define WM9713_CLKB_X1 (0 << 2) +#define WM9713_CLKB_X2 (1 << 2) + +/* MCLK clock MUX */ +#define WM9713_CLK_MUX_A (0 << 0) +#define WM9713_CLK_MUX_B (1 << 0) + +/* Voice DAI BCLK divider */ +#define WM9713_PCMBCLK_DIV_1 (0 << 9) +#define WM9713_PCMBCLK_DIV_2 (1 << 9) +#define WM9713_PCMBCLK_DIV_4 (2 << 9) +#define WM9713_PCMBCLK_DIV_8 (3 << 9) +#define WM9713_PCMBCLK_DIV_16 (4 << 9) + +#define WM9713_DAI_AC97_HIFI 0 +#define WM9713_DAI_AC97_AUX 1 +#define WM9713_DAI_PCM_VOICE 2 + +extern struct snd_soc_codec_device soc_codec_dev_wm9713; +extern struct snd_soc_codec_dai wm9713_dai[3]; + +int wm9713_reset(struct snd_soc_codec *codec, int try_warm); + +#endif diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig new file mode 100644 index 00000000000..20680c551aa --- /dev/null +++ b/sound/soc/davinci/Kconfig @@ -0,0 +1,19 @@ +config SND_DAVINCI_SOC + tristate "SoC Audio for the TI DAVINCI chip" + depends on ARCH_DAVINCI && SND_SOC + help + Say Y or M if you want to add support for codecs attached to + the DAVINCI AC97 or I2S interface. You will also need + to select the audio interfaces to support below. + +config SND_DAVINCI_SOC_I2S + tristate + +config SND_DAVINCI_SOC_EVM + tristate "SoC Audio support for DaVinci EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_EVM + select SND_DAVINCI_SOC_I2S + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + DaVinci EVM platform. diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile new file mode 100644 index 00000000000..ca772e5b463 --- /dev/null +++ b/sound/soc/davinci/Makefile @@ -0,0 +1,11 @@ +# DAVINCI Platform Support +snd-soc-davinci-objs := davinci-pcm.o +snd-soc-davinci-i2s-objs := davinci-i2s.o + +obj-$(CONFIG_SND_DAVINCI_SOC) += snd-soc-davinci.o +obj-$(CONFIG_SND_DAVINCI_SOC_I2S) += snd-soc-davinci-i2s.o + +# DAVINCI Machine Support +snd-soc-evm-objs := davinci-evm.o + +obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c new file mode 100644 index 00000000000..fcd16524033 --- /dev/null +++ b/sound/soc/davinci/davinci-evm.c @@ -0,0 +1,208 @@ +/* + * ASoC driver for TI DAVINCI EVM platform + * + * Author: Vladimir Barinov, <vbarinov@ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.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/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <asm/dma.h> +#include <asm/arch/hardware.h> + +#include "../codecs/tlv320aic3x.h" +#include "davinci-pcm.h" +#include "davinci-i2s.h" + +#define EVM_CODEC_CLOCK 22579200 + +static int evm_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_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + /* set codec DAI configuration */ + ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = codec_dai->dai_ops.set_sysclk(codec_dai, 0, EVM_CODEC_CLOCK, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops evm_ops = { + .hw_params = evm_hw_params, +}; + +/* davinci-evm machine dapm widgets */ +static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +/* davinci-evm machine audio_mapnections to the codec pins */ +static const char *audio_map[][3] = { + /* Headphone connected to HPLOUT, HPROUT */ + {"Headphone Jack", NULL, "HPLOUT"}, + {"Headphone Jack", NULL, "HPROUT"}, + + /* Line Out connected to LLOUT, RLOUT */ + {"Line Out", NULL, "LLOUT"}, + {"Line Out", NULL, "RLOUT"}, + + /* Mic connected to (MIC3L | MIC3R) */ + {"MIC3L", NULL, "Mic Bias 2V"}, + {"MIC3R", NULL, "Mic Bias 2V"}, + {"Mic Bias 2V", NULL, "Mic Jack"}, + + /* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */ + {"LINE1L", NULL, "Line In"}, + {"LINE2L", NULL, "Line In"}, + {"LINE1R", NULL, "Line In"}, + {"LINE2R", NULL, "Line In"}, + + {NULL, NULL, NULL}, +}; + +/* Logic for a aic3x as connected on a davinci-evm */ +static int evm_aic3x_init(struct snd_soc_codec *codec) +{ + int i; + + /* Add davinci-evm specific widgets */ + for (i = 0; i < ARRAY_SIZE(aic3x_dapm_widgets); i++) + snd_soc_dapm_new_control(codec, &aic3x_dapm_widgets[i]); + + /* Set up davinci-evm specific audio path audio_map */ + for (i = 0; audio_map[i][0] != NULL; i++) + snd_soc_dapm_connect_input(codec, audio_map[i][0], + audio_map[i][1], audio_map[i][2]); + + /* not connected */ + snd_soc_dapm_set_endpoint(codec, "MONO_LOUT", 0); + snd_soc_dapm_set_endpoint(codec, "HPLCOM", 0); + snd_soc_dapm_set_endpoint(codec, "HPRCOM", 0); + + /* always connected */ + snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1); + snd_soc_dapm_set_endpoint(codec, "Line Out", 1); + snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1); + snd_soc_dapm_set_endpoint(codec, "Line In", 1); + + snd_soc_dapm_sync_endpoints(codec); + + return 0; +} + +/* davinci-evm digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_i2s_dai, + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, +}; + +/* davinci-evm audio machine driver */ +static struct snd_soc_machine snd_soc_machine_evm = { + .name = "DaVinci EVM", + .dai_link = &evm_dai, + .num_links = 1, +}; + +/* evm audio private data */ +static struct aic3x_setup_data evm_aic3x_setup = { + .i2c_address = 0x1b, +}; + +/* evm audio subsystem */ +static struct snd_soc_device evm_snd_devdata = { + .machine = &snd_soc_machine_evm, + .platform = &davinci_soc_platform, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &evm_aic3x_setup, +}; + +static struct resource evm_snd_resources[] = { + { + .start = DAVINCI_MCBSP_BASE, + .end = DAVINCI_MCBSP_BASE + SZ_8K - 1, + .flags = IORESOURCE_MEM, + }, +}; + +static struct evm_snd_platform_data evm_snd_data = { + .tx_dma_ch = DM644X_DMACH_MCBSP_TX, + .rx_dma_ch = DM644X_DMACH_MCBSP_RX, +}; + +static struct platform_device *evm_snd_device; + +static int __init evm_init(void) +{ + int ret; + + evm_snd_device = platform_device_alloc("soc-audio", 0); + if (!evm_snd_device) + return -ENOMEM; + + platform_set_drvdata(evm_snd_device, &evm_snd_devdata); + evm_snd_devdata.dev = &evm_snd_device->dev; + evm_snd_device->dev.platform_data = &evm_snd_data; + + ret = platform_device_add_resources(evm_snd_device, evm_snd_resources, + ARRAY_SIZE(evm_snd_resources)); + if (ret) { + platform_device_put(evm_snd_device); + return ret; + } + + ret = platform_device_add(evm_snd_device); + if (ret) + platform_device_put(evm_snd_device); + + return ret; +} + +static void __exit evm_exit(void) +{ + platform_device_unregister(evm_snd_device); +} + +module_init(evm_init); +module_exit(evm_exit); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI EVM ASoC driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c new file mode 100644 index 00000000000..c421774b33e --- /dev/null +++ b/sound/soc/davinci/davinci-i2s.c @@ -0,0 +1,407 @@ +/* + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov@ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.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/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/clk.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "davinci-pcm.h" + +#define DAVINCI_MCBSP_DRR_REG 0x00 +#define DAVINCI_MCBSP_DXR_REG 0x04 +#define DAVINCI_MCBSP_SPCR_REG 0x08 +#define DAVINCI_MCBSP_RCR_REG 0x0c +#define DAVINCI_MCBSP_XCR_REG 0x10 +#define DAVINCI_MCBSP_SRGR_REG 0x14 +#define DAVINCI_MCBSP_PCR_REG 0x24 + +#define DAVINCI_MCBSP_SPCR_RRST (1 << 0) +#define DAVINCI_MCBSP_SPCR_RINTM(v) ((v) << 4) +#define DAVINCI_MCBSP_SPCR_XRST (1 << 16) +#define DAVINCI_MCBSP_SPCR_XINTM(v) ((v) << 20) +#define DAVINCI_MCBSP_SPCR_GRST (1 << 22) +#define DAVINCI_MCBSP_SPCR_FRST (1 << 23) +#define DAVINCI_MCBSP_SPCR_FREE (1 << 25) + +#define DAVINCI_MCBSP_RCR_RWDLEN1(v) ((v) << 5) +#define DAVINCI_MCBSP_RCR_RFRLEN1(v) ((v) << 8) +#define DAVINCI_MCBSP_RCR_RDATDLY(v) ((v) << 16) +#define DAVINCI_MCBSP_RCR_RWDLEN2(v) ((v) << 21) + +#define DAVINCI_MCBSP_XCR_XWDLEN1(v) ((v) << 5) +#define DAVINCI_MCBSP_XCR_XFRLEN1(v) ((v) << 8) +#define DAVINCI_MCBSP_XCR_XDATDLY(v) ((v) << 16) +#define DAVINCI_MCBSP_XCR_XFIG (1 << 18) +#define DAVINCI_MCBSP_XCR_XWDLEN2(v) ((v) << 21) + +#define DAVINCI_MCBSP_SRGR_FWID(v) ((v) << 8) +#define DAVINCI_MCBSP_SRGR_FPER(v) ((v) << 16) +#define DAVINCI_MCBSP_SRGR_FSGM (1 << 28) + +#define DAVINCI_MCBSP_PCR_CLKRP (1 << 0) +#define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) +#define DAVINCI_MCBSP_PCR_FSRP (1 << 2) +#define DAVINCI_MCBSP_PCR_FSXP (1 << 3) +#define DAVINCI_MCBSP_PCR_CLKRM (1 << 8) +#define DAVINCI_MCBSP_PCR_CLKXM (1 << 9) +#define DAVINCI_MCBSP_PCR_FSRM (1 << 10) +#define DAVINCI_MCBSP_PCR_FSXM (1 << 11) + +#define MOD_REG_BIT(val, mask, set) do { \ + if (set) { \ + val |= mask; \ + } else { \ + val &= ~mask; \ + } \ +} while (0) + +enum { + DAVINCI_MCBSP_WORD_8 = 0, + DAVINCI_MCBSP_WORD_12, + DAVINCI_MCBSP_WORD_16, + DAVINCI_MCBSP_WORD_20, + DAVINCI_MCBSP_WORD_24, + DAVINCI_MCBSP_WORD_32, +}; + +static struct davinci_pcm_dma_params davinci_i2s_pcm_out = { + .name = "I2S PCM Stereo out", +}; + +static struct davinci_pcm_dma_params davinci_i2s_pcm_in = { + .name = "I2S PCM Stereo in", +}; + +struct davinci_mcbsp_dev { + void __iomem *base; + struct clk *clk; + struct davinci_pcm_dma_params *dma_params[2]; +}; + +static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev, + int reg, u32 val) +{ + __raw_writel(val, dev->base + reg); +} + +static inline u32 davinci_mcbsp_read_reg(struct davinci_mcbsp_dev *dev, int reg) +{ + return __raw_readl(dev->base + reg); +} + +static void davinci_mcbsp_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + u32 w; + + /* Start the sample generator and enable transmitter/receiver */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1); + else + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + /* Start frame sync */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_FRST, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); +} + +static void davinci_mcbsp_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + u32 w; + + /* Reset transmitter/receiver and sample rate/frame sync generators */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST | + DAVINCI_MCBSP_SPCR_FRST, 0); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0); + else + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 0); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); +} + +static int davinci_i2s_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + + cpu_dai->dma_data = dev->dma_params[substream->stream]; + + return 0; +} + +static int davinci_i2s_set_dai_fmt(struct snd_soc_cpu_dai *cpu_dai, + unsigned int fmt) +{ + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + u32 w; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, + DAVINCI_MCBSP_PCR_FSXM | + DAVINCI_MCBSP_PCR_FSRM | + DAVINCI_MCBSP_PCR_CLKXM | + DAVINCI_MCBSP_PCR_CLKRM); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, + DAVINCI_MCBSP_SRGR_FSGM); + break; + case SND_SOC_DAIFMT_CBM_CFM: + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, 0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP | + DAVINCI_MCBSP_PCR_CLKRP, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + break; + case SND_SOC_DAIFMT_NB_IF: + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_FSXP | + DAVINCI_MCBSP_PCR_FSRP, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + break; + case SND_SOC_DAIFMT_IB_IF: + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_PCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_PCR_CLKXP | + DAVINCI_MCBSP_PCR_CLKRP | + DAVINCI_MCBSP_PCR_FSXP | + DAVINCI_MCBSP_PCR_FSRP, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, w); + break; + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct snd_interval *i = NULL; + int mcbsp_word_length; + u32 w; + + /* general line settings */ + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, + DAVINCI_MCBSP_SPCR_RINTM(3) | + DAVINCI_MCBSP_SPCR_XINTM(3) | + DAVINCI_MCBSP_SPCR_FREE); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, + DAVINCI_MCBSP_RCR_RFRLEN1(1) | + DAVINCI_MCBSP_RCR_RDATDLY(1)); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, + DAVINCI_MCBSP_XCR_XFRLEN1(1) | + DAVINCI_MCBSP_XCR_XDATDLY(1) | + DAVINCI_MCBSP_XCR_XFIG); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w); + + i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS); + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, w); + + /* Determine xfer data type */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + dma_params->data_type = 1; + mcbsp_word_length = DAVINCI_MCBSP_WORD_8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + dma_params->data_type = 2; + mcbsp_word_length = DAVINCI_MCBSP_WORD_16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + dma_params->data_type = 4; + mcbsp_word_length = DAVINCI_MCBSP_WORD_32; + break; + default: + printk(KERN_WARNING "davinci-i2s: unsupported PCM format"); + return -EINVAL; + } + + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_RCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, w); + + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_XCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) | + DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length), 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, w); + + return 0; +} + +static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_mcbsp_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_mcbsp_stop(substream); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static int davinci_i2s_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct davinci_mcbsp_dev *dev; + struct resource *mem, *ioarea; + struct evm_snd_platform_data *pdata; + int ret; + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "no mem resource?\n"); + return -ENODEV; + } + + ioarea = request_mem_region(mem->start, (mem->end - mem->start) + 1, + pdev->name); + if (!ioarea) { + dev_err(&pdev->dev, "McBSP region already claimed\n"); + return -EBUSY; + } + + dev = kzalloc(sizeof(struct davinci_mcbsp_dev), GFP_KERNEL); + if (!dev) { + ret = -ENOMEM; + goto err_release_region; + } + + cpu_dai->private_data = dev; + + dev->clk = clk_get(&pdev->dev, "McBSPCLK"); + if (IS_ERR(dev->clk)) { + ret = -ENODEV; + goto err_free_mem; + } + clk_enable(dev->clk); + + dev->base = (void __iomem *)IO_ADDRESS(mem->start); + pdata = pdev->dev.platform_data; + + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK] = &davinci_i2s_pcm_out; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->channel = pdata->tx_dma_ch; + dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK]->dma_addr = + (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG); + + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE] = &davinci_i2s_pcm_in; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->channel = pdata->rx_dma_ch; + dev->dma_params[SNDRV_PCM_STREAM_CAPTURE]->dma_addr = + (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG); + + return 0; + +err_free_mem: + kfree(dev); +err_release_region: + release_mem_region(mem->start, (mem->end - mem->start) + 1); + + return ret; +} + +static void davinci_i2s_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_machine *machine = socdev->machine; + struct snd_soc_cpu_dai *cpu_dai = machine->dai_link[pdev->id].cpu_dai; + struct davinci_mcbsp_dev *dev = cpu_dai->private_data; + struct resource *mem; + + clk_disable(dev->clk); + clk_put(dev->clk); + dev->clk = NULL; + + kfree(dev); + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(mem->start, (mem->end - mem->start) + 1); +} + +#define DAVINCI_I2S_RATES SNDRV_PCM_RATE_8000_96000 + +struct snd_soc_cpu_dai davinci_i2s_dai = { + .name = "davinci-i2s", + .id = 0, + .type = SND_SOC_DAI_I2S, + .probe = davinci_i2s_probe, + .remove = davinci_i2s_remove, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = DAVINCI_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = { + .startup = davinci_i2s_startup, + .trigger = davinci_i2s_trigger, + .hw_params = davinci_i2s_hw_params,}, + .dai_ops = { + .set_fmt = davinci_i2s_set_dai_fmt, + }, +}; +EXPORT_SYMBOL_GPL(davinci_i2s_dai); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-i2s.h b/sound/soc/davinci/davinci-i2s.h new file mode 100644 index 00000000000..9592d17db32 --- /dev/null +++ b/sound/soc/davinci/davinci-i2s.h @@ -0,0 +1,17 @@ +/* + * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov@ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.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 _DAVINCI_I2S_H +#define _DAVINCI_I2S_H + +extern struct snd_soc_cpu_dai davinci_i2s_dai; + +#endif diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c new file mode 100644 index 00000000000..6a76927c997 --- /dev/null +++ b/sound/soc/davinci/davinci-pcm.c @@ -0,0 +1,389 @@ +/* + * ALSA PCM interface for the TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov@ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.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/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> + +#include "davinci-pcm.h" + +#define DAVINCI_PCM_DEBUG 0 +#if DAVINCI_PCM_DEBUG +#define DPRINTK(x...) printk(KERN_DEBUG x) +#else +#define DPRINTK(x...) +#endif + +static struct snd_pcm_hardware davinci_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_LE), + .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_KNOT), + .rate_min = 8000, + .rate_max = 96000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 128 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 8 * 1024, + .periods_min = 16, + .periods_max = 255, + .fifo_size = 0, +}; + +struct davinci_runtime_data { + spinlock_t lock; + int period; /* current DMA period */ + int master_lch; /* Master DMA channel */ + int slave_lch; /* Slave DMA channel */ + struct davinci_pcm_dma_params *params; /* DMA params */ +}; + +static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + int lch = prtd->slave_lch; + unsigned int period_size; + unsigned int dma_offset; + dma_addr_t dma_pos; + dma_addr_t src, dst; + unsigned short src_bidx, dst_bidx; + unsigned int data_type; + unsigned int count; + + period_size = snd_pcm_lib_period_bytes(substream); + dma_offset = prtd->period * period_size; + dma_pos = runtime->dma_addr + dma_offset; + + DPRINTK("audio_set_dma_params_play channel = %d dma_ptr = %x " + "period_size=%x\n", lch, dma_pos, period_size); + + data_type = prtd->params->data_type; + count = period_size / data_type; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = dma_pos; + dst = prtd->params->dma_addr; + src_bidx = data_type; + dst_bidx = 0; + } else { + src = prtd->params->dma_addr; + dst = dma_pos; + src_bidx = 0; + dst_bidx = data_type; + } + + davinci_set_dma_src_params(lch, src, INCR, W8BIT); + davinci_set_dma_dest_params(lch, dst, INCR, W8BIT); + davinci_set_dma_src_index(lch, src_bidx, 0); + davinci_set_dma_dest_index(lch, dst_bidx, 0); + davinci_set_dma_transfer_params(lch, data_type, count, 1, 0, ASYNC); + + prtd->period++; + if (unlikely(prtd->period >= runtime->periods)) + prtd->period = 0; +} + +static void davinci_pcm_dma_irq(int lch, u16 ch_status, void *data) +{ + struct snd_pcm_substream *substream = data; + struct davinci_runtime_data *prtd = substream->runtime->private_data; + + DPRINTK("lch=%d, status=0x%x\n", lch, ch_status); + + if (unlikely(ch_status != DMA_COMPLETE)) + return; + + if (snd_pcm_running(substream)) { + snd_pcm_period_elapsed(substream); + + spin_lock(&prtd->lock); + davinci_pcm_enqueue_dma(substream); + spin_unlock(&prtd->lock); + } +} + +static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct davinci_pcm_dma_params *dma_data = rtd->dai->cpu_dai->dma_data; + int tcc = TCC_ANY; + int ret; + + if (!dma_data) + return -ENODEV; + + prtd->params = dma_data; + + /* Request master DMA channel */ + ret = davinci_request_dma(prtd->params->channel, prtd->params->name, + davinci_pcm_dma_irq, substream, + &prtd->master_lch, &tcc, EVENTQ_0); + if (ret) + return ret; + + /* Request slave DMA channel */ + ret = davinci_request_dma(PARAM_ANY, "Link", + NULL, NULL, &prtd->slave_lch, &tcc, EVENTQ_0); + if (ret) { + davinci_free_dma(prtd->master_lch); + return ret; + } + + /* Link slave DMA channel in loopback */ + davinci_dma_link_lch(prtd->slave_lch, prtd->slave_lch); + + return 0; +} + +static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + int ret = 0; + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + davinci_start_dma(prtd->master_lch); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + davinci_stop_dma(prtd->master_lch); + break; + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static int davinci_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct paramentry_descriptor temp; + + prtd->period = 0; + davinci_pcm_enqueue_dma(substream); + + /* Get slave channel dma params for master channel startup */ + davinci_get_dma_params(prtd->slave_lch, &temp); + davinci_set_dma_params(prtd->master_lch, &temp); + + return 0; +} + +static snd_pcm_uframes_t +davinci_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd = runtime->private_data; + unsigned int offset; + dma_addr_t count; + dma_addr_t src, dst; + + spin_lock(&prtd->lock); + + davinci_dma_getposition(prtd->master_lch, &src, &dst); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + count = src - runtime->dma_addr; + else + count = dst - runtime->dma_addr;; + + spin_unlock(&prtd->lock); + + offset = bytes_to_frames(runtime, count); + if (offset >= runtime->buffer_size) + offset = 0; + + return offset; +} + +static int davinci_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd; + int ret = 0; + + snd_soc_set_runtime_hwparams(substream, &davinci_pcm_hardware); + + prtd = kzalloc(sizeof(struct davinci_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + ret = davinci_pcm_dma_request(substream); + if (ret) { + printk(KERN_ERR "davinci_pcm: Failed to get dma channels\n"); + kfree(prtd); + } + + return ret; +} + +static int davinci_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct davinci_runtime_data *prtd = runtime->private_data; + + davinci_dma_unlink_lch(prtd->slave_lch, prtd->slave_lch); + + davinci_free_dma(prtd->slave_lch); + davinci_free_dma(prtd->master_lch); + + kfree(prtd); + + return 0; +} + +static int davinci_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int davinci_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int davinci_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +struct snd_pcm_ops davinci_pcm_ops = { + .open = davinci_pcm_open, + .close = davinci_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = davinci_pcm_hw_params, + .hw_free = davinci_pcm_hw_free, + .prepare = davinci_pcm_prepare, + .trigger = davinci_pcm_trigger, + .pointer = davinci_pcm_pointer, + .mmap = davinci_pcm_mmap, +}; + +static int davinci_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = davinci_pcm_hardware.buffer_bytes_max; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + + DPRINTK("preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *) buf->area, (void *) buf->addr, size); + + if (!buf->area) + return -ENOMEM; + + buf->bytes = size; + return 0; +} + +static void davinci_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 davinci_pcm_dmamask = 0xffffffff; + +static int davinci_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) +{ + int ret; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &davinci_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = 0xffffffff; + + if (dai->playback.channels_min) { + ret = davinci_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + return ret; + } + + if (dai->capture.channels_min) { + ret = davinci_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + return ret; + } + + return 0; +} + +struct snd_soc_platform davinci_soc_platform = { + .name = "davinci-audio", + .pcm_ops = &davinci_pcm_ops, + .pcm_new = davinci_pcm_new, + .pcm_free = davinci_pcm_free, +}; +EXPORT_SYMBOL_GPL(davinci_soc_platform); + +MODULE_AUTHOR("Vladimir Barinov"); +MODULE_DESCRIPTION("TI DAVINCI PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h new file mode 100644 index 00000000000..8d6a45e75a6 --- /dev/null +++ b/sound/soc/davinci/davinci-pcm.h @@ -0,0 +1,29 @@ +/* + * ALSA PCM interface for the TI DAVINCI processor + * + * Author: Vladimir Barinov, <vbarinov@ru.mvista.com> + * Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.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 _DAVINCI_PCM_H +#define _DAVINCI_PCM_H + +struct davinci_pcm_dma_params { + char *name; /* stream identifier */ + int channel; /* sync dma channel ID */ + dma_addr_t dma_addr; /* device physical address for DMA */ + unsigned int data_type; /* xfer data type */ +}; + +struct evm_snd_platform_data { + int tx_dma_ch; + int rx_dma_ch; +}; + +extern struct snd_soc_platform davinci_soc_platform; + +#endif diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 652514fc814..78de7168d2b 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -20,7 +20,6 @@ #include <linux/interrupt.h> #include <linux/delay.h> -#include <sound/driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 145ad13d52d..b2a11b0d2e4 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -15,7 +15,6 @@ #include <linux/device.h> #include <linux/delay.h> -#include <sound/driver.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 1a70a6ac98c..7f32a116757 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -297,21 +297,19 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec) /* Add corgi specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) { err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8731_corgi_controls[i],codec, NULL)); + snd_soc_cnew(&wm8731_corgi_controls[i], codec, NULL)); if (err < 0) return err; } /* Add corgi specific widgets */ - for(i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) { + for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); - } /* Set up corgi specific audio path audio_map */ - for(i = 0; audio_map[i][0] != NULL; i++) { + for (i = 0; audio_map[i][0] != NULL; i++) snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); - } snd_soc_dapm_sync_endpoints(codec); return 0; @@ -353,7 +351,8 @@ static int __init corgi_init(void) { int ret; - if (!(machine_is_corgi() || machine_is_shepherd() || machine_is_husky())) + if (!(machine_is_corgi() || machine_is_shepherd() || + machine_is_husky())) return -ENODEV; corgi_snd_device = platform_device_alloc("soc-audio", -1); diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 4fbf8bba962..7e830b21894 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -257,21 +257,19 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec) /* Add poodle specific controls */ for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) { err = snd_ctl_add(codec->card, - snd_soc_cnew(&wm8731_poodle_controls[i],codec, NULL)); + snd_soc_cnew(&wm8731_poodle_controls[i], codec, NULL)); if (err < 0) return err; } /* Add poodle specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) { + for (i = 0; i < ARRAY_SIZE(wm8731_dapm_widgets); i++) snd_soc_dapm_new_control(codec, &wm8731_dapm_widgets[i]); - } /* Set up poodle specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) { + for (i = 0; audio_map[i][0] != NULL; i++) snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); - } snd_soc_dapm_sync_endpoints(codec); return 0; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index e1737999880..97ec2d90547 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -61,7 +61,7 @@ static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, mutex_lock(&car_mutex); /* set up primary or secondary codec/modem space */ -#ifdef CONFIG_PXA27x +#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx) reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; #else if (reg == AC97_GPIO_STATUS) @@ -87,7 +87,7 @@ static unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1); if (!((GSR | gsr_bits) & GSR_SDONE)) { printk(KERN_ERR "%s: read error (ac97_reg=%x GSR=%#lx)\n", - __FUNCTION__, reg, GSR | gsr_bits); + __func__, reg, GSR | gsr_bits); val = -1; goto out; } @@ -111,7 +111,7 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, mutex_lock(&car_mutex); /* set up primary or secondary codec/modem space */ -#ifdef CONFIG_PXA27x +#if defined(CONFIG_PXA27x) || defined(CONFIG_PXA3xx) reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; #else if (reg == AC97_GPIO_STATUS) @@ -127,13 +127,16 @@ static void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1); if (!((GSR | gsr_bits) & GSR_CDONE)) printk(KERN_ERR "%s: write error (ac97_reg=%x GSR=%#lx)\n", - __FUNCTION__, reg, GSR | gsr_bits); + __func__, reg, GSR | gsr_bits); mutex_unlock(&car_mutex); } static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) { +#ifdef CONFIG_PXA3xx + int timeout = 100; +#endif gsr_bits = 0; #ifdef CONFIG_PXA27x @@ -144,6 +147,11 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) GCR |= GCR_WARM_RST; pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT); udelay(500); +#elif defined(CONFIG_PXA3xx) + /* Can't use interrupts */ + GCR |= GCR_WARM_RST; + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); #else GCR |= GCR_WARM_RST | GCR_PRIRDY_IEN | GCR_SECRDY_IEN; wait_event_timeout(gsr_wq, gsr_bits & (GSR_PCR | GSR_SCR), 1); @@ -151,7 +159,7 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", - __FUNCTION__, gsr_bits); + __func__, gsr_bits); GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); GCR |= GCR_SDONE_IE|GCR_CDONE_IE; @@ -159,6 +167,16 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) { +#ifdef CONFIG_PXA3xx + int timeout = 1000; + + /* Hold CLKBPB for 100us */ + GCR = 0; + GCR = GCR_CLKBPB; + udelay(100); + GCR = 0; +#endif + GCR &= GCR_COLD_RST; /* clear everything but nCRST */ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ @@ -170,6 +188,13 @@ static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) clk_disable(ac97conf_clk); GCR = GCR_COLD_RST; udelay(50); +#elif defined(CONFIG_PXA3xx) + /* Can't use interrupts on PXA3xx */ + GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); + + GCR = GCR_WARM_RST | GCR_COLD_RST; + while (!(GSR & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(10); #else GCR = GCR_COLD_RST; GCR |= GCR_CDONE_IE|GCR_SDONE_IE; @@ -178,7 +203,7 @@ static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", - __FUNCTION__, gsr_bits); + __func__, gsr_bits); GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); GCR |= GCR_SDONE_IE|GCR_CDONE_IE; diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index daeaa4c8b87..01ad7bf716b 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -64,8 +64,8 @@ static void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) if (dcsr & DCSR_ENDINTR) { snd_pcm_period_elapsed(substream); } else { - printk( KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n", - prtd->params->name, dma_ch, dcsr ); + printk(KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n", + prtd->params->name, dma_ch, dcsr); } } @@ -84,8 +84,8 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ - if (!dma) - return 0; + if (!dma) + return 0; /* this may get called several times by oss emulation * with different params */ @@ -363,7 +363,6 @@ struct snd_soc_platform pxa2xx_soc_platform = { .pcm_new = pxa2xx_pcm_new, .pcm_free = pxa2xx_pcm_free_dma_buffers, }; - EXPORT_SYMBOL_GPL(pxa2xx_soc_platform); MODULE_AUTHOR("Nicolas Pitre"); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index ecca39033fc..d8b8372db00 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -313,15 +313,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) } /* Add spitz specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) { + for (i = 0; i < ARRAY_SIZE(wm8750_dapm_widgets); i++) snd_soc_dapm_new_control(codec, &wm8750_dapm_widgets[i]); - } /* Set up spitz specific audio path audio_map */ - for (i = 0; audio_map[i][0] != NULL; i++) { + for (i = 0; audio_map[i][0] != NULL; i++) snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); - } snd_soc_dapm_sync_endpoints(codec); return 0; diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 6ee115ceb01..962cc20b1af 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -659,6 +659,7 @@ static int __init neo1973_init(void) static void __exit neo1973_exit(void) { + i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); } diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 0a3c630951b..4ebcd6a8bf2 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -25,6 +25,7 @@ #include <linux/delay.h> #include <linux/clk.h> #include <linux/jiffies.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -32,7 +33,6 @@ #include <sound/soc.h> #include <asm/hardware.h> -#include <asm/io.h> #include <asm/arch/regs-gpio.h> #include <asm/arch/regs-clock.h> #include <asm/arch/audio.h> @@ -46,7 +46,7 @@ #define S3C24XX_I2S_DEBUG 0 #if S3C24XX_I2S_DEBUG -#define DBG(x...) printk(KERN_DEBUG x) +#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x) #else #define DBG(x...) #endif @@ -89,7 +89,7 @@ static void s3c24xx_snd_txctrl(int on) u32 iiscon; u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); @@ -134,7 +134,7 @@ static void s3c24xx_snd_rxctrl(int on) u32 iiscon; u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); @@ -159,10 +159,10 @@ static void s3c24xx_snd_rxctrl(int on) * DMA engine will simply freeze randomly. */ - iisfcon &= ~S3C2410_IISFCON_RXENABLE; - iisfcon &= ~S3C2410_IISFCON_RXDMA; - iiscon |= S3C2410_IISCON_RXIDLE; - iiscon &= ~S3C2410_IISCON_RXDMAEN; + iisfcon &= ~S3C2410_IISFCON_RXENABLE; + iisfcon &= ~S3C2410_IISFCON_RXDMA; + iiscon |= S3C2410_IISCON_RXIDLE; + iiscon &= ~S3C2410_IISCON_RXDMAEN; iismod &= ~S3C2410_IISMOD_RXMODE; writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON); @@ -182,7 +182,7 @@ static int s3c24xx_snd_lrsync(void) u32 iiscon; unsigned long timeout = jiffies + msecs_to_jiffies(5); - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); while (1) { iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); @@ -201,7 +201,7 @@ static int s3c24xx_snd_lrsync(void) */ static inline int s3c24xx_snd_is_clkmaster(void) { - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1; } @@ -214,7 +214,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, { u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); DBG("hw_params r: IISMOD: %lx \n", iismod); @@ -250,7 +250,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; u32 iismod; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out; @@ -278,7 +278,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd) { int ret = 0; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -320,7 +320,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, { u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); iismod &= ~S3C2440_IISMOD_MPLL; @@ -346,7 +346,7 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, { u32 reg; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); switch (div_id) { case S3C24XX_DIV_BCLK: @@ -381,13 +381,13 @@ EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); static int s3c24xx_i2s_probe(struct platform_device *pdev) { - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); if (s3c24xx_i2s.regs == NULL) return -ENXIO; - s3c24xx_i2s.iis_clk=clk_get(&pdev->dev, "iis"); + s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis"); if (s3c24xx_i2s.iis_clk == NULL) { DBG("failed to get iis_clock\n"); iounmap(s3c24xx_i2s.regs); @@ -411,9 +411,11 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev) } #ifdef CONFIG_PM -int s3c24xx_i2s_suspend(struct platform_device *pdev, +static int s3c24xx_i2s_suspend(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai) { + DBG("Entered %s\n", __func__); + s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON); s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); s3c24xx_i2s.iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON); @@ -424,9 +426,10 @@ int s3c24xx_i2s_suspend(struct platform_device *pdev, return 0; } -int s3c24xx_i2s_resume(struct platform_device *pdev, +static int s3c24xx_i2s_resume(struct platform_device *pdev, struct snd_soc_cpu_dai *cpu_dai) { + DBG("Entered %s\n", __func__); clk_enable(s3c24xx_i2s.iis_clk); writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON); diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 29a6c82f873..49580fb481d 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -39,7 +39,7 @@ #define S3C24XX_PCM_DEBUG 0 #if S3C24XX_PCM_DEBUG -#define DBG(x...) printk(KERN_DEBUG x) +#define DBG(x...) printk(KERN_DEBUG "s3c24xx-pcm: " x) #else #define DBG(x...) #endif @@ -88,7 +88,7 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) dma_addr_t pos = prtd->dma_pos; int ret; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); while (prtd->dma_loaded < prtd->dma_limit) { unsigned long len = prtd->dma_period; @@ -98,7 +98,7 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) if ((pos + len) > prtd->dma_end) { len = prtd->dma_end - pos; DBG(KERN_DEBUG "%s: corrected dma len %ld\n", - __FUNCTION__, len); + __func__, len); } ret = s3c2410_dma_enqueue(prtd->params->channel, @@ -123,7 +123,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel, struct snd_pcm_substream *substream = dev_id; struct s3c24xx_runtime_data *prtd; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR) return; @@ -152,7 +152,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream, unsigned long totbytes = params_buffer_bytes(params); int ret=0; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ @@ -200,7 +200,7 @@ static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); /* TODO - do we need to ensure DMA flushed */ snd_pcm_set_runtime_buffer(substream, NULL); @@ -218,7 +218,7 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ @@ -263,7 +263,7 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; int ret = 0; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); spin_lock(&prtd->lock); @@ -301,7 +301,7 @@ static snd_pcm_uframes_t unsigned long res; dma_addr_t src, dst; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); spin_lock(&prtd->lock); s3c2410_dma_getposition(prtd->params->channel, &src, &dst); @@ -334,7 +334,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware); @@ -353,7 +353,7 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); if (prtd) kfree(prtd); @@ -368,7 +368,7 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); return dma_mmap_writecombine(substream->pcm->card->dev, vma, runtime->dma_area, @@ -394,7 +394,7 @@ static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) struct snd_dma_buffer *buf = &substream->dma_buffer; size_t size = s3c24xx_pcm_hardware.buffer_bytes_max; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); buf->dev.type = SNDRV_DMA_TYPE_DEV; buf->dev.dev = pcm->card->dev; @@ -413,7 +413,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) struct snd_dma_buffer *buf; int stream; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); for (stream = 0; stream < 2; stream++) { substream = pcm->streams[stream].substream; @@ -437,7 +437,7 @@ static int s3c24xx_pcm_new(struct snd_card *card, { int ret = 0; - DBG("Entered %s\n", __FUNCTION__); + DBG("Entered %s\n", __func__); if (!card->dev->dma_mask) card->dev->dma_mask = &s3c24xx_pcm_dmamask; diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index f03220d23e7..4c1e013381c 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -1,4 +1,5 @@ menu "SoC Audio support for SuperH" + depends on SUPERH config SND_SOC_PCM_SH7760 tristate "SoC Audio support for Renesas SH7760" diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9eb5479787c..e148db940cf 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -839,6 +839,7 @@ static int soc_remove(struct platform_device *pdev) static struct platform_driver soc_driver = { .driver = { .name = "soc-audio", + .owner = THIS_MODULE, }, .probe = soc_probe, .remove = soc_remove, @@ -1601,3 +1602,4 @@ module_exit(snd_soc_exit); MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); MODULE_DESCRIPTION("ALSA SoC Core"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:soc-audio"); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 620d7ea3c15..af3326c6350 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -226,7 +226,7 @@ static int dapm_update_bits(struct snd_soc_dapm_widget *widget) snd_soc_write(codec, widget->reg, new); pop_wait(POP_TIME); } - dbg("reg old %x new %x change %d\n", old, new, change); + dbg("reg %x old %x new %x change %d\n", widget->reg, old, new, change); return change; } @@ -1288,7 +1288,7 @@ int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, mutex_unlock(&codec->mutex); dapm_power_widgets(codec, event); - dump_dapm(codec, __FUNCTION__); + dump_dapm(codec, __func__); return 0; } EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event); @@ -1334,10 +1334,11 @@ int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec, list_for_each_entry(w, &codec->dapm_widgets, list) { if (!strcmp(w->name, endpoint)) { w->connected = status; + return 0; } } - return 0; + return -ENODEV; } EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint); |