o Mic - Mic (and optional Jack)
o Line - Line Input/Output (and optional Jack)
o Speaker - Speaker
+ o Supply - Power or clock supply widget used by other widgets.
o Pre - Special PRE widget (exec before all others)
o Post - Special POST widget (exec after all others)
F: drivers/spi/pxa2xx*
F: drivers/usb/gadget/pxa2*
F: include/sound/pxa2xx-lib.h
-F: sound/soc/pxa/pxa2xx*
+F: sound/arm/pxa*
+F: sound/soc/pxa
PXA168 SUPPORT
P: Eric Miao
M: lrg@slimlogic.co.uk
P: Mark Brown
M: broonie@opensource.wolfsonmicro.com
-T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc
+T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git
L: alsa-devel@alsa-project.org (subscribers-only)
W: http://alsa-project.org/main/index.php/ASoC
S: Supported
F: sound/soc/
+F: include/sound/soc*
SPARC + UltraSPARC (sparc/sparc64)
P: David S. Miller
#define MPC52xx_PSC_MAXNUM 6
/* Programmable Serial Controller (PSC) status register bits */
+#define MPC52xx_PSC_SR_UNEX_RX 0x0001
+#define MPC52xx_PSC_SR_DATA_VAL 0x0002
+#define MPC52xx_PSC_SR_DATA_OVR 0x0004
+#define MPC52xx_PSC_SR_CMDSEND 0x0008
#define MPC52xx_PSC_SR_CDE 0x0080
#define MPC52xx_PSC_SR_RXRDY 0x0100
#define MPC52xx_PSC_SR_RXFULL 0x0200
#define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001
/* PSC interrupt status/mask bits */
+#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001
+#define MPC52xx_PSC_IMR_DATA_VALID 0x0002
+#define MPC52xx_PSC_IMR_DATA_OVR 0x0004
+#define MPC52xx_PSC_IMR_CMD_SEND 0x0008
+#define MPC52xx_PSC_IMR_ERROR 0x0040
+#define MPC52xx_PSC_IMR_DEOF 0x0080
#define MPC52xx_PSC_IMR_TXRDY 0x0100
#define MPC52xx_PSC_IMR_RXRDY 0x0200
#define MPC52xx_PSC_IMR_DB 0x0400
#define MPC52xx_PSC_SICR_SIM_FIR (0x6 << 24)
#define MPC52xx_PSC_SICR_SIM_CODEC_24 (0x7 << 24)
#define MPC52xx_PSC_SICR_SIM_CODEC_32 (0xf << 24)
+#define MPC52xx_PSC_SICR_AWR (1 << 30)
#define MPC52xx_PSC_SICR_GENCLK (1 << 23)
#define MPC52xx_PSC_SICR_I2S (1 << 22)
#define MPC52xx_PSC_SICR_CLKPOL (1 << 21)
#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */
#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */
-/*
- * DAI Left/Right Clocks.
- *
- * Specifies whether the DAI can support different samples for similtanious
- * playback and capture. This usually requires a seperate physical frame
- * clock for playback and capture.
- */
-#define SND_SOC_DAIFMT_SYNC (0 << 5) /* Tx FRM = Rx FRM */
-#define SND_SOC_DAIFMT_ASYNC (1 << 5) /* Tx FRM ~ Rx FRM */
-
-/*
- * TDM
- *
- * Time Division Multiplexing. Allows PCM data to be multplexed with other
- * data on the DAI.
- */
-#define SND_SOC_DAIFMT_TDM (1 << 6)
-
/*
* DAI hardware signal inversions.
*
#define SND_SOC_CLOCK_IN 0
#define SND_SOC_CLOCK_OUT 1
+#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S32_LE |\
+ SNDRV_PCM_FMTBIT_S32_BE)
+
struct snd_soc_dai_ops;
struct snd_soc_dai;
struct snd_ac97_bus_ops;
/* DAI capabilities */
struct snd_soc_pcm_stream capture;
struct snd_soc_pcm_stream playback;
+ unsigned int symmetric_rates:1;
/* DAI runtime info */
struct snd_pcm_runtime *runtime;
/* DAI private data */
void *private_data;
- /* parent codec/platform */
- union {
- struct snd_soc_codec *codec;
- struct snd_soc_platform *platform;
- };
+ /* parent platform */
+ struct snd_soc_platform *platform;
struct list_head list;
};
#define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert}
+#define SND_SOC_DAPM_DAC_E(wname, stname, wreg, wshift, winvert, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
+ .shift = wshift, .invert = winvert, \
+ .event = wevent, .event_flags = wflags}
#define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \
{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
.shift = wshift, .invert = winvert}
+#define SND_SOC_DAPM_ADC_E(wname, stname, wreg, wshift, winvert, \
+ wevent, wflags) \
+{ .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
+ .shift = wshift, .invert = winvert, \
+ .event = wevent, .event_flags = wflags}
-/* generic register modifier widget */
+/* generic widgets */
#define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \
{ .id = wid, .name = wname, .kcontrols = NULL, .num_kcontrols = 0, \
.reg = -((wreg) + 1), .shift = wshift, .mask = wmask, \
.on_val = won_val, .off_val = woff_val, .event = dapm_reg_event, \
.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD}
+#define SND_SOC_DAPM_SUPPLY(wname, wreg, wshift, winvert, wevent, wflags) \
+{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \
+ .shift = wshift, .invert = winvert, .event = wevent, \
+ .event_flags = wflags}
/* dapm kcontrol types */
#define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \
/* dapm events */
int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream,
int event);
-int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
- enum snd_soc_bias_level level);
/* dapm sys fs - used by the core */
int snd_soc_dapm_sys_add(struct device *dev);
snd_soc_dapm_vmid, /* codec bias/vmid - to minimise pops */
snd_soc_dapm_pre, /* machine specific pre widget - exec first */
snd_soc_dapm_post, /* machine specific post widget - exec last */
+ snd_soc_dapm_supply, /* power/clock supply */
};
/*
unsigned char suspend:1; /* was active before suspend */
unsigned char pmdown:1; /* waiting for timeout */
+ int (*power_check)(struct snd_soc_dapm_widget *w);
+
/* external events */
unsigned short event_flags; /* flags to specify event types */
int (*event)(struct snd_soc_dapm_widget*, struct snd_kcontrol *, int);
/* widget input and outputs */
struct list_head sources;
struct list_head sinks;
+
+ /* used during DAPM updates */
+ struct list_head power_list;
};
#endif
.info = snd_soc_info_volsw, \
.get = xhandler_get, .put = xhandler_put, \
.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
+#define SOC_DOUBLE_EXT(xname, xreg, shift_left, shift_right, xmax, xinvert,\
+ xhandler_get, xhandler_put) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+ .info = snd_soc_info_volsw, \
+ .get = xhandler_get, .put = xhandler_put, \
+ .private_value = (unsigned long)&(struct soc_mixer_control) \
+ {.reg = xreg, .shift = shift_left, .rshift = shift_right, \
+ .max = xmax, .invert = xinvert} }
#define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\
xhandler_get, xhandler_put, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
struct snd_soc_jack_gpio *gpios);
#endif
-/* codec IO */
-#define snd_soc_read(codec, reg) codec->read(codec, reg)
-#define snd_soc_write(codec, reg, value) codec->write(codec, reg, value)
-
/* codec register bit access */
int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
unsigned short mask, unsigned short value);
struct module *owner;
struct mutex mutex;
struct device *dev;
+ struct snd_soc_device *socdev;
struct list_head list;
enum snd_soc_bias_level bias_level;
enum snd_soc_bias_level suspend_bias_level;
struct delayed_work delayed_work;
+ struct list_head up_list;
+ struct list_head down_list;
/* codec DAI's */
struct snd_soc_dai *dai;
/* codec/machine specific init - e.g. add machine controls */
int (*init)(struct snd_soc_codec *codec);
+ /* Symmetry requirements */
+ unsigned int symmetric_rates:1;
+
+ /* Symmetry data - only valid if symmetry is being enforced */
+ unsigned int rate;
+
/* DAI pcm */
struct snd_pcm *pcm;
};
void *dapm;
};
+/* codec IO */
+static inline unsigned int snd_soc_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ return codec->read(codec, reg);
+}
+
+static inline unsigned int snd_soc_write(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int val)
+{
+ return codec->write(codec, reg, val);
+}
+
#include <sound/soc-dai.h>
#endif
--- /dev/null
+/*
+ * linux/sound/wm9081.h -- Platform data for WM9081
+ *
+ * Copyright 2009 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM_9081_H
+#define __LINUX_SND_WM_9081_H
+
+struct wm9081_retune_mobile_setting {
+ const char *name;
+ unsigned int rate;
+ u16 config[20];
+};
+
+struct wm9081_retune_mobile_config {
+ struct wm9081_retune_mobile_setting *configs;
+ int num_configs;
+};
+
+#endif
}
ldev->selfptr_headphone.ptr = ldev;
ldev->selfptr_lineout.ptr = ldev;
- sdev->ofdev.dev.driver_data = ldev;
+ dev_set_drvdata(&sdev->ofdev.dev, ldev);
list_add(&ldev->list, &layouts_list);
layouts_list_items++;
static int aoa_fabric_layout_remove(struct soundbus_dev *sdev)
{
- struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+ struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
int i;
for (i=0; i<MAX_CODECS_PER_BUS; i++) {
#ifdef CONFIG_PM
static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t state)
{
- struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+ struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
ldev->gpio.methods->all_amps_off(&ldev->gpio);
static int aoa_fabric_layout_resume(struct soundbus_dev *sdev)
{
- struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+ struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
ldev->gpio.methods->all_amps_restore(&ldev->gpio);
return -ENODEV;
}
- dev->ofdev.dev.driver_data = control;
+ dev_set_drvdata(&dev->ofdev.dev, control);
return 0;
}
static int i2sbus_remove(struct macio_dev* dev)
{
- struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+ struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
struct i2sbus_dev *i2sdev, *tmp;
list_for_each_entry_safe(i2sdev, tmp, &control->list, item)
#ifdef CONFIG_PM
static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state)
{
- struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+ struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
struct codec_info_item *cii;
struct i2sbus_dev* i2sdev;
int err, ret = 0;
static int i2sbus_resume(struct macio_dev* dev)
{
- struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+ struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
struct codec_info_item *cii;
struct i2sbus_dev* i2sdev;
int err, ret = 0;
source "sound/soc/omap/Kconfig"
source "sound/soc/pxa/Kconfig"
source "sound/soc/s3c24xx/Kconfig"
+source "sound/soc/s6000/Kconfig"
source "sound/soc/sh/Kconfig"
+source "sound/soc/txx9/Kconfig"
# Supported codecs
source "sound/soc/codecs/Kconfig"
obj-$(CONFIG_SND_SOC) += omap/
obj-$(CONFIG_SND_SOC) += pxa/
obj-$(CONFIG_SND_SOC) += s3c24xx/
+obj-$(CONFIG_SND_SOC) += s6000/
obj-$(CONFIG_SND_SOC) += sh/
+obj-$(CONFIG_SND_SOC) += txx9/
and FRAME signals on the PlayPaq. Unless you want to play
with the AT32 as the SSC master, you probably want to say N here,
as this will give you better sound quality.
+
+config SND_AT91_SOC_AFEB9260
+ tristate "SoC Audio support for AFEB9260 board"
+ depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
+ select SND_ATMEL_SOC_SSC
+ select SND_SOC_TLV320AIC23
+ help
+ Say Y here to support sound on AFEB9260 board.
obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
+obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
* Find actual rate, compare to requested rate
*/
actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1));
- pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n",
+ pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n",
rate, actual_rate);
--- /dev/null
+/*
+ * afeb9260.c -- SoC audio for AFEB9260
+ *
+ * Copyright (C) 2009 Sergey Lapin <slapin@ossfans.org>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+
+#include <linux/atmel-ssc.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+
+#include "../codecs/tlv320aic23.h"
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
+
+#define CODEC_CLOCK 12000000
+
+static int afeb9260_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+ struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+ int err;
+
+ /* Set codec DAI configuration */
+ err = snd_soc_dai_set_fmt(codec_dai,
+ SND_SOC_DAIFMT_I2S|
+ SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (err < 0) {
+ printk(KERN_ERR "can't set codec DAI configuration\n");
+ return err;
+ }
+
+ /* Set cpu DAI configuration */
+ err = snd_soc_dai_set_fmt(cpu_dai,
+ SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_IF |
+ SND_SOC_DAIFMT_CBM_CFM);
+ if (err < 0) {
+ printk(KERN_ERR "can't set cpu DAI configuration\n");
+ return err;
+ }
+
+ /* Set the codec system clock for DAC and ADC */
+ err =
+ snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
+
+ if (err < 0) {
+ printk(KERN_ERR "can't set codec system clock\n");
+ return err;
+ }
+
+ return err;
+}
+
+static struct snd_soc_ops afeb9260_ops = {
+ .hw_params = afeb9260_hw_params,
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_LINE("Line In", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ {"Headphone Jack", NULL, "LHPOUT"},
+ {"Headphone Jack", NULL, "RHPOUT"},
+
+ {"LLINEIN", NULL, "Line In"},
+ {"RLINEIN", NULL, "Line In"},
+
+ {"MICIN", NULL, "Mic Jack"},
+};
+
+static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+
+ /* Add afeb9260 specific widgets */
+ snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+ ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+ /* Set up afeb9260 specific audio path audio_map */
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+ snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+ snd_soc_dapm_enable_pin(codec, "Line In");
+ snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+ snd_soc_dapm_sync(codec);
+
+ return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link afeb9260_dai = {
+ .name = "TLV320AIC23",
+ .stream_name = "AIC23",
+ .cpu_dai = &atmel_ssc_dai[0],
+ .codec_dai = &tlv320aic23_dai,
+ .init = afeb9260_tlv320aic23_init,
+ .ops = &afeb9260_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_machine_afeb9260 = {
+ .name = "AFEB9260",
+ .platform = &atmel_soc_platform,
+ .dai_link = &afeb9260_dai,
+ .num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device afeb9260_snd_devdata = {
+ .card = &snd_soc_machine_afeb9260,
+ .codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *afeb9260_snd_device;
+
+static int __init afeb9260_soc_init(void)
+{
+ int err;
+ struct device *dev;
+ struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data;
+ struct ssc_device *ssc = NULL;
+
+ if (!(machine_is_afeb9260()))
+ return -ENODEV;
+
+ ssc = ssc_request(0);
+ if (IS_ERR(ssc)) {
+ printk(KERN_ERR "ASoC: Failed to request SSC 0\n");
+ err = PTR_ERR(ssc);
+ ssc = NULL;
+ goto err_ssc;
+ }
+ ssc_p->ssc = ssc;
+
+ afeb9260_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!afeb9260_snd_device) {
+ printk(KERN_ERR "ASoC: Platform device allocation failed\n");
+ return -ENOMEM;
+ }
+
+ platform_set_drvdata(afeb9260_snd_device, &afeb9260_snd_devdata);
+ afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev;
+ err = platform_device_add(afeb9260_snd_device);
+ if (err)
+ goto err1;
+
+ dev = &afeb9260_snd_device->dev;
+
+ return 0;
+err1:
+ platform_device_del(afeb9260_snd_device);
+ platform_device_put(afeb9260_snd_device);
+err_ssc:
+ return err;
+
+}
+
+static void __exit afeb9260_soc_exit(void)
+{
+ platform_device_unregister(afeb9260_snd_device);
+}
+
+module_init(afeb9260_soc_init);
+module_exit(afeb9260_soc_exit);
+
+MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>");
+MODULE_DESCRIPTION("ALSA SoC for AFEB9260");
+MODULE_LICENSE("GPL");
+
#include "bf5xx-sport.h"
#include "bf5xx-ac97.h"
+/* Anomaly notes:
+ * 05000250 - AD1980 is running in TDM mode and RFS/TFS are generated by SPORT
+ * contrtoller. But, RFSDIV and TFSDIV are always set to 16*16-1,
+ * while the max AC97 data size is 13*16. The DIV is always larger
+ * than data size. AD73311 and ad2602 are not running in TDM mode.
+ * AD1836 and AD73322 depend on external RFS/TFS only. So, this
+ * anomaly does not affect blackfin sound drivers.
+*/
+
static int *cmd_count;
static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
desc = get_dma_next_desc_ptr(sport->dma_rx_chan);
/* Copy the descriptor which will be damaged to backup */
temp_desc = *desc;
- desc->x_count = 0xa;
+ desc->x_count = sport->dummy_count / 2;
desc->y_count = 0;
desc->next_desc_addr = sport->dummy_rx_desc;
local_irq_restore(flags);
desc = get_dma_next_desc_ptr(sport->dma_tx_chan);
/* Store the descriptor which will be damaged */
temp_desc = *desc;
- desc->x_count = 0xa;
+ desc->x_count = sport->dummy_count / 2;
desc->y_count = 0;
desc->next_desc_addr = sport->dummy_tx_desc;
local_irq_restore(flags);
select SND_SOC_AK4535 if I2C
select SND_SOC_CS4270 if I2C
select SND_SOC_PCM3008
+ select SND_SOC_SPDIF
select SND_SOC_SSM2602 if I2C
+ select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
select SND_SOC_TLV320AIC23 if I2C
select SND_SOC_TLV320AIC26 if SPI_MASTER
select SND_SOC_TLV320AIC3X if I2C
select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8900 if I2C
select SND_SOC_WM8903 if I2C
+ select SND_SOC_WM8940 if I2C
+ select SND_SOC_WM8960 if I2C
select SND_SOC_WM8971 if I2C
+ select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8990 if I2C
+ select SND_SOC_WM9081 if I2C
select SND_SOC_WM9705 if SND_SOC_AC97_BUS
select SND_SOC_WM9712 if SND_SOC_AC97_BUS
select SND_SOC_WM9713 if SND_SOC_AC97_BUS
config SND_SOC_PCM3008
tristate
+config SND_SOC_SPDIF
+ tristate
+
config SND_SOC_SSM2602
tristate
+config SND_SOC_STAC9766
+ tristate
+
config SND_SOC_TLV320AIC23
tristate
config SND_SOC_WM8903
tristate
+config SND_SOC_WM8940
+ tristate
+
+config SND_SOC_WM8960
+ tristate
+
config SND_SOC_WM8971
tristate
+config SND_SOC_WM8988
+ tristate
+
config SND_SOC_WM8990
tristate
+config SND_SOC_WM9081
+ tristate
+
config SND_SOC_WM9705
tristate
snd-soc-cs4270-objs := cs4270.o
snd-soc-l3-objs := l3.o
snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-spdif-objs := spdif_transciever.o
snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-stac9766-objs := stac9766.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
snd-soc-wm8753-objs := wm8753.o
snd-soc-wm8900-objs := wm8900.o
snd-soc-wm8903-objs := wm8903.o
+snd-soc-wm8940-objs := wm8940.o
+snd-soc-wm8960-objs := wm8960.o
snd-soc-wm8971-objs := wm8971.o
+snd-soc-wm8988-objs := wm8988.o
snd-soc-wm8990-objs := wm8990.o
+snd-soc-wm9081-objs := wm9081.o
snd-soc-wm9705-objs := wm9705.o
snd-soc-wm9712-objs := wm9712.o
snd-soc-wm9713-objs := wm9713.o
obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o
obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o
obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o
+obj-$(CONFIG_SND_SOC_WM8940) += snd-soc-wm8940.o
+obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o
+obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o
obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o
+obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o
obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o
obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o
obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o
.channels_min = 1,
.channels_max = 2,
.rates = STD_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.capture = {
.stream_name = "AC97 Capture",
.channels_min = 1,
.channels_max = 2,
.rates = STD_AC97_RATES,
- .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .formats = SND_SOC_STD_AC97_FMTS,},
.ops = &ac97_dai_ops,
};
EXPORT_SYMBOL_GPL(ac97_dai);
.channels_min = 2,
.channels_max = 6,
.rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+ .formats = SND_SOC_STD_AC97_FMTS, },
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_48000,
- .formats = SNDRV_PCM_FMTBIT_S16_LE, },
+ .formats = SND_SOC_STD_AC97_FMTS, },
};
EXPORT_SYMBOL_GPL(ad1980_dai);
* - The machine driver's 'startup' function must call
* cs4270_set_dai_sysclk() with the value of MCLK.
* - Only I2S and left-justified modes are supported
- * - Power management is not supported
+ * - Power management is supported
*/
#include <linux/module.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <linux/i2c.h>
+#include <linux/delay.h>
#include "cs4270.h"
#define CS4270_FIRSTREG 0x01
#define CS4270_LASTREG 0x08
#define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1)
+#define CS4270_I2C_INCR 0x80
/* Bit masks for the CS4270 registers */
#define CS4270_CHIPID_ID 0xF0
#define CS4270_PWRCTL_PDN_ADC 0x20
#define CS4270_PWRCTL_PDN_DAC 0x02
#define CS4270_PWRCTL_PDN 0x01
+#define CS4270_PWRCTL_PDN_ALL \
+ (CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | CS4270_PWRCTL_PDN)
#define CS4270_MODE_SPEED_MASK 0x30
#define CS4270_MODE_1X 0x00
#define CS4270_MODE_2X 0x10
unsigned int mclk; /* Input frequency of the MCLK pin */
unsigned int mode; /* The mode (I2S or left-justified) */
unsigned int slave_mode;
+ unsigned int manual_mute;
};
/**
s32 length;
length = i2c_smbus_read_i2c_block_data(i2c_client,
- CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache);
+ CS4270_FIRSTREG | CS4270_I2C_INCR, CS4270_NUMREGS, cache);
if (length != CS4270_NUMREGS) {
dev_err(codec->dev, "i2c read failure, addr=0x%x\n",
}
/**
- * cs4270_mute - enable/disable the CS4270 external mute
+ * cs4270_dai_mute - enable/disable the CS4270 external mute
* @dai: the SOC DAI
* @mute: 0 = disable mute, 1 = enable mute
*
* board does not have the MUTEA or MUTEB pins connected to such circuitry,
* then this function will do nothing.
*/
-static int cs4270_mute(struct snd_soc_dai *dai, int mute)
+static int cs4270_dai_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
+ struct cs4270_private *cs4270 = codec->private_data;
int reg6;
reg6 = snd_soc_read(codec, CS4270_MUTE);
if (mute)
reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B;
- else
+ else {
reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B);
+ reg6 |= cs4270->manual_mute;
+ }
return snd_soc_write(codec, CS4270_MUTE, reg6);
}
+/**
+ * cs4270_soc_put_mute - put callback for the 'Master Playback switch'
+ * alsa control.
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * This function basically passes the arguments on to the generic
+ * snd_soc_put_volsw() function and saves the mute information in
+ * our private data structure. This is because we want to prevent
+ * cs4270_dai_mute() neglecting the user's decision to manually
+ * mute the codec's output.
+ *
+ * Returns 0 for success.
+ */
+static int cs4270_soc_put_mute(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct cs4270_private *cs4270 = codec->private_data;
+ int left = !ucontrol->value.integer.value[0];
+ int right = !ucontrol->value.integer.value[1];
+
+ cs4270->manual_mute = (left ? CS4270_MUTE_DAC_A : 0) |
+ (right ? CS4270_MUTE_DAC_B : 0);
+
+ return snd_soc_put_volsw(kcontrol, ucontrol);
+}
+
/* A list of non-DAPM controls that the CS4270 supports */
static const struct snd_kcontrol_new cs4270_snd_controls[] = {
SOC_DOUBLE_R("Master Playback Volume",
SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0),
SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1),
SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0),
- SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0)
+ SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1),
+ SOC_DOUBLE_EXT("Master Playback Switch", CS4270_MUTE, 0, 1, 1, 1,
+ snd_soc_get_volsw, cs4270_soc_put_mute),
};
/*
.hw_params = cs4270_hw_params,
.set_sysclk = cs4270_set_dai_sysclk,
.set_fmt = cs4270_set_dai_fmt,
- .digital_mute = cs4270_mute,
+ .digital_mute = cs4270_dai_mute,
};
struct snd_soc_dai cs4270_dai = {
};
MODULE_DEVICE_TABLE(i2c, cs4270_id);
+#ifdef CONFIG_PM
+
+/* This suspend/resume implementation can handle both - a simple standby
+ * where the codec remains powered, and a full suspend, where the voltage
+ * domain the codec is connected to is teared down and/or any other hardware
+ * reset condition is asserted.
+ *
+ * The codec's own power saving features are enabled in the suspend callback,
+ * and all registers are written back to the hardware when resuming.
+ */
+
+static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg)
+{
+ struct cs4270_private *cs4270 = i2c_get_clientdata(client);
+ struct snd_soc_codec *codec = &cs4270->codec;
+ int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL;
+
+ return snd_soc_write(codec, CS4270_PWRCTL, reg);
+}
+
+static int cs4270_i2c_resume(struct i2c_client *client)
+{
+ struct cs4270_private *cs4270 = i2c_get_clientdata(client);
+ struct snd_soc_codec *codec = &cs4270->codec;
+ int reg;
+
+ /* In case the device was put to hard reset during sleep, we need to
+ * wait 500ns here before any I2C communication. */
+ ndelay(500);
+
+ /* first restore the entire register cache ... */
+ for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) {
+ u8 val = snd_soc_read(codec, reg);
+
+ if (i2c_smbus_write_byte_data(client, reg, val)) {
+ dev_err(codec->dev, "i2c write failed\n");
+ return -EIO;
+ }
+ }
+
+ /* ... then disable the power-down bits */
+ reg = snd_soc_read(codec, CS4270_PWRCTL);
+ reg &= ~CS4270_PWRCTL_PDN_ALL;
+
+ return snd_soc_write(codec, CS4270_PWRCTL, reg);
+}
+#else
+#define cs4270_i2c_suspend NULL
+#define cs4270_i2c_resume NULL
+#endif /* CONFIG_PM */
+
/*
* cs4270_i2c_driver - I2C device identification
*
.id_table = cs4270_id,
.probe = cs4270_i2c_probe,
.remove = cs4270_i2c_remove,
+ .suspend = cs4270_i2c_suspend,
+ .resume = cs4270_i2c_resume,
};
/*
--- /dev/null
+/*
+ * ALSA SoC SPDIF DIT driver
+ *
+ * This driver is used by controllers which can operate in DIT (SPDI/F) where
+ * no codec is needed. This file provides stub codec that can be used
+ * in these configurations. TI DaVinci Audio controller uses this driver.
+ *
+ * Author: Steve Chen, <schen@mvista.com>
+ * Copyright: (C) 2009 MontaVista Software, Inc., <source@mvista.com>
+ * Copyright: (C) 2009 Texas Instruments, India
+ *
+ * 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 <sound/soc.h>
+#include <sound/pcm.h>
+
+#include "spdif_transciever.h"
+
+#define STUB_RATES SNDRV_PCM_RATE_8000_96000
+#define STUB_FORMATS SNDRV_PCM_FMTBIT_S16_LE
+
+struct snd_soc_dai dit_stub_dai = {
+ .name = "DIT",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 384,
+ .rates = STUB_RATES,
+ .formats = STUB_FORMATS,
+ },
+};
+
+static int spdif_dit_probe(struct platform_device *pdev)
+{
+ dit_stub_dai.dev = &pdev->dev;
+ return snd_soc_register_dai(&dit_stub_dai);
+}
+
+static int spdif_dit_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_dai(&dit_stub_dai);
+ return 0;
+}
+
+static struct platform_driver spdif_dit_driver = {
+ .probe = spdif_dit_probe,
+ .remove = spdif_dit_remove,
+ .driver = {
+ .name = "spdif-dit",
+ .owner = THIS_MODULE,
+ },
+};
+
+static int __init dit_modinit(void)
+{
+ return platform_driver_register(&spdif_dit_driver);
+}
+
+static void __exit dit_exit(void)
+{
+ platform_driver_unregister(&spdif_dit_driver);
+}
+
+module_init(dit_modinit);
+module_exit(dit_exit);
+
--- /dev/null
+/*
+ * ALSA SoC DIT/DIR driver header
+ *
+ * Author: Steve Chen, <schen@mvista.com>
+ * Copyright: (C) 2008 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 CODEC_STUBS_H
+#define CODEC_STUBS_H
+
+extern struct snd_soc_dai dit_stub_dai;
+
+#endif /* CODEC_STUBS_H */
master_runtime->sample_bits,
master_runtime->rate);
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_RATE,
- master_runtime->rate,
- master_runtime->rate);
-
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
- master_runtime->sample_bits,
- master_runtime->sample_bits);
+ if (master_runtime->rate != 0)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ master_runtime->rate,
+ master_runtime->rate);
+
+ if (master_runtime->sample_bits != 0)
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ master_runtime->sample_bits,
+ master_runtime->sample_bits);
ssm2602->slave_substream = substream;
} else
struct snd_soc_device *socdev = rtd->socdev;
struct snd_soc_codec *codec = socdev->card->codec;
struct ssm2602_priv *ssm2602 = codec->private_data;
+
+ if (ssm2602->master_substream == substream)
+ ssm2602->master_substream = ssm2602->slave_substream;
+
+ ssm2602->slave_substream = NULL;
/* deactivate */
if (!codec->active)
ssm2602_write(codec, SSM2602_ACTIVE, 0);
return 0;
}
-#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
- 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)
+#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+ SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
#define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
--- /dev/null
+/*
+ * stac9766.c -- ALSA SoC STAC9766 codec support
+ *
+ * Copyright 2009 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.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.
+ *
+ * Features:-
+ *
+ * o Support for AC97 Codec, S/PDIF
+ */
+
+#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/tlv.h>
+#include <sound/soc-of-simple.h>
+
+#include "stac9766.h"
+
+#define STAC9766_VERSION "0.10"
+
+/*
+ * STAC9766 register cache
+ */
+static const u16 stac9766_reg[] = {
+ 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
+ 0x0000, 0x0000, 0x8008, 0x8008, /* e */
+ 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
+ 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
+ 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
+ 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
+ 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
+ 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
+ 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
+ 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
+ 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
+ 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
+ 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
+};
+
+static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
+ "Line", "Stereo Mix", "Mono Mix", "Phone"};
+static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
+static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
+static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
+static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
+static const char *stac9766_record_all_mux[] = {"All analog",
+ "Analog plus DAC"};
+static const char *stac9766_boost1[] = {"0dB", "10dB"};
+static const char *stac9766_boost2[] = {"0dB", "20dB"};
+static const char *stac9766_stereo_mic[] = {"Off", "On"};
+
+static const struct soc_enum stac9766_record_enum =
+ SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
+static const struct soc_enum stac9766_mono_enum =
+ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
+static const struct soc_enum stac9766_mic_enum =
+ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
+static const struct soc_enum stac9766_SPDIF_enum =
+ SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
+static const struct soc_enum stac9766_popbypass_enum =
+ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
+static const struct soc_enum stac9766_record_all_enum =
+ SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
+ stac9766_record_all_mux);
+static const struct soc_enum stac9766_boost1_enum =
+ SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
+static const struct soc_enum stac9766_boost2_enum =
+ SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
+static const struct soc_enum stac9766_stereo_mic_enum =
+ SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic);
+
+static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
+static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
+static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
+static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);
+
+static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
+ SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
+ SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
+ SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
+ master_tlv),
+ SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
+ SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
+ master_tlv),
+ SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),
+
+ SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
+ SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),
+
+
+ SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
+ SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
+ SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
+ SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),
+
+ SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
+ SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
+ SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
+ SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),
+
+ SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
+ SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
+ SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
+ SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),
+
+ SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
+ SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
+ SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
+ SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
+
+ SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
+ SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
+ SOC_ENUM("Record All Mux", stac9766_record_all_enum),
+ SOC_ENUM("Record Mux", stac9766_record_enum),
+ SOC_ENUM("Mono Mux", stac9766_mono_enum),
+ SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
+};
+
+static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val)
+{
+ u16 *cache = codec->reg_cache;
+
+ if (reg > AC97_STAC_PAGE0) {
+ stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
+ return 0;
+ }
+ if (reg / 2 > ARRAY_SIZE(stac9766_reg))
+ return -EIO;
+
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ cache[reg / 2] = val;
+ return 0;
+}
+
+static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 val = 0, *cache = codec->reg_cache;
+
+ if (reg > AC97_STAC_PAGE0) {
+ stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
+ val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0);
+ stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
+ return val;
+ }
+ if (reg / 2 > ARRAY_SIZE(stac9766_reg))
+ return -EIO;
+
+ if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
+ reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
+ reg == AC97_VENDOR_ID2) {
+
+ val = soc_ac97_ops.read(codec->ac97, reg);
+ return val;
+ }
+ return cache[reg / 2];
+}
+
+static int ac97_analog_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned short reg, vra;
+
+ vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+
+ vra |= 0x1; /* enable variable rate audio */
+
+ stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ reg = AC97_PCM_FRONT_DAC_RATE;
+ else
+ reg = AC97_PCM_LR_ADC_RATE;
+
+ return stac9766_ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_digital_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned short reg, vra;
+
+ stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);
+
+ vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+ vra |= 0x5; /* Enable VRA and SPDIF out */
+
+ stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+
+ reg = AC97_PCM_FRONT_DAC_RATE;
+
+ return stac9766_ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_digital_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned short vra;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_STOP:
+ vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+ vra &= !0x04;
+ stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+ break;
+ }
+ return 0;
+}
+
+static int stac9766_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_ON: /* full On */
+ case SND_SOC_BIAS_PREPARE: /* partial On */
+ case SND_SOC_BIAS_STANDBY: /* Off, with power */
+ stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
+ break;
+ case SND_SOC_BIAS_OFF: /* Off, without power */
+ /* disable everything including AC link */
+ stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+static int stac9766_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 (stac9766_ac97_read(codec, 0) == stac9766_reg[0])
+ return 1;
+ }
+
+ soc_ac97_ops.reset(codec->ac97);
+ if (soc_ac97_ops.warm_reset)
+ soc_ac97_ops.warm_reset(codec->ac97);
+ if (stac9766_ac97_read(codec, 0) != stac9766_reg[0])
+ return -EIO;
+ return 0;
+}
+
+static int stac9766_codec_suspend(struct platform_device *pdev,
+ pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int stac9766_codec_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ u16 id, reset;
+
+ reset = 0;
+ /* give the codec an AC97 warm reset to start the link */
+reset:
+ if (reset > 5) {
+ printk(KERN_ERR "stac9766 failed to resume");
+ return -EIO;
+ }
+ codec->ac97->bus->ops->warm_reset(codec->ac97);
+ id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2);
+ if (id != 0x4c13) {
+ stac9766_reset(codec, 0);
+ reset++;
+ goto reset;
+ }
+ stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+ stac9766_set_bias_level(codec, SND_SOC_BIAS_ON);
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops stac9766_dai_ops_analog = {
+ .prepare = ac97_analog_prepare,
+};
+
+static struct snd_soc_dai_ops stac9766_dai_ops_digital = {
+ .prepare = ac97_digital_prepare,
+ .trigger = ac97_digital_trigger,
+};
+
+struct snd_soc_dai stac9766_dai[] = {
+{
+ .name = "stac9766 analog",
+ .id = 0,
+ .ac97_control = 1,
+
+ /* stream cababilities */
+ .playback = {
+ .stream_name = "stac9766 analog",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SND_SOC_STD_AC97_FMTS,
+ },
+ .capture = {
+ .stream_name = "stac9766 analog",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SND_SOC_STD_AC97_FMTS,
+ },
+ /* alsa ops */
+ .ops = &stac9766_dai_ops_analog,
+},
+{
+ .name = "stac9766 IEC958",
+ .id = 1,
+ .ac97_control = 1,
+
+ /* stream cababilities */
+ .playback = {
+ .stream_name = "stac9766 IEC958",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE,
+ },
+ /* alsa ops */
+ .ops = &stac9766_dai_ops_digital,
+}
+};
+EXPORT_SYMBOL_GPL(stac9766_dai);
+
+static int stac9766_codec_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);
+
+ socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->card->codec == NULL)
+ return -ENOMEM;
+ codec = socdev->card->codec;
+ mutex_init(&codec->mutex);
+
+ codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg),
+ GFP_KERNEL);
+ if (codec->reg_cache == NULL) {
+ ret = -ENOMEM;
+ goto cache_err;
+ }
+ codec->reg_cache_size = sizeof(stac9766_reg);
+ codec->reg_cache_step = 2;
+
+ codec->name = "STAC9766";
+ codec->owner = THIS_MODULE;
+ codec->dai = stac9766_dai;
+ codec->num_dai = ARRAY_SIZE(stac9766_dai);
+ codec->write = stac9766_ac97_write;
+ codec->read = stac9766_ac97_read;
+ codec->set_bias_level = stac9766_set_bias_level;
+ 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 */
+ stac9766_reset(codec, 0);
+ ret = stac9766_reset(codec, 1);
+ if (ret < 0) {
+ printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n");
+ goto reset_err;
+ }
+
+ stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ snd_soc_add_controls(codec, stac9766_snd_ac97_controls,
+ ARRAY_SIZE(stac9766_snd_ac97_controls));
+
+ ret = snd_soc_init_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);
+cache_err:
+ kfree(socdev->card->codec);
+ socdev->card->codec = NULL;
+ return ret;
+}
+
+static int stac9766_codec_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ if (codec == NULL)
+ return 0;
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_free_ac97_codec(codec);
+ kfree(codec->reg_cache);
+ kfree(codec);
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_stac9766 = {
+ .probe = stac9766_codec_probe,
+ .remove = stac9766_codec_remove,
+ .suspend = stac9766_codec_suspend,
+ .resume = stac9766_codec_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766);
+
+MODULE_DESCRIPTION("ASoC stac9766 driver");
+MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
+MODULE_LICENSE("GPL");
--- /dev/null
+/*
+ * stac9766.h -- STAC9766 Soc Audio driver
+ */
+
+#ifndef _STAC9766_H
+#define _STAC9766_H
+
+#define AC97_STAC_PAGE0 0x1000
+#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A)
+#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E)
+#define AC97_STAC_STEREO_MIC 0x78
+
+/* STAC9766 DAI ID's */
+#define STAC9766_DAI_AC97_ANALOG 0
+#define STAC9766_DAI_AC97_DIGITAL 1
+
+extern struct snd_soc_dai stac9766_dai[];
+extern struct snd_soc_codec_device soc_codec_dev_stac9766;
+
+
+#endif
*/
if ((reg < 0 || reg > 9) && (reg != 15)) {
- printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg);
+ printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg);
return -1;
}
if (codec->hw_write(codec->control_data, data, 2) == 2)
return 0;
- printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__,
+ printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__,
value, reg);
return -EIO;
* Every divisor is a factor of 11*12
*/
#define SR_MULT (11*12)
-#define A(x) (x) ? (SR_MULT/x) : 0
+#define A(x) (SR_MULT/x)
static const unsigned char sr_adc_mult_table[] = {
- A(2), A(2), A(12), A(12), A(0), A(0), A(3), A(1),
- A(2), A(2), A(11), A(11), A(0), A(0), A(0), A(1)
+ A(2), A(2), A(12), A(12), 0, 0, A(3), A(1),
+ A(2), A(2), A(11), A(11), 0, 0, 0, A(1)
};
static const unsigned char sr_dac_mult_table[] = {
- A(2), A(12), A(2), A(12), A(0), A(0), A(3), A(1),
- A(2), A(11), A(2), A(11), A(0), A(0), A(0), A(1)
+ A(2), A(12), A(2), A(12), 0, 0, A(3), A(1),
+ A(2), A(11), A(2), A(11), 0, 0, 0, A(1)
};
static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
case SND_SOC_DAIFMT_I2S:
iface_reg |= TLV320AIC23_FOR_I2S;
break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface_reg |= TLV320AIC23_LRP_ON;
case SND_SOC_DAIFMT_DSP_B:
iface_reg |= TLV320AIC23_FOR_DSP;
break;
0x00, /* REG_VIBRA_PWM_SET (0x47) */
0x00, /* REG_ANAMIC_GAIN (0x48) */
0x00, /* REG_MISC_SET_2 (0x49) */
+ 0x00, /* REG_SW_SHADOW (0x4A) - Shadow, non HW register */
};
/* codec private data */
struct snd_pcm_substream *master_substream;
struct snd_pcm_substream *slave_substream;
+
+ unsigned int configured;
+ unsigned int rate;
+ unsigned int sample_bits;
+ unsigned int channels;
+
+ unsigned int sysclk;
+
+ /* Headset output state handling */
+ unsigned int hsl_enabled;
+ unsigned int hsr_enabled;
};
/*
unsigned int reg, unsigned int value)
{
twl4030_write_reg_cache(codec, reg, value);
- return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
+ if (likely(reg < TWL4030_REG_SW_SHADOW))
+ return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value,
+ reg);
+ else
+ return 0;
}
static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
static void twl4030_init_chip(struct snd_soc_codec *codec)
{
+ u8 *cache = codec->reg_cache;
int i;
/* clear CODECPDZ prior to setting register defaults */
/* set all audio section registers to reasonable defaults */
for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
- twl4030_write(codec, i, twl4030_reg[i]);
+ twl4030_write(codec, i, cache[i]);
}
TWL4030_REG_PRECKL_CTL);
reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL);
twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
- reg_val & (~TWL4030_PRECKL_GAIN),
+ reg_val & (~TWL4030_PRECKR_GAIN),
TWL4030_REG_PRECKR_CTL);
/* Disable PLL */
}
/* Earpiece */
-static const char *twl4030_earpiece_texts[] =
- {"Off", "DACL1", "DACL2", "DACR1"};
-
-static const unsigned int twl4030_earpiece_values[] =
- {0x0, 0x1, 0x2, 0x4};
-
-static const struct soc_enum twl4030_earpiece_enum =
- SOC_VALUE_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, 0x7,
- ARRAY_SIZE(twl4030_earpiece_texts),
- twl4030_earpiece_texts,
- twl4030_earpiece_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_earpiece_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_earpiece_enum);
+static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = {
+ SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0),
+ SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_EAR_CTL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_EAR_CTL, 2, 1, 0),
+ SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_EAR_CTL, 3, 1, 0),
+};
/* PreDrive Left */
-static const char *twl4030_predrivel_texts[] =
- {"Off", "DACL1", "DACL2", "DACR2"};
-
-static const unsigned int twl4030_predrivel_values[] =
- {0x0, 0x1, 0x2, 0x4};
-
-static const struct soc_enum twl4030_predrivel_enum =
- SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, 0x7,
- ARRAY_SIZE(twl4030_predrivel_texts),
- twl4030_predrivel_texts,
- twl4030_predrivel_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_predrivel_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_predrivel_enum);
+static const struct snd_kcontrol_new twl4030_dapm_predrivel_controls[] = {
+ SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDL_CTL, 0, 1, 0),
+ SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PREDL_CTL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDL_CTL, 2, 1, 0),
+ SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDL_CTL, 3, 1, 0),
+};
/* PreDrive Right */
-static const char *twl4030_predriver_texts[] =
- {"Off", "DACR1", "DACR2", "DACL2"};
-
-static const unsigned int twl4030_predriver_values[] =
- {0x0, 0x1, 0x2, 0x4};
-
-static const struct soc_enum twl4030_predriver_enum =
- SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7,
- ARRAY_SIZE(twl4030_predriver_texts),
- twl4030_predriver_texts,
- twl4030_predriver_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_predriver_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_predriver_enum);
+static const struct snd_kcontrol_new twl4030_dapm_predriver_controls[] = {
+ SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDR_CTL, 0, 1, 0),
+ SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PREDR_CTL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDR_CTL, 2, 1, 0),
+ SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDR_CTL, 3, 1, 0),
+};
/* Headset Left */
-static const char *twl4030_hsol_texts[] =
- {"Off", "DACL1", "DACL2"};
-
-static const struct soc_enum twl4030_hsol_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1,
- ARRAY_SIZE(twl4030_hsol_texts),
- twl4030_hsol_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_hsol_control =
-SOC_DAPM_ENUM("Route", twl4030_hsol_enum);
+static const struct snd_kcontrol_new twl4030_dapm_hsol_controls[] = {
+ SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 0, 1, 0),
+ SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_HS_SEL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_HS_SEL, 2, 1, 0),
+};
/* Headset Right */
-static const char *twl4030_hsor_texts[] =
- {"Off", "DACR1", "DACR2"};
-
-static const struct soc_enum twl4030_hsor_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4,
- ARRAY_SIZE(twl4030_hsor_texts),
- twl4030_hsor_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_hsor_control =
-SOC_DAPM_ENUM("Route", twl4030_hsor_enum);
+static const struct snd_kcontrol_new twl4030_dapm_hsor_controls[] = {
+ SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 3, 1, 0),
+ SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_HS_SEL, 4, 1, 0),
+ SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_HS_SEL, 5, 1, 0),
+};
/* Carkit Left */
-static const char *twl4030_carkitl_texts[] =
- {"Off", "DACL1", "DACL2"};
-
-static const struct soc_enum twl4030_carkitl_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1,
- ARRAY_SIZE(twl4030_carkitl_texts),
- twl4030_carkitl_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_carkitl_control =
-SOC_DAPM_ENUM("Route", twl4030_carkitl_enum);
+static const struct snd_kcontrol_new twl4030_dapm_carkitl_controls[] = {
+ SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKL_CTL, 0, 1, 0),
+ SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PRECKL_CTL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PRECKL_CTL, 2, 1, 0),
+};
/* Carkit Right */
-static const char *twl4030_carkitr_texts[] =
- {"Off", "DACR1", "DACR2"};
-
-static const struct soc_enum twl4030_carkitr_enum =
- SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1,
- ARRAY_SIZE(twl4030_carkitr_texts),
- twl4030_carkitr_texts);
-
-static const struct snd_kcontrol_new twl4030_dapm_carkitr_control =
-SOC_DAPM_ENUM("Route", twl4030_carkitr_enum);
+static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = {
+ SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKR_CTL, 0, 1, 0),
+ SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PRECKR_CTL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PRECKR_CTL, 2, 1, 0),
+};
/* Handsfree Left */
static const char *twl4030_handsfreel_texts[] =
- {"Voice", "DACL1", "DACL2", "DACR2"};
+ {"Voice", "AudioL1", "AudioL2", "AudioR2"};
static const struct soc_enum twl4030_handsfreel_enum =
SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0,
static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control =
SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum);
+/* Handsfree Left virtual mute */
+static const struct snd_kcontrol_new twl4030_dapm_handsfreelmute_control =
+ SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 0, 1, 0);
+
/* Handsfree Right */
static const char *twl4030_handsfreer_texts[] =
- {"Voice", "DACR1", "DACR2", "DACL2"};
+ {"Voice", "AudioR1", "AudioR2", "AudioL2"};
static const struct soc_enum twl4030_handsfreer_enum =
SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0,
static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control =
SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum);
-/* Left analog microphone selection */
-static const char *twl4030_analoglmic_texts[] =
- {"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"};
+/* Handsfree Right virtual mute */
+static const struct snd_kcontrol_new twl4030_dapm_handsfreermute_control =
+ SOC_DAPM_SINGLE("Switch", TWL4030_REG_SW_SHADOW, 1, 1, 0);
-static const unsigned int twl4030_analoglmic_values[] =
- {0x0, 0x1, 0x2, 0x4, 0x8};
+/* Vibra */
+/* Vibra audio path selection */
+static const char *twl4030_vibra_texts[] =
+ {"AudioL1", "AudioR1", "AudioL2", "AudioR2"};
-static const struct soc_enum twl4030_analoglmic_enum =
- SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0, 0xf,
- ARRAY_SIZE(twl4030_analoglmic_texts),
- twl4030_analoglmic_texts,
- twl4030_analoglmic_values);
+static const struct soc_enum twl4030_vibra_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 2,
+ ARRAY_SIZE(twl4030_vibra_texts),
+ twl4030_vibra_texts);
-static const struct snd_kcontrol_new twl4030_dapm_analoglmic_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_analoglmic_enum);
+static const struct snd_kcontrol_new twl4030_dapm_vibra_control =
+SOC_DAPM_ENUM("Route", twl4030_vibra_enum);
-/* Right analog microphone selection */
-static const char *twl4030_analogrmic_texts[] =
- {"Off", "Sub mic", "AUXR"};
+/* Vibra path selection: local vibrator (PWM) or audio driven */
+static const char *twl4030_vibrapath_texts[] =
+ {"Local vibrator", "Audio"};
-static const unsigned int twl4030_analogrmic_values[] =
- {0x0, 0x1, 0x4};
+static const struct soc_enum twl4030_vibrapath_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 4,
+ ARRAY_SIZE(twl4030_vibrapath_texts),
+ twl4030_vibrapath_texts);
-static const struct soc_enum twl4030_analogrmic_enum =
- SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0, 0x5,
- ARRAY_SIZE(twl4030_analogrmic_texts),
- twl4030_analogrmic_texts,
- twl4030_analogrmic_values);
+static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control =
+SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum);
-static const struct snd_kcontrol_new twl4030_dapm_analogrmic_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_analogrmic_enum);
+/* Left analog microphone selection */
+static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = {
+ SOC_DAPM_SINGLE("Main mic", TWL4030_REG_ANAMICL, 0, 1, 0),
+ SOC_DAPM_SINGLE("Headset mic", TWL4030_REG_ANAMICL, 1, 1, 0),
+ SOC_DAPM_SINGLE("AUXL", TWL4030_REG_ANAMICL, 2, 1, 0),
+ SOC_DAPM_SINGLE("Carkit mic", TWL4030_REG_ANAMICL, 3, 1, 0),
+};
+
+/* Right analog microphone selection */
+static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = {
+ SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0),
+ SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 2, 1, 0),
+};
/* TX1 L/R Analog/Digital microphone selection */
static const char *twl4030_micpathtx1_texts[] =
static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control =
SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0);
+/* Analog bypass for Voice */
+static const struct snd_kcontrol_new twl4030_dapm_abypassv_control =
+ SOC_DAPM_SINGLE("Switch", TWL4030_REG_VDL_APGA_CTL, 2, 1, 0);
+
/* Digital bypass gain, 0 mutes the bypass */
static const unsigned int twl4030_dapm_dbypass_tlv[] = {
TLV_DB_RANGE_HEAD(2),
TWL4030_REG_ATX2ARXPGA, 0, 7, 0,
twl4030_dapm_dbypass_tlv);
+/*
+ * Voice Sidetone GAIN volume control:
+ * from -51 to -10 dB in 1 dB steps (mute instead of -51 dB)
+ */
+static DECLARE_TLV_DB_SCALE(twl4030_dapm_dbypassv_tlv, -5100, 100, 1);
+
+/* Digital bypass voice: sidetone (VUL -> VDL)*/
+static const struct snd_kcontrol_new twl4030_dapm_dbypassv_control =
+ SOC_DAPM_SINGLE_TLV("Volume",
+ TWL4030_REG_VSTPGA, 0, 0x29, 0,
+ twl4030_dapm_dbypassv_tlv);
+
static int micpath_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
return 0;
}
-static int handsfree_event(struct snd_soc_dapm_widget *w,
- struct snd_kcontrol *kcontrol, int event)
+static void handsfree_ramp(struct snd_soc_codec *codec, int reg, int ramp)
{
- struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value;
unsigned char hs_ctl;
- hs_ctl = twl4030_read_reg_cache(w->codec, e->reg);
+ hs_ctl = twl4030_read_reg_cache(codec, reg);
- if (hs_ctl & TWL4030_HF_CTL_REF_EN) {
+ if (ramp) {
+ /* HF ramp-up */
+ hs_ctl |= TWL4030_HF_CTL_REF_EN;
+ twl4030_write(codec, reg, hs_ctl);
+ udelay(10);
hs_ctl |= TWL4030_HF_CTL_RAMP_EN;
- twl4030_write(w->codec, e->reg, hs_ctl);
+ twl4030_write(codec, reg, hs_ctl);
+ udelay(40);
hs_ctl |= TWL4030_HF_CTL_LOOP_EN;
- twl4030_write(w->codec, e->reg, hs_ctl);
hs_ctl |= TWL4030_HF_CTL_HB_EN;
- twl4030_write(w->codec, e->reg, hs_ctl);
+ twl4030_write(codec, reg, hs_ctl);
} else {
- hs_ctl &= ~(TWL4030_HF_CTL_RAMP_EN | TWL4030_HF_CTL_LOOP_EN
- | TWL4030_HF_CTL_HB_EN);
- twl4030_write(w->codec, e->reg, hs_ctl);
+ /* HF ramp-down */
+ hs_ctl &= ~TWL4030_HF_CTL_LOOP_EN;
+ hs_ctl &= ~TWL4030_HF_CTL_HB_EN;
+ twl4030_write(codec, reg, hs_ctl);
+ hs_ctl &= ~TWL4030_HF_CTL_RAMP_EN;
+ twl4030_write(codec, reg, hs_ctl);
+ udelay(40);
+ hs_ctl &= ~TWL4030_HF_CTL_REF_EN;
+ twl4030_write(codec, reg, hs_ctl);
}
+}
+static int handsfreelpga_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 1);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ handsfree_ramp(w->codec, TWL4030_REG_HFL_CTL, 0);
+ break;
+ }
return 0;
}
-static int headsetl_event(struct snd_soc_dapm_widget *w,
+static int handsfreerpga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
+{
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 1);
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ handsfree_ramp(w->codec, TWL4030_REG_HFR_CTL, 0);
+ break;
+ }
+ return 0;
+}
+
+static void headset_ramp(struct snd_soc_codec *codec, int ramp)
{
unsigned char hs_gain, hs_pop;
+ struct twl4030_priv *twl4030 = codec->private_data;
+ /* Base values for ramp delay calculation: 2^19 - 2^26 */
+ unsigned int ramp_base[] = {524288, 1048576, 2097152, 4194304,
+ 8388608, 16777216, 33554432, 67108864};
- /* Save the current volume */
- hs_gain = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_GAIN_SET);
- hs_pop = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_POPN_SET);
+ hs_gain = twl4030_read_reg_cache(codec, TWL4030_REG_HS_GAIN_SET);
+ hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
- switch (event) {
- case SND_SOC_DAPM_POST_PMU:
- /* Do the anti-pop/bias ramp enable according to the TRM */
+ if (ramp) {
+ /* Headset ramp-up according to the TRM */
hs_pop |= TWL4030_VMID_EN;
- twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
- /* Is this needed? Can we just use whatever gain here? */
- twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET,
- (hs_gain & (~0x0f)) | 0x0a);
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ twl4030_write(codec, TWL4030_REG_HS_GAIN_SET, hs_gain);
hs_pop |= TWL4030_RAMP_EN;
- twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
-
- /* Restore the original volume */
- twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, hs_gain);
- break;
- case SND_SOC_DAPM_POST_PMD:
- /* Do the anti-pop/bias ramp disable according to the TRM */
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ } else {
+ /* Headset ramp-down _not_ according to
+ * the TRM, but in a way that it is working */
hs_pop &= ~TWL4030_RAMP_EN;
- twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ /* Wait ramp delay time + 1, so the VMID can settle */
+ mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] /
+ twl4030->sysclk) + 1);
/* Bypass the reg_cache to mute the headset */
twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
hs_gain & (~0x0f),
TWL4030_REG_HS_GAIN_SET);
+
hs_pop &= ~TWL4030_VMID_EN;
- twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ twl4030_write(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ }
+}
+
+static int headsetlpga_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct twl4030_priv *twl4030 = w->codec->private_data;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ /* Do the ramp-up only once */
+ if (!twl4030->hsr_enabled)
+ headset_ramp(w->codec, 1);
+
+ twl4030->hsl_enabled = 1;
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ /* Do the ramp-down only if both headsetL/R is disabled */
+ if (!twl4030->hsr_enabled)
+ headset_ramp(w->codec, 0);
+
+ twl4030->hsl_enabled = 0;
+ break;
+ }
+ return 0;
+}
+
+static int headsetrpga_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct twl4030_priv *twl4030 = w->codec->private_data;
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ /* Do the ramp-up only once */
+ if (!twl4030->hsl_enabled)
+ headset_ramp(w->codec, 1);
+
+ twl4030->hsr_enabled = 1;
+ break;
+ case SND_SOC_DAPM_POST_PMD:
+ /* Do the ramp-down only if both headsetL/R is disabled */
+ if (!twl4030->hsl_enabled)
+ headset_ramp(w->codec, 0);
+
+ twl4030->hsr_enabled = 0;
break;
}
return 0;
struct soc_mixer_control *m =
(struct soc_mixer_control *)w->kcontrols->private_value;
struct twl4030_priv *twl4030 = w->codec->private_data;
- unsigned char reg;
+ unsigned char reg, misc;
reg = twl4030_read_reg_cache(w->codec, m->reg);
else
twl4030->bypass_state &=
~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
+ } else if (m->reg == TWL4030_REG_VDL_APGA_CTL) {
+ /* Analog voice bypass */
+ if (reg & (1 << m->shift))
+ twl4030->bypass_state |= (1 << 4);
+ else
+ twl4030->bypass_state &= ~(1 << 4);
+ } else if (m->reg == TWL4030_REG_VSTPGA) {
+ /* Voice digital bypass */
+ if (reg)
+ twl4030->bypass_state |= (1 << 5);
+ else
+ twl4030->bypass_state &= ~(1 << 5);
} else {
/* Digital bypass */
if (reg & (0x7 << m->shift))
- twl4030->bypass_state |= (1 << (m->shift ? 5 : 4));
+ twl4030->bypass_state |= (1 << (m->shift ? 7 : 6));
else
- twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4));
+ twl4030->bypass_state &= ~(1 << (m->shift ? 7 : 6));
}
+ /* Enable master analog loopback mode if any analog switch is enabled*/
+ misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1);
+ if (twl4030->bypass_state & 0x1F)
+ misc |= TWL4030_FMLOOP_EN;
+ else
+ misc &= ~TWL4030_FMLOOP_EN;
+ twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc);
+
if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) {
if (twl4030->bypass_state)
twl4030_codec_mute(w->codec, 0);
return err;
}
+/* Codec operation modes */
+static const char *twl4030_op_modes_texts[] = {
+ "Option 2 (voice/audio)", "Option 1 (audio)"
+};
+
+static const struct soc_enum twl4030_op_modes_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_CODEC_MODE, 0,
+ ARRAY_SIZE(twl4030_op_modes_texts),
+ twl4030_op_modes_texts);
+
+int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct twl4030_priv *twl4030 = codec->private_data;
+ struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+ unsigned short val;
+ unsigned short mask, bitmask;
+
+ if (twl4030->configured) {
+ printk(KERN_ERR "twl4030 operation mode cannot be "
+ "changed on-the-fly\n");
+ return -EBUSY;
+ }
+
+ for (bitmask = 1; bitmask < e->max; bitmask <<= 1)
+ ;
+ if (ucontrol->value.enumerated.item[0] > e->max - 1)
+ return -EINVAL;
+
+ val = ucontrol->value.enumerated.item[0] << e->shift_l;
+ mask = (bitmask - 1) << e->shift_l;
+ if (e->shift_l != e->shift_r) {
+ if (ucontrol->value.enumerated.item[1] > e->max - 1)
+ return -EINVAL;
+ val |= ucontrol->value.enumerated.item[1] << e->shift_r;
+ mask |= (bitmask - 1) << e->shift_r;
+ }
+
+ return snd_soc_update_bits(codec, e->reg, mask, val);
+}
+
/*
* FGAIN volume control:
* from -62 to 0 dB in 1 dB steps (mute instead of -63 dB)
*/
static DECLARE_TLV_DB_SCALE(digital_coarse_tlv, 0, 600, 0);
+/*
+ * Voice Downlink GAIN volume control:
+ * from -37 to 12 dB in 1 dB steps (mute instead of -37 dB)
+ */
+static DECLARE_TLV_DB_SCALE(digital_voice_downlink_tlv, -3700, 100, 1);
+
/*
* Analog playback gain
* -24 dB to 12 dB in 2 dB steps
ARRAY_SIZE(twl4030_rampdelay_texts),
twl4030_rampdelay_texts);
+/* Vibra H-bridge direction mode */
+static const char *twl4030_vibradirmode_texts[] = {
+ "Vibra H-bridge direction", "Audio data MSB",
+};
+
+static const struct soc_enum twl4030_vibradirmode_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 5,
+ ARRAY_SIZE(twl4030_vibradirmode_texts),
+ twl4030_vibradirmode_texts);
+
+/* Vibra H-bridge direction */
+static const char *twl4030_vibradir_texts[] = {
+ "Positive polarity", "Negative polarity",
+};
+
+static const struct soc_enum twl4030_vibradir_enum =
+ SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 1,
+ ARRAY_SIZE(twl4030_vibradir_texts),
+ twl4030_vibradir_texts);
+
static const struct snd_kcontrol_new twl4030_snd_controls[] = {
+ /* Codec operation mode control */
+ SOC_ENUM_EXT("Codec Operation Mode", twl4030_op_modes_enum,
+ snd_soc_get_enum_double,
+ snd_soc_put_twl4030_opmode_enum_double),
+
/* Common playback gain controls */
SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume",
TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA,
TWL4030_REG_ARXL2_APGA_CTL, TWL4030_REG_ARXR2_APGA_CTL,
1, 1, 0),
+ /* Common voice downlink gain controls */
+ SOC_SINGLE_TLV("DAC Voice Digital Downlink Volume",
+ TWL4030_REG_VRXPGA, 0, 0x31, 0, digital_voice_downlink_tlv),
+
+ SOC_SINGLE_TLV("DAC Voice Analog Downlink Volume",
+ TWL4030_REG_VDL_APGA_CTL, 3, 0x12, 1, analog_tlv),
+
+ SOC_SINGLE("DAC Voice Analog Downlink Switch",
+ TWL4030_REG_VDL_APGA_CTL, 1, 1, 0),
+
/* Separate output gain controls */
SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume",
TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL,
0, 3, 5, 0, input_gain_tlv),
SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum),
+
+ SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum),
+ SOC_ENUM("Vibra H-bridge direction", twl4030_vibradir_enum),
};
static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
SND_SOC_DAPM_OUTPUT("CARKITR"),
SND_SOC_DAPM_OUTPUT("HFL"),
SND_SOC_DAPM_OUTPUT("HFR"),
+ SND_SOC_DAPM_OUTPUT("VIBRA"),
/* DACs */
- SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback",
+ SND_SOC_DAPM_DAC("DAC Right1", "Right Front HiFi Playback",
SND_SOC_NOPM, 0, 0),
- SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback",
+ SND_SOC_DAPM_DAC("DAC Left1", "Left Front HiFi Playback",
SND_SOC_NOPM, 0, 0),
- SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback",
+ SND_SOC_DAPM_DAC("DAC Right2", "Right Rear HiFi Playback",
SND_SOC_NOPM, 0, 0),
- SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback",
+ SND_SOC_DAPM_DAC("DAC Left2", "Left Rear HiFi Playback",
+ SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback",
SND_SOC_NOPM, 0, 0),
-
- /* Analog PGAs */
- SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL,
- 0, 0, NULL, 0),
- SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL,
- 0, 0, NULL, 0),
- SND_SOC_DAPM_PGA("ARXR2_APGA", TWL4030_REG_ARXR2_APGA_CTL,
- 0, 0, NULL, 0),
- SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL,
- 0, 0, NULL, 0),
/* Analog bypasses */
SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0,
SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_abypassl2_control,
bypass_event, SND_SOC_DAPM_POST_REG),
+ SND_SOC_DAPM_SWITCH_E("Voice Analog Loopback", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_abypassv_control,
+ bypass_event, SND_SOC_DAPM_POST_REG),
/* Digital bypasses */
SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0,
SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0,
&twl4030_dapm_dbypassr_control, bypass_event,
SND_SOC_DAPM_POST_REG),
+ SND_SOC_DAPM_SWITCH_E("Voice Digital Loopback", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_dbypassv_control, bypass_event,
+ SND_SOC_DAPM_POST_REG),
- SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
- 0, 0, NULL, 0),
- SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
- 1, 0, NULL, 0),
- SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
- 2, 0, NULL, 0),
- SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
- 3, 0, NULL, 0),
-
- /* Output MUX controls */
+ /* Digital mixers, power control for the physical DACs */
+ SND_SOC_DAPM_MIXER("Digital R1 Playback Mixer",
+ TWL4030_REG_AVDAC_CTL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Digital L1 Playback Mixer",
+ TWL4030_REG_AVDAC_CTL, 1, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Digital R2 Playback Mixer",
+ TWL4030_REG_AVDAC_CTL, 2, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Digital L2 Playback Mixer",
+ TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Digital Voice Playback Mixer",
+ TWL4030_REG_AVDAC_CTL, 4, 0, NULL, 0),
+
+ /* Analog mixers, power control for the physical PGAs */
+ SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer",
+ TWL4030_REG_ARXR1_APGA_CTL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer",
+ TWL4030_REG_ARXL1_APGA_CTL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer",
+ TWL4030_REG_ARXR2_APGA_CTL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer",
+ TWL4030_REG_ARXL2_APGA_CTL, 0, 0, NULL, 0),
+ SND_SOC_DAPM_MIXER("Analog Voice Playback Mixer",
+ TWL4030_REG_VDL_APGA_CTL, 0, 0, NULL, 0),
+
+ /* Output MIXER controls */
/* Earpiece */
- SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0,
- &twl4030_dapm_earpiece_control),
+ SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_earpiece_controls[0],
+ ARRAY_SIZE(twl4030_dapm_earpiece_controls)),
/* PreDrivL/R */
- SND_SOC_DAPM_VALUE_MUX("PredriveL Mux", SND_SOC_NOPM, 0, 0,
- &twl4030_dapm_predrivel_control),
- SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0,
- &twl4030_dapm_predriver_control),
+ SND_SOC_DAPM_MIXER("PredriveL Mixer", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_predrivel_controls[0],
+ ARRAY_SIZE(twl4030_dapm_predrivel_controls)),
+ SND_SOC_DAPM_MIXER("PredriveR Mixer", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_predriver_controls[0],
+ ARRAY_SIZE(twl4030_dapm_predriver_controls)),
/* HeadsetL/R */
- SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
- &twl4030_dapm_hsol_control, headsetl_event,
- SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
- SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0,
- &twl4030_dapm_hsor_control),
+ SND_SOC_DAPM_MIXER("HeadsetL Mixer", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_hsol_controls[0],
+ ARRAY_SIZE(twl4030_dapm_hsol_controls)),
+ SND_SOC_DAPM_PGA_E("HeadsetL PGA", SND_SOC_NOPM,
+ 0, 0, NULL, 0, headsetlpga_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MIXER("HeadsetR Mixer", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_hsor_controls[0],
+ ARRAY_SIZE(twl4030_dapm_hsor_controls)),
+ SND_SOC_DAPM_PGA_E("HeadsetR PGA", SND_SOC_NOPM,
+ 0, 0, NULL, 0, headsetrpga_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
/* CarkitL/R */
- SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0,
- &twl4030_dapm_carkitl_control),
- SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0,
- &twl4030_dapm_carkitr_control),
+ SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_carkitl_controls[0],
+ ARRAY_SIZE(twl4030_dapm_carkitl_controls)),
+ SND_SOC_DAPM_MIXER("CarkitR Mixer", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_carkitr_controls[0],
+ ARRAY_SIZE(twl4030_dapm_carkitr_controls)),
+
+ /* Output MUX controls */
/* HandsfreeL/R */
- SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0,
- &twl4030_dapm_handsfreel_control, handsfree_event,
- SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
- SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0,
- &twl4030_dapm_handsfreer_control, handsfree_event,
- SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX("HandsfreeL Mux", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_handsfreel_control),
+ SND_SOC_DAPM_SWITCH("HandsfreeL Switch", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_handsfreelmute_control),
+ SND_SOC_DAPM_PGA_E("HandsfreeL PGA", SND_SOC_NOPM,
+ 0, 0, NULL, 0, handsfreelpga_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_MUX("HandsfreeR Mux", SND_SOC_NOPM, 5, 0,
+ &twl4030_dapm_handsfreer_control),
+ SND_SOC_DAPM_SWITCH("HandsfreeR Switch", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_handsfreermute_control),
+ SND_SOC_DAPM_PGA_E("HandsfreeR PGA", SND_SOC_NOPM,
+ 0, 0, NULL, 0, handsfreerpga_event,
+ SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
+ /* Vibra */
+ SND_SOC_DAPM_MUX("Vibra Mux", TWL4030_REG_VIBRA_CTL, 0, 0,
+ &twl4030_dapm_vibra_control),
+ SND_SOC_DAPM_MUX("Vibra Route", SND_SOC_NOPM, 0, 0,
+ &twl4030_dapm_vibrapath_control),
/* Introducing four virtual ADC, since TWL4030 have four channel for
capture */
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
SND_SOC_DAPM_POST_REG),
- /* Analog input muxes with switch for the capture amplifiers */
- SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route",
- TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control),
- SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route",
- TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control),
+ /* Analog input mixers for the capture amplifiers */
+ SND_SOC_DAPM_MIXER("Analog Left Capture Route",
+ TWL4030_REG_ANAMICL, 4, 0,
+ &twl4030_dapm_analoglmic_controls[0],
+ ARRAY_SIZE(twl4030_dapm_analoglmic_controls)),
+ SND_SOC_DAPM_MIXER("Analog Right Capture Route",
+ TWL4030_REG_ANAMICR, 4, 0,
+ &twl4030_dapm_analogrmic_controls[0],
+ ARRAY_SIZE(twl4030_dapm_analogrmic_controls)),
SND_SOC_DAPM_PGA("ADC Physical Left",
TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0),
};
static const struct snd_soc_dapm_route intercon[] = {
- {"Analog L1 Playback Mixer", NULL, "DAC Left1"},
- {"Analog R1 Playback Mixer", NULL, "DAC Right1"},
- {"Analog L2 Playback Mixer", NULL, "DAC Left2"},
- {"Analog R2 Playback Mixer", NULL, "DAC Right2"},
-
- {"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"},
- {"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"},
- {"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"},
- {"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"},
+ {"Digital L1 Playback Mixer", NULL, "DAC Left1"},
+ {"Digital R1 Playback Mixer", NULL, "DAC Right1"},
+ {"Digital L2 Playback Mixer", NULL, "DAC Left2"},
+ {"Digital R2 Playback Mixer", NULL, "DAC Right2"},
+ {"Digital Voice Playback Mixer", NULL, "DAC Voice"},
+
+ {"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"},
+ {"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"},
+ {"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"},
+ {"Analog R2 Playback Mixer", NULL, "Digital R2 Playback Mixer"},
+ {"Analog Voice Playback Mixer", NULL, "Digital Voice Playback Mixer"},
/* Internal playback routings */
/* Earpiece */
- {"Earpiece Mux", "DACL1", "ARXL1_APGA"},
- {"Earpiece Mux", "DACL2", "ARXL2_APGA"},
- {"Earpiece Mux", "DACR1", "ARXR1_APGA"},
+ {"Earpiece Mixer", "Voice", "Analog Voice Playback Mixer"},
+ {"Earpiece Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+ {"Earpiece Mixer", "AudioL2", "Analog L2 Playback Mixer"},
+ {"Earpiece Mixer", "AudioR1", "Analog R1 Playback Mixer"},
/* PreDrivL */
- {"PredriveL Mux", "DACL1", "ARXL1_APGA"},
- {"PredriveL Mux", "DACL2", "ARXL2_APGA"},
- {"PredriveL Mux", "DACR2", "ARXR2_APGA"},
+ {"PredriveL Mixer", "Voice", "Analog Voice Playback Mixer"},
+ {"PredriveL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+ {"PredriveL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
+ {"PredriveL Mixer", "AudioR2", "Analog R2 Playback Mixer"},
/* PreDrivR */
- {"PredriveR Mux", "DACR1", "ARXR1_APGA"},
- {"PredriveR Mux", "DACR2", "ARXR2_APGA"},
- {"PredriveR Mux", "DACL2", "ARXL2_APGA"},
+ {"PredriveR Mixer", "Voice", "Analog Voice Playback Mixer"},
+ {"PredriveR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
+ {"PredriveR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
+ {"PredriveR Mixer", "AudioL2", "Analog L2 Playback Mixer"},
/* HeadsetL */
- {"HeadsetL Mux", "DACL1", "ARXL1_APGA"},
- {"HeadsetL Mux", "DACL2", "ARXL2_APGA"},
+ {"HeadsetL Mixer", "Voice", "Analog Voice Playback Mixer"},
+ {"HeadsetL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+ {"HeadsetL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
+ {"HeadsetL PGA", NULL, "HeadsetL Mixer"},
/* HeadsetR */
- {"HeadsetR Mux", "DACR1", "ARXR1_APGA"},
- {"HeadsetR Mux", "DACR2", "ARXR2_APGA"},
+ {"HeadsetR Mixer", "Voice", "Analog Voice Playback Mixer"},
+ {"HeadsetR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
+ {"HeadsetR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
+ {"HeadsetR PGA", NULL, "HeadsetR Mixer"},
/* CarkitL */
- {"CarkitL Mux", "DACL1", "ARXL1_APGA"},
- {"CarkitL Mux", "DACL2", "ARXL2_APGA"},
+ {"CarkitL Mixer", "Voice", "Analog Voice Playback Mixer"},
+ {"CarkitL Mixer", "AudioL1", "Analog L1 Playback Mixer"},
+ {"CarkitL Mixer", "AudioL2", "Analog L2 Playback Mixer"},
/* CarkitR */
- {"CarkitR Mux", "DACR1", "ARXR1_APGA"},
- {"CarkitR Mux", "DACR2", "ARXR2_APGA"},
+ {"CarkitR Mixer", "Voice", "Analog Voice Playback Mixer"},
+ {"CarkitR Mixer", "AudioR1", "Analog R1 Playback Mixer"},
+ {"CarkitR Mixer", "AudioR2", "Analog R2 Playback Mixer"},
/* HandsfreeL */
- {"HandsfreeL Mux", "DACL1", "ARXL1_APGA"},
- {"HandsfreeL Mux", "DACL2", "ARXL2_APGA"},
- {"HandsfreeL Mux", "DACR2", "ARXR2_APGA"},
+ {"HandsfreeL Mux", "Voice", "Analog Voice Playback Mixer"},
+ {"HandsfreeL Mux", "AudioL1", "Analog L1 Playback Mixer"},
+ {"HandsfreeL Mux", "AudioL2", "Analog L2 Playback Mixer"},
+ {"HandsfreeL Mux", "AudioR2", "Analog R2 Playback Mixer"},
+ {"HandsfreeL Switch", "Switch", "HandsfreeL Mux"},
+ {"HandsfreeL PGA", NULL, "HandsfreeL Switch"},
/* HandsfreeR */
- {"HandsfreeR Mux", "DACR1", "ARXR1_APGA"},
- {"HandsfreeR Mux", "DACR2", "ARXR2_APGA"},
- {"HandsfreeR Mux", "DACL2", "ARXL2_APGA"},
+ {"HandsfreeR Mux", "Voice", "Analog Voice Playback Mixer"},
+ {"HandsfreeR Mux", "AudioR1", "Analog R1 Playback Mixer"},
+ {"HandsfreeR Mux", "AudioR2", "Analog R2 Playback Mixer"},
+ {"HandsfreeR Mux", "AudioL2", "Analog L2 Playback Mixer"},
+ {"HandsfreeR Switch", "Switch", "HandsfreeR Mux"},
+ {"HandsfreeR PGA", NULL, "HandsfreeR Switch"},
+ /* Vibra */
+ {"Vibra Mux", "AudioL1", "DAC Left1"},
+ {"Vibra Mux", "AudioR1", "DAC Right1"},
+ {"Vibra Mux", "AudioL2", "DAC Left2"},
+ {"Vibra Mux", "AudioR2", "DAC Right2"},
/* outputs */
- {"OUTL", NULL, "ARXL2_APGA"},
- {"OUTR", NULL, "ARXR2_APGA"},
- {"EARPIECE", NULL, "Earpiece Mux"},
- {"PREDRIVEL", NULL, "PredriveL Mux"},
- {"PREDRIVER", NULL, "PredriveR Mux"},
- {"HSOL", NULL, "HeadsetL Mux"},
- {"HSOR", NULL, "HeadsetR Mux"},
- {"CARKITL", NULL, "CarkitL Mux"},
- {"CARKITR", NULL, "CarkitR Mux"},
- {"HFL", NULL, "HandsfreeL Mux"},
- {"HFR", NULL, "HandsfreeR Mux"},
+ {"OUTL", NULL, "Analog L2 Playback Mixer"},
+ {"OUTR", NULL, "Analog R2 Playback Mixer"},
+ {"EARPIECE", NULL, "Earpiece Mixer"},
+ {"PREDRIVEL", NULL, "PredriveL Mixer"},
+ {"PREDRIVER", NULL, "PredriveR Mixer"},
+ {"HSOL", NULL, "HeadsetL PGA"},
+ {"HSOR", NULL, "HeadsetR PGA"},
+ {"CARKITL", NULL, "CarkitL Mixer"},
+ {"CARKITR", NULL, "CarkitR Mixer"},
+ {"HFL", NULL, "HandsfreeL PGA"},
+ {"HFR", NULL, "HandsfreeR PGA"},
+ {"Vibra Route", "Audio", "Vibra Mux"},
+ {"VIBRA", NULL, "Vibra Route"},
/* Capture path */
{"Analog Left Capture Route", "Main mic", "MAINMIC"},
{"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"},
{"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"},
{"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"},
+ {"Voice Analog Loopback", "Switch", "Analog Left Capture Route"},
{"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"},
{"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"},
{"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"},
{"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"},
+ {"Analog Voice Playback Mixer", NULL, "Voice Analog Loopback"},
/* Digital bypass routes */
{"Right Digital Loopback", "Volume", "TX1 Capture Route"},
{"Left Digital Loopback", "Volume", "TX1 Capture Route"},
+ {"Voice Digital Loopback", "Volume", "TX2 Capture Route"},
- {"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"},
- {"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"},
+ {"Digital R2 Playback Mixer", NULL, "Right Digital Loopback"},
+ {"Digital L2 Playback Mixer", NULL, "Left Digital Loopback"},
+ {"Digital Voice Playback Mixer", NULL, "Voice Digital Loopback"},
};
return 0;
}
+static void twl4030_constraints(struct twl4030_priv *twl4030,
+ struct snd_pcm_substream *mst_substream)
+{
+ struct snd_pcm_substream *slv_substream;
+
+ /* Pick the stream, which need to be constrained */
+ if (mst_substream == twl4030->master_substream)
+ slv_substream = twl4030->slave_substream;
+ else if (mst_substream == twl4030->slave_substream)
+ slv_substream = twl4030->master_substream;
+ else /* This should not happen.. */
+ return;
+
+ /* Set the constraints according to the already configured stream */
+ snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ twl4030->rate,
+ twl4030->rate);
+
+ snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+ twl4030->sample_bits,
+ twl4030->sample_bits);
+
+ snd_pcm_hw_constraint_minmax(slv_substream->runtime,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ twl4030->channels,
+ twl4030->channels);
+}
+
+/* In case of 4 channel mode, the RX1 L/R for playback and the TX2 L/R for
+ * capture has to be enabled/disabled. */
+static void twl4030_tdm_enable(struct snd_soc_codec *codec, int direction,
+ int enable)
+{
+ u8 reg, mask;
+
+ reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION);
+
+ if (direction == SNDRV_PCM_STREAM_PLAYBACK)
+ mask = TWL4030_ARXL1_VRX_EN | TWL4030_ARXR1_EN;
+ else
+ mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN;
+
+ if (enable)
+ reg |= mask;
+ else
+ reg &= ~mask;
+
+ twl4030_write(codec, TWL4030_REG_OPTION, reg);
+}
+
static int twl4030_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = socdev->card->codec;
struct twl4030_priv *twl4030 = codec->private_data;
- /* If we already have a playback or capture going then constrain
- * this substream to match it.
- */
if (twl4030->master_substream) {
- struct snd_pcm_runtime *master_runtime;
- master_runtime = twl4030->master_substream->runtime;
-
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_RATE,
- master_runtime->rate,
- master_runtime->rate);
-
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
- master_runtime->sample_bits,
- master_runtime->sample_bits);
-
twl4030->slave_substream = substream;
- } else
+ /* The DAI has one configuration for playback and capture, so
+ * if the DAI has been already configured then constrain this
+ * substream to match it. */
+ if (twl4030->configured)
+ twl4030_constraints(twl4030, twl4030->master_substream);
+ } else {
+ if (!(twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) &
+ TWL4030_OPTION_1)) {
+ /* In option2 4 channel is not supported, set the
+ * constraint for the first stream for channels, the
+ * second stream will 'inherit' this cosntraint */
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ 2, 2);
+ }
twl4030->master_substream = substream;
+ }
return 0;
}
twl4030->master_substream = twl4030->slave_substream;
twl4030->slave_substream = NULL;
+
+ /* If all streams are closed, or the remaining stream has not yet
+ * been configured than set the DAI as not configured. */
+ if (!twl4030->master_substream)
+ twl4030->configured = 0;
+ else if (!twl4030->master_substream->runtime->channels)
+ twl4030->configured = 0;
+
+ /* If the closing substream had 4 channel, do the necessary cleanup */
+ if (substream->runtime->channels == 4)
+ twl4030_tdm_enable(codec, substream->stream, 0);
}
static int twl4030_hw_params(struct snd_pcm_substream *substream,
struct twl4030_priv *twl4030 = codec->private_data;
u8 mode, old_mode, format, old_format;
- if (substream == twl4030->slave_substream)
- /* Ignoring hw_params for slave substream */
+ /* If the substream has 4 channel, do the necessary setup */
+ if (params_channels(params) == 4) {
+ u8 format, mode;
+
+ format = twl4030_read_reg_cache(codec, TWL4030_REG_AUDIO_IF);
+ mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
+
+ /* Safety check: are we in the correct operating mode and
+ * the interface is in TDM mode? */
+ if ((mode & TWL4030_OPTION_1) &&
+ ((format & TWL4030_AIF_FORMAT) == TWL4030_AIF_FORMAT_TDM))
+ twl4030_tdm_enable(codec, substream->stream, 1);
+ else
+ return -EINVAL;
+ }
+
+ if (twl4030->configured)
+ /* Ignoring hw_params for already configured DAI */
return 0;
/* bit rate */
/* set CODECPDZ afterwards */
twl4030_codec_enable(codec, 1);
}
+
+ /* Store the important parameters for the DAI configuration and set
+ * the DAI as configured */
+ twl4030->configured = 1;
+ twl4030->rate = params_rate(params);
+ twl4030->sample_bits = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min;
+ twl4030->channels = params_channels(params);
+
+ /* If both playback and capture streams are open, and one of them
+ * is setting the hw parameters right now (since we are here), set
+ * constraints to the other stream to match the current one. */
+ if (twl4030->slave_substream)
+ twl4030_constraints(twl4030, substream);
+
return 0;
}
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
+ struct twl4030_priv *twl4030 = codec->private_data;
u8 infreq;
switch (freq) {
case 19200000:
infreq = TWL4030_APLL_INFREQ_19200KHZ;
+ twl4030->sysclk = 19200;
break;
case 26000000:
infreq = TWL4030_APLL_INFREQ_26000KHZ;
+ twl4030->sysclk = 26000;
break;
case 38400000:
infreq = TWL4030_APLL_INFREQ_38400KHZ;
+ twl4030->sysclk = 38400;
break;
default:
printk(KERN_ERR "TWL4030 set sysclk: unknown rate %d\n",
case SND_SOC_DAIFMT_I2S:
format |= TWL4030_AIF_FORMAT_CODEC;
break;
+ case SND_SOC_DAIFMT_DSP_A:
+ format |= TWL4030_AIF_FORMAT_TDM;
+ break;
default:
return -EINVAL;
}
return 0;
}
+/* In case of voice mode, the RX1 L(VRX) for downlink and the TX2 L/R
+ * (VTXL, VTXR) for uplink has to be enabled/disabled. */
+static void twl4030_voice_enable(struct snd_soc_codec *codec, int direction,
+ int enable)
+{
+ u8 reg, mask;
+
+ reg = twl4030_read_reg_cache(codec, TWL4030_REG_OPTION);
+
+ if (direction == SNDRV_PCM_STREAM_PLAYBACK)
+ mask = TWL4030_ARXL1_VRX_EN;
+ else
+ mask = TWL4030_ATXL2_VTXL_EN | TWL4030_ATXR2_VTXR_EN;
+
+ if (enable)
+ reg |= mask;
+ else
+ reg &= ~mask;
+
+ twl4030_write(codec, TWL4030_REG_OPTION, reg);
+}
+
+static int twl4030_voice_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ u8 infreq;
+ u8 mode;
+
+ /* If the system master clock is not 26MHz, the voice PCM interface is
+ * not avilable.
+ */
+ infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL)
+ & TWL4030_APLL_INFREQ;
+
+ if (infreq != TWL4030_APLL_INFREQ_26000KHZ) {
+ printk(KERN_ERR "TWL4030 voice startup: "
+ "MCLK is not 26MHz, call set_sysclk() on init\n");
+ return -EINVAL;
+ }
+
+ /* If the codec mode is not option2, the voice PCM interface is not
+ * avilable.
+ */
+ mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
+ & TWL4030_OPT_MODE;
+
+ if (mode != TWL4030_OPTION_2) {
+ printk(KERN_ERR "TWL4030 voice startup: "
+ "the codec mode is not option2\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static void twl4030_voice_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ /* Enable voice digital filters */
+ twl4030_voice_enable(codec, substream->stream, 0);
+}
+
+static int twl4030_voice_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ u8 old_mode, mode;
+
+ /* Enable voice digital filters */
+ twl4030_voice_enable(codec, substream->stream, 1);
+
+ /* bit rate */
+ old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE)
+ & ~(TWL4030_CODECPDZ);
+ mode = old_mode;
+
+ switch (params_rate(params)) {
+ case 8000:
+ mode &= ~(TWL4030_SEL_16K);
+ break;
+ case 16000:
+ mode |= TWL4030_SEL_16K;
+ break;
+ default:
+ printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n",
+ params_rate(params));
+ return -EINVAL;
+ }
+
+ if (mode != old_mode) {
+ /* change rate and set CODECPDZ */
+ twl4030_codec_enable(codec, 0);
+ twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+ twl4030_codec_enable(codec, 1);
+ }
+
+ return 0;
+}
+
+static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 infreq;
+
+ switch (freq) {
+ case 26000000:
+ infreq = TWL4030_APLL_INFREQ_26000KHZ;
+ break;
+ default:
+ printk(KERN_ERR "TWL4030 voice set sysclk: unknown rate %d\n",
+ freq);
+ return -EINVAL;
+ }
+
+ infreq |= TWL4030_APLL_EN;
+ twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq);
+
+ return 0;
+}
+
+static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u8 old_format, format;
+
+ /* get format */
+ old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF);
+ format = old_format;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBS_CFM:
+ format &= ~(TWL4030_VIF_SLAVE_EN);
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ format |= TWL4030_VIF_SLAVE_EN;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_IB_NF:
+ format &= ~(TWL4030_VIF_FORMAT);
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ format |= TWL4030_VIF_FORMAT;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ if (format != old_format) {
+ /* change format and set CODECPDZ */
+ twl4030_codec_enable(codec, 0);
+ twl4030_write(codec, TWL4030_REG_VOICE_IF, format);
+ twl4030_codec_enable(codec, 1);
+ }
+
+ return 0;
+}
+
#define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000)
#define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
.set_fmt = twl4030_set_dai_fmt,
};
-struct snd_soc_dai twl4030_dai = {
+static struct snd_soc_dai_ops twl4030_dai_voice_ops = {
+ .startup = twl4030_voice_startup,
+ .shutdown = twl4030_voice_shutdown,
+ .hw_params = twl4030_voice_hw_params,
+ .set_sysclk = twl4030_voice_set_dai_sysclk,
+ .set_fmt = twl4030_voice_set_dai_fmt,
+};
+
+struct snd_soc_dai twl4030_dai[] = {
+{
.name = "twl4030",
.playback = {
- .stream_name = "Playback",
+ .stream_name = "HiFi Playback",
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 4,
.rates = TWL4030_RATES | SNDRV_PCM_RATE_96000,
.formats = TWL4030_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
- .channels_max = 2,
+ .channels_max = 4,
.rates = TWL4030_RATES,
.formats = TWL4030_FORMATS,},
.ops = &twl4030_dai_ops,
+},
+{
+ .name = "twl4030 Voice",
+ .playback = {
+ .stream_name = "Voice Playback",
+ .channels_min = 1,
+ .channels_max = 1,
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = &twl4030_dai_voice_ops,
+},
};
EXPORT_SYMBOL_GPL(twl4030_dai);
static int twl4030_init(struct snd_soc_device *socdev)
{
struct snd_soc_codec *codec = socdev->card->codec;
+ struct twl4030_setup_data *setup = socdev->codec_data;
+ struct twl4030_priv *twl4030 = codec->private_data;
int ret = 0;
printk(KERN_INFO "TWL4030 Audio Codec init \n");
codec->read = twl4030_read_reg_cache;
codec->write = twl4030_write;
codec->set_bias_level = twl4030_set_bias_level;
- codec->dai = &twl4030_dai;
- codec->num_dai = 1;
+ codec->dai = twl4030_dai;
+ codec->num_dai = ARRAY_SIZE(twl4030_dai),
codec->reg_cache_size = sizeof(twl4030_reg);
codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg),
GFP_KERNEL);
if (codec->reg_cache == NULL)
return -ENOMEM;
+ /* Configuration for headset ramp delay from setup data */
+ if (setup) {
+ unsigned char hs_pop;
+
+ if (setup->sysclk)
+ twl4030->sysclk = setup->sysclk;
+ else
+ twl4030->sysclk = 26000;
+
+ hs_pop = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
+ hs_pop &= ~TWL4030_RAMP_DELAY;
+ hs_pop |= (setup->ramp_delay_value << 2);
+ twl4030_write_reg_cache(codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+ } else {
+ twl4030->sysclk = 26000;
+ }
+
/* register pcms */
ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
if (ret < 0) {
static int __init twl4030_modinit(void)
{
- return snd_soc_register_dai(&twl4030_dai);
+ return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai));
}
module_init(twl4030_modinit);
static void __exit twl4030_exit(void)
{
- snd_soc_unregister_dai(&twl4030_dai);
+ snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai));
}
module_exit(twl4030_exit);
#define TWL4030_REG_VIBRA_PWM_SET 0x47
#define TWL4030_REG_ANAMIC_GAIN 0x48
#define TWL4030_REG_MISC_SET_2 0x49
+#define TWL4030_REG_SW_SHADOW 0x4A
-#define TWL4030_CACHEREGNUM (TWL4030_REG_MISC_SET_2 + 1)
+#define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1)
/* Bitfield Definitions */
#define TWL4030_APLL_RATE_44100 0x90
#define TWL4030_APLL_RATE_48000 0xA0
#define TWL4030_APLL_RATE_96000 0xE0
-#define TWL4030_SEL_16K 0x04
+#define TWL4030_SEL_16K 0x08
#define TWL4030_CODECPDZ 0x02
#define TWL4030_OPT_MODE 0x01
+#define TWL4030_OPTION_1 (1 << 0)
+#define TWL4030_OPTION_2 (0 << 0)
+
+/* TWL4030_OPTION (0x02) Fields */
+
+#define TWL4030_ATXL1_EN (1 << 0)
+#define TWL4030_ATXR1_EN (1 << 1)
+#define TWL4030_ATXL2_VTXL_EN (1 << 2)
+#define TWL4030_ATXR2_VTXR_EN (1 << 3)
+#define TWL4030_ARXL1_VRX_EN (1 << 4)
+#define TWL4030_ARXR1_EN (1 << 5)
+#define TWL4030_ARXL2_EN (1 << 6)
+#define TWL4030_ARXR2_EN (1 << 7)
/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */
#define TWL4030_CLK256FS_EN 0x02
#define TWL4030_AIF_EN 0x01
+/* VOICE_IF (0x0F) Fields */
+
+#define TWL4030_VIF_SLAVE_EN 0x80
+#define TWL4030_VIF_DIN_EN 0x40
+#define TWL4030_VIF_DOUT_EN 0x20
+#define TWL4030_VIF_SWAP 0x10
+#define TWL4030_VIF_FORMAT 0x08
+#define TWL4030_VIF_TRI_EN 0x04
+#define TWL4030_VIF_SUB_EN 0x02
+#define TWL4030_VIF_EN 0x01
+
/* EAR_CTL (0x21) */
#define TWL4030_EAR_GAIN 0x30
#define TWL4030_SMOOTH_ANAVOL_EN 0x02
#define TWL4030_DIGMIC_LR_SWAP_EN 0x01
-extern struct snd_soc_dai twl4030_dai;
+/* TWL4030_REG_SW_SHADOW (0x4A) Fields */
+#define TWL4030_HFL_EN 0x01
+#define TWL4030_HFR_EN 0x02
+
+#define TWL4030_DAI_HIFI 0
+#define TWL4030_DAI_VOICE 1
+
+extern struct snd_soc_dai twl4030_dai[2];
extern struct snd_soc_codec_device soc_codec_dev_twl4030;
+struct twl4030_setup_data {
+ unsigned int ramp_delay_value;
+ unsigned int sysclk;
+};
+
#endif /* End of __TWL4030_AUDIO_H__ */
pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
if (reg >= UDA134X_REGS_NUM) {
- printk(KERN_ERR "%s unkown register: reg: %d",
+ printk(KERN_ERR "%s unkown register: reg: %u",
__func__, reg);
return -EINVAL;
}
struct snd_soc_codec *codec = codec_dai->codec;
struct uda134x_priv *uda134x = codec->private_data;
- pr_debug("%s clk_id: %d, freq: %d, dir: %d\n", __func__,
+ pr_debug("%s clk_id: %d, freq: %u, dir: %d\n", __func__,
clk_id, freq, dir);
/* Anything between 256fs*8Khz and 512fs*48Khz should be acceptable
if (ret < 0)
return ret;
dev_dbg(wm8350->dev,
- "FLL in %d FLL out %d N 0x%x K 0x%x div %d ratio %d",
+ "FLL in %u FLL out %u N 0x%x K 0x%x div %d ratio %d",
freq_in, freq_out, fll_div.n, fll_div.k, fll_div.div,
fll_div.ratio);
#define _WM8350_H
#include <sound/soc.h>
+#include <linux/mfd/wm8350/audio.h>
extern struct snd_soc_dai wm8350_dai;
extern struct snd_soc_codec_device soc_codec_dev_wm8350;
factors->outdiv *= 2;
if (factors->outdiv > 32) {
dev_err(wm8400->wm8400->dev,
- "Unsupported FLL output frequency %dHz\n",
+ "Unsupported FLL output frequency %uHz\n",
Fout);
return -EINVAL;
}
factors->k = K / 10;
dev_dbg(wm8400->wm8400->dev,
- "FLL: Fref=%d Fout=%d N=%x K=%x, FRATIO=%x OUTDIV=%x\n",
+ "FLL: Fref=%u Fout=%u N=%x K=%x, FRATIO=%x OUTDIV=%x\n",
Fref, Fout,
factors->n, factors->k, factors->fratio, factors->outdiv);
codec = &priv->codec;
codec->private_data = priv;
- codec->control_data = dev->dev.driver_data;
- priv->wm8400 = dev->dev.driver_data;
+ codec->control_data = dev_get_drvdata(&dev->dev);
+ priv->wm8400 = dev_get_drvdata(&dev->dev);
ret = regulator_bulk_get(priv->wm8400->dev,
ARRAY_SIZE(power), &power[0]);
if ((Ndiv < 6) || (Ndiv > 12))
printk(KERN_WARNING
- "WM8510 N value %d outwith recommended range!d\n",
+ "WM8510 N value %u outwith recommended range!d\n",
Ndiv);
pll_div.n = Ndiv;
unsigned int K, Ndiv, Nmod;
int i;
- pr_debug("wm8580: PLL %dHz->%dHz\n", source, target);
+ pr_debug("wm8580: PLL %uHz->%uHz\n", source, target);
/* Scale the output frequency up; the PLL should run in the
* region of 90-100MHz.
if ((Ndiv < 5) || (Ndiv > 13)) {
printk(KERN_ERR
- "WM8580 N=%d outside supported range\n", Ndiv);
+ "WM8580 N=%u outside supported range\n", Ndiv);
return -EINVAL;
}
codec->hw_write = (hw_write_t)wm8731_spi_write;
codec->dev = &spi->dev;
- spi->dev.driver_data = wm8731;
+ dev_set_drvdata(&spi->dev, wm8731);
return wm8731_register(wm8731);
}
static int __devexit wm8731_spi_remove(struct spi_device *spi)
{
- struct wm8731_priv *wm8731 = spi->dev.driver_data;
+ struct wm8731_priv *wm8731 = dev_get_drvdata(&spi->dev);
wm8731_unregister(wm8731);
if ((Ndiv < 6) || (Ndiv > 12))
printk(KERN_WARNING
- "wm8753: unsupported N = %d\n", Ndiv);
+ "wm8753: unsupported N = %u\n", Ndiv);
pll_div->n = Ndiv;
Nmod = target % source;
codec->hw_write = (hw_write_t)wm8753_spi_write;
codec->dev = &spi->dev;
- spi->dev.driver_data = wm8753;
+ dev_set_drvdata(&spi->dev, wm8753);
return wm8753_register(wm8753);
}
static int __devexit wm8753_spi_remove(struct spi_device *spi)
{
- struct wm8753_priv *wm8753 = spi->dev.driver_data;
+ struct wm8753_priv *wm8753 = dev_get_drvdata(&spi->dev);
wm8753_unregister(wm8753);
return 0;
}
}
if (target > 100000000)
- printk(KERN_WARNING "wm8900: FLL rate %d out of range, Fref=%d"
- " Fout=%d\n", target, Fref, Fout);
+ printk(KERN_WARNING "wm8900: FLL rate %u out of range, Fref=%u"
+ " Fout=%u\n", target, Fref, Fout);
if (div > 32) {
printk(KERN_ERR "wm8900: Invalid FLL division rate %u, "
- "Fref=%d, Fout=%d, target=%d\n",
+ "Fref=%u, Fout=%u, target=%u\n",
div, Fref, Fout, target);
return -EINVAL;
}
int sysclk;
/* Reference counts */
- int charge_pump_users;
int class_w_users;
int playback_active;
int capture_active;
#define WM8903_OUTPUT_INT 0x2
#define WM8903_OUTPUT_IN 0x1
+static int wm8903_cp_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ WARN_ON(event != SND_SOC_DAPM_POST_PMU);
+ mdelay(4);
+
+ return 0;
+}
+
/*
* Event for headphone and line out amplifier power changes. Special
* power up/down sequences are required in order to maximise pop/click
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
- struct wm8903_priv *wm8903 = codec->private_data;
- struct i2c_client *i2c = codec->control_data;
u16 val;
u16 reg;
+ u16 dcs_reg;
+ u16 dcs_bit;
int shift;
- u16 cp_reg = wm8903_read(codec, WM8903_CHARGE_PUMP_0);
switch (w->reg) {
case WM8903_POWER_MANAGEMENT_2:
reg = WM8903_ANALOGUE_HP_0;
+ dcs_bit = 0 + w->shift;
break;
case WM8903_POWER_MANAGEMENT_3:
reg = WM8903_ANALOGUE_LINEOUT_0;
+ dcs_bit = 2 + w->shift;
break;
default:
BUG();
/* Short the output */
val &= ~(WM8903_OUTPUT_SHORT << shift);
wm8903_write(codec, reg, val);
-
- wm8903->charge_pump_users++;
-
- dev_dbg(&i2c->dev, "Charge pump use count now %d\n",
- wm8903->charge_pump_users);
-
- if (wm8903->charge_pump_users == 1) {
- dev_dbg(&i2c->dev, "Enabling charge pump\n");
- wm8903_write(codec, WM8903_CHARGE_PUMP_0,
- cp_reg | WM8903_CP_ENA);
- mdelay(4);
- }
}
if (event & SND_SOC_DAPM_POST_PMU) {
val |= (WM8903_OUTPUT_OUT << shift);
wm8903_write(codec, reg, val);
+ /* Enable the DC servo */
+ dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0);
+ dcs_reg |= dcs_bit;
+ wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg);
+
/* Remove the short */
val |= (WM8903_OUTPUT_SHORT << shift);
wm8903_write(codec, reg, val);
val &= ~(WM8903_OUTPUT_SHORT << shift);
wm8903_write(codec, reg, val);
+ /* Disable the DC servo */
+ dcs_reg = wm8903_read(codec, WM8903_DC_SERVO_0);
+ dcs_reg &= ~dcs_bit;
+ wm8903_write(codec, WM8903_DC_SERVO_0, dcs_reg);
+
/* Then disable the intermediate and output stages */
val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT |
WM8903_OUTPUT_IN) << shift);
wm8903_write(codec, reg, val);
}
- if (event & SND_SOC_DAPM_POST_PMD) {
- wm8903->charge_pump_users--;
-
- dev_dbg(&i2c->dev, "Charge pump use count now %d\n",
- wm8903->charge_pump_users);
-
- if (wm8903->charge_pump_users == 0) {
- dev_dbg(&i2c->dev, "Disabling charge pump\n");
- wm8903_write(codec, WM8903_CHARGE_PUMP_0,
- cp_reg & ~WM8903_CP_ENA);
- }
- }
-
return 0;
}
/* ALSA can only do steps of .01dB */
static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
+static const DECLARE_TLV_DB_SCALE(digital_sidetone_tlv, -3600, 300, 0);
static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
static const DECLARE_TLV_DB_SCALE(drc_tlv_thresh, 0, 75, 0);
SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 4, 3, rinput_mux_text);
+static const char *sidetone_text[] = {
+ "None", "Left", "Right"
+};
+
+static const struct soc_enum lsidetone_enum =
+ SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 2, 3, sidetone_text);
+
+static const struct soc_enum rsidetone_enum =
+ SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text);
+
static const struct snd_kcontrol_new wm8903_snd_controls[] = {
/* Input PGAs - No TLV since the scale depends on PGA mode */
SOC_ENUM("ADC Companding Mode", adc_companding),
SOC_SINGLE("ADC Companding Switch", WM8903_AUDIO_INTERFACE_0, 3, 1, 0),
+SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8903_DAC_DIGITAL_0, 4, 8,
+ 12, 0, digital_sidetone_tlv),
+
/* DAC */
SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8903_DAC_DIGITAL_VOLUME_LEFT,
WM8903_DAC_DIGITAL_VOLUME_RIGHT, 1, 120, 0, digital_tlv),
static const struct snd_kcontrol_new rinput_inv_mux =
SOC_DAPM_ENUM("Right Inverting Input Mux", rinput_inv_enum);
+static const struct snd_kcontrol_new lsidetone_mux =
+ SOC_DAPM_ENUM("DACL Sidetone Mux", lsidetone_enum);
+
+static const struct snd_kcontrol_new rsidetone_mux =
+ SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum);
+
static const struct snd_kcontrol_new left_output_mixer[] = {
SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0),
SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0),
SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8903_POWER_MANAGEMENT_6, 1, 0),
SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8903_POWER_MANAGEMENT_6, 0, 0),
+SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &lsidetone_mux),
+SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &rsidetone_mux),
+
SND_SOC_DAPM_DAC("DACL", "Left Playback", WM8903_POWER_MANAGEMENT_6, 3, 0),
SND_SOC_DAPM_DAC("DACR", "Right Playback", WM8903_POWER_MANAGEMENT_6, 2, 0),
SND_SOC_DAPM_PGA_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
1, 0, NULL, 0, wm8903_output_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
0, 0, NULL, 0, wm8903_output_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0,
NULL, 0, wm8903_output_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0,
NULL, 0, wm8903_output_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
- SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
+ SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0,
NULL, 0),
SND_SOC_DAPM_PGA("Right Speaker PGA", WM8903_POWER_MANAGEMENT_5, 0, 0,
NULL, 0),
+SND_SOC_DAPM_SUPPLY("Charge Pump", WM8903_CHARGE_PUMP_0, 0, 0,
+ wm8903_cp_event, SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8903_CLOCK_RATES_2, 1, 0, NULL, 0),
};
static const struct snd_soc_dapm_route intercon[] = {
{ "Right Input PGA", NULL, "Right Input Mode Mux" },
{ "ADCL", NULL, "Left Input PGA" },
+ { "ADCL", NULL, "CLK_DSP" },
{ "ADCR", NULL, "Right Input PGA" },
+ { "ADCR", NULL, "CLK_DSP" },
+
+ { "DACL Sidetone", "Left", "ADCL" },
+ { "DACL Sidetone", "Right", "ADCR" },
+ { "DACR Sidetone", "Left", "ADCL" },
+ { "DACR Sidetone", "Right", "ADCR" },
+
+ { "DACL", NULL, "DACL Sidetone" },
+ { "DACL", NULL, "CLK_DSP" },
+ { "DACR", NULL, "DACR Sidetone" },
+ { "DACR", NULL, "CLK_DSP" },
{ "Left Output Mixer", "Left Bypass Switch", "Left Input PGA" },
{ "Left Output Mixer", "Right Bypass Switch", "Right Input PGA" },
{ "ROP", NULL, "Right Speaker PGA" },
{ "RON", NULL, "Right Speaker PGA" },
+
+ { "Left Headphone Output PGA", NULL, "Charge Pump" },
+ { "Right Headphone Output PGA", NULL, "Charge Pump" },
+ { "Left Line Output PGA", NULL, "Charge Pump" },
+ { "Right Line Output PGA", NULL, "Charge Pump" },
};
static int wm8903_add_widgets(struct snd_soc_codec *codec)
wm8903_write(codec, WM8903_CLOCK_RATES_2,
WM8903_CLK_SYS_ENA);
+ /* Change DC servo dither level in startup sequence */
+ wm8903_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11);
+ wm8903_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257);
+ wm8903_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2);
+
wm8903_run_sequence(codec, 0);
wm8903_sync_reg_cache(codec, codec->reg_cache);
if (wm8903->master_substream) {
master_runtime = wm8903->master_substream->runtime;
- dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n",
- master_runtime->sample_bits,
- master_runtime->rate);
-
- snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_RATE,
- master_runtime->rate,
- master_runtime->rate);
+ dev_dbg(&i2c->dev, "Constraining to %d bits\n",
+ master_runtime->sample_bits);
snd_pcm_hw_constraint_minmax(substream->runtime,
SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
.formats = WM8903_FORMATS,
},
.ops = &wm8903_dai_ops,
+ .symmetric_rates = 1,
};
EXPORT_SYMBOL_GPL(wm8903_dai);
--- /dev/null
+/*
+ * wm8940.c -- WM8940 ALSA Soc Audio driver
+ *
+ * Author: Jonathan Cameron <jic23@cam.ac.uk>
+ *
+ * Based on wm8510.c
+ * Copyright 2006 Wolfson Microelectronics PLC.
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * 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.
+ *
+ * Not currently handled:
+ * Notch filter control
+ * AUXMode (inverting vs mixer)
+ * No means to obtain current gain if alc enabled.
+ * No use made of gpio
+ * Fast VMID discharge for power down
+ * Soft Start
+ * DLR and ALR Swaps not enabled
+ * Digital Sidetone not supported
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/spi/spi.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8940.h"
+
+struct wm8940_priv {
+ unsigned int sysclk;
+ u16 reg_cache[WM8940_CACHEREGNUM];
+ struct snd_soc_codec codec;
+};
+
+static u16 wm8940_reg_defaults[] = {
+ 0x8940, /* Soft Reset */
+ 0x0000, /* Power 1 */
+ 0x0000, /* Power 2 */
+ 0x0000, /* Power 3 */
+ 0x0010, /* Interface Control */
+ 0x0000, /* Companding Control */
+ 0x0140, /* Clock Control */
+ 0x0000, /* Additional Controls */
+ 0x0000, /* GPIO Control */
+ 0x0002, /* Auto Increment Control */
+ 0x0000, /* DAC Control */
+ 0x00FF, /* DAC Volume */
+ 0,
+ 0,
+ 0x0100, /* ADC Control */
+ 0x00FF, /* ADC Volume */
+ 0x0000, /* Notch Filter 1 Control 1 */
+ 0x0000, /* Notch Filter 1 Control 2 */
+ 0x0000, /* Notch Filter 2 Control 1 */
+ 0x0000, /* Notch Filter 2 Control 2 */
+ 0x0000, /* Notch Filter 3 Control 1 */
+ 0x0000, /* Notch Filter 3 Control 2 */
+ 0x0000, /* Notch Filter 4 Control 1 */
+ 0x0000, /* Notch Filter 4 Control 2 */
+ 0x0032, /* DAC Limit Control 1 */
+ 0x0000, /* DAC Limit Control 2 */
+ 0,
+ 0,
+ 0,
+ 0,
+ 0,
+ 0,
+ 0x0038, /* ALC Control 1 */
+ 0x000B, /* ALC Control 2 */
+ 0x0032, /* ALC Control 3 */
+ 0x0000, /* Noise Gate */
+ 0x0041, /* PLLN */
+ 0x000C, /* PLLK1 */
+ 0x0093, /* PLLK2 */
+ 0x00E9, /* PLLK3 */
+ 0,
+ 0,
+ 0x0030, /* ALC Control 4 */
+ 0,
+ 0x0002, /* Input Control */
+ 0x0050, /* PGA Gain */
+ 0,
+ 0x0002, /* ADC Boost Control */
+ 0,
+ 0x0002, /* Output Control */
+ 0x0000, /* Speaker Mixer Control */
+ 0,
+ 0,
+ 0,
+ 0x0079, /* Speaker Volume */
+ 0,
+ 0x0000, /* Mono Mixer Control */
+};
+
+static inline unsigned int wm8940_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+
+ if (reg >= ARRAY_SIZE(wm8940_reg_defaults))
+ return -1;
+
+ return cache[reg];
+}
+
+static inline int wm8940_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+
+ if (reg >= ARRAY_SIZE(wm8940_reg_defaults))
+ return -1;
+
+ cache[reg] = value;
+
+ return 0;
+}
+
+static int wm8940_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ int ret;
+ u8 data[3] = { reg,
+ (value & 0xff00) >> 8,
+ (value & 0x00ff)
+ };
+
+ wm8940_write_reg_cache(codec, reg, value);
+
+ ret = codec->hw_write(codec->control_data, data, 3);
+
+ if (ret < 0)
+ return ret;
+ else if (ret != 3)
+ return -EIO;
+ return 0;
+}
+
+static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" };
+static const struct soc_enum wm8940_adc_companding_enum
+= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 1, 4, wm8940_companding);
+static const struct soc_enum wm8940_dac_companding_enum
+= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 3, 4, wm8940_companding);
+
+static const char *wm8940_alc_mode_text[] = {"ALC", "Limiter"};
+static const struct soc_enum wm8940_alc_mode_enum
+= SOC_ENUM_SINGLE(WM8940_ALC3, 8, 2, wm8940_alc_mode_text);
+
+static const char *wm8940_mic_bias_level_text[] = {"0.9", "0.65"};
+static const struct soc_enum wm8940_mic_bias_level_enum
+= SOC_ENUM_SINGLE(WM8940_INPUTCTL, 8, 2, wm8940_mic_bias_level_text);
+
+static const char *wm8940_filter_mode_text[] = {"Audio", "Application"};
+static const struct soc_enum wm8940_filter_mode_enum
+= SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text);
+
+static DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1);
+static DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_pga_vol_tlv, -1200, 75, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_min_tlv, -1200, 600, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_max_tlv, 675, 600, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_alc_tar_tlv, -2250, 50, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_lim_boost_tlv, 0, 100, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_lim_thresh_tlv, -600, 100, 0);
+static DECLARE_TLV_DB_SCALE(wm8940_adc_tlv, -12750, 50, 1);
+static DECLARE_TLV_DB_SCALE(wm8940_capture_boost_vol_tlv, 0, 2000, 0);
+
+static const struct snd_kcontrol_new wm8940_snd_controls[] = {
+ SOC_SINGLE("Digital Loopback Switch", WM8940_COMPANDINGCTL,
+ 6, 1, 0),
+ SOC_ENUM("DAC Companding", wm8940_dac_companding_enum),
+ SOC_ENUM("ADC Companding", wm8940_adc_companding_enum),
+
+ SOC_ENUM("ALC Mode", wm8940_alc_mode_enum),
+ SOC_SINGLE("ALC Switch", WM8940_ALC1, 8, 1, 0),
+ SOC_SINGLE_TLV("ALC Capture Max Gain", WM8940_ALC1,
+ 3, 7, 1, wm8940_alc_max_tlv),
+ SOC_SINGLE_TLV("ALC Capture Min Gain", WM8940_ALC1,
+ 0, 7, 0, wm8940_alc_min_tlv),
+ SOC_SINGLE_TLV("ALC Capture Target", WM8940_ALC2,
+ 0, 14, 0, wm8940_alc_tar_tlv),
+ SOC_SINGLE("ALC Capture Hold", WM8940_ALC2, 4, 10, 0),
+ SOC_SINGLE("ALC Capture Decay", WM8940_ALC3, 4, 10, 0),
+ SOC_SINGLE("ALC Capture Attach", WM8940_ALC3, 0, 10, 0),
+ SOC_SINGLE("ALC ZC Switch", WM8940_ALC4, 1, 1, 0),
+ SOC_SINGLE("ALC Capture Noise Gate Switch", WM8940_NOISEGATE,
+ 3, 1, 0),
+ SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8940_NOISEGATE,
+ 0, 7, 0),
+
+ SOC_SINGLE("DAC Playback Limiter Switch", WM8940_DACLIM1, 8, 1, 0),
+ SOC_SINGLE("DAC Playback Limiter Attack", WM8940_DACLIM1, 0, 9, 0),
+ SOC_SINGLE("DAC Playback Limiter Decay", WM8940_DACLIM1, 4, 11, 0),
+ SOC_SINGLE_TLV("DAC Playback Limiter Threshold", WM8940_DACLIM2,
+ 4, 9, 1, wm8940_lim_thresh_tlv),
+ SOC_SINGLE_TLV("DAC Playback Limiter Boost", WM8940_DACLIM2,
+ 0, 12, 0, wm8940_lim_boost_tlv),
+
+ SOC_SINGLE("Capture PGA ZC Switch", WM8940_PGAGAIN, 7, 1, 0),
+ SOC_SINGLE_TLV("Capture PGA Volume", WM8940_PGAGAIN,
+ 0, 63, 0, wm8940_pga_vol_tlv),
+ SOC_SINGLE_TLV("Digital Playback Volume", WM8940_DACVOL,
+ 0, 255, 0, wm8940_adc_tlv),
+ SOC_SINGLE_TLV("Digital Capture Volume", WM8940_ADCVOL,
+ 0, 255, 0, wm8940_adc_tlv),
+ SOC_ENUM("Mic Bias Level", wm8940_mic_bias_level_enum),
+ SOC_SINGLE_TLV("Capture Boost Volue", WM8940_ADCBOOST,
+ 8, 1, 0, wm8940_capture_boost_vol_tlv),
+ SOC_SINGLE_TLV("Speaker Playback Volume", WM8940_SPKVOL,
+ 0, 63, 0, wm8940_spk_vol_tlv),
+ SOC_SINGLE("Speaker Playback Switch", WM8940_SPKVOL, 6, 1, 1),
+
+ SOC_SINGLE_TLV("Speaker Mixer Line Bypass Volume", WM8940_SPKVOL,
+ 8, 1, 1, wm8940_att_tlv),
+ SOC_SINGLE("Speaker Playback ZC Switch", WM8940_SPKVOL, 7, 1, 0),
+
+ SOC_SINGLE("Mono Out Switch", WM8940_MONOMIX, 6, 1, 1),
+ SOC_SINGLE_TLV("Mono Mixer Line Bypass Volume", WM8940_MONOMIX,
+ 7, 1, 1, wm8940_att_tlv),
+
+ SOC_SINGLE("High Pass Filter Switch", WM8940_ADC, 8, 1, 0),
+ SOC_ENUM("High Pass Filter Mode", wm8940_filter_mode_enum),
+ SOC_SINGLE("High Pass Filter Cut Off", WM8940_ADC, 4, 7, 0),
+ SOC_SINGLE("ADC Inversion Switch", WM8940_ADC, 0, 1, 0),
+ SOC_SINGLE("DAC Inversion Switch", WM8940_DAC, 0, 1, 0),
+ SOC_SINGLE("DAC Auto Mute Switch", WM8940_DAC, 2, 1, 0),
+ SOC_SINGLE("ZC Timeout Clock Switch", WM8940_ADDCNTRL, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8940_speaker_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_SPKMIX, 1, 1, 0),
+ SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_SPKMIX, 5, 1, 0),
+ SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_SPKMIX, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8940_mono_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Line Bypass Switch", WM8940_MONOMIX, 1, 1, 0),
+ SOC_DAPM_SINGLE("Aux Playback Switch", WM8940_MONOMIX, 2, 1, 0),
+ SOC_DAPM_SINGLE("PCM Playback Switch", WM8940_MONOMIX, 0, 1, 0),
+};
+
+static DECLARE_TLV_DB_SCALE(wm8940_boost_vol_tlv, -1500, 300, 1);
+static const struct snd_kcontrol_new wm8940_input_boost_controls[] = {
+ SOC_DAPM_SINGLE("Mic PGA Switch", WM8940_PGAGAIN, 6, 1, 1),
+ SOC_DAPM_SINGLE_TLV("Aux Volume", WM8940_ADCBOOST,
+ 0, 7, 0, wm8940_boost_vol_tlv),
+ SOC_DAPM_SINGLE_TLV("Mic Volume", WM8940_ADCBOOST,
+ 4, 7, 0, wm8940_boost_vol_tlv),
+};
+
+static const struct snd_kcontrol_new wm8940_micpga_controls[] = {
+ SOC_DAPM_SINGLE("AUX Switch", WM8940_INPUTCTL, 2, 1, 0),
+ SOC_DAPM_SINGLE("MICP Switch", WM8940_INPUTCTL, 0, 1, 0),
+ SOC_DAPM_SINGLE("MICN Switch", WM8940_INPUTCTL, 1, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8940_dapm_widgets[] = {
+ SND_SOC_DAPM_MIXER("Speaker Mixer", WM8940_POWER3, 2, 0,
+ &wm8940_speaker_mixer_controls[0],
+ ARRAY_SIZE(wm8940_speaker_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Mono Mixer", WM8940_POWER3, 3, 0,
+ &wm8940_mono_mixer_controls[0],
+ ARRAY_SIZE(wm8940_mono_mixer_controls)),
+ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8940_POWER3, 0, 0),
+
+ SND_SOC_DAPM_PGA("SpkN Out", WM8940_POWER3, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("SpkP Out", WM8940_POWER3, 6, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Mono Out", WM8940_POWER3, 7, 0, NULL, 0),
+ SND_SOC_DAPM_OUTPUT("MONOOUT"),
+ SND_SOC_DAPM_OUTPUT("SPKOUTP"),
+ SND_SOC_DAPM_OUTPUT("SPKOUTN"),
+
+ SND_SOC_DAPM_PGA("Aux Input", WM8940_POWER1, 6, 0, NULL, 0),
+ SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8940_POWER2, 0, 0),
+ SND_SOC_DAPM_MIXER("Mic PGA", WM8940_POWER2, 2, 0,
+ &wm8940_micpga_controls[0],
+ ARRAY_SIZE(wm8940_micpga_controls)),
+ SND_SOC_DAPM_MIXER("Boost Mixer", WM8940_POWER2, 4, 0,
+ &wm8940_input_boost_controls[0],
+ ARRAY_SIZE(wm8940_input_boost_controls)),
+ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8940_POWER1, 4, 0),
+
+ SND_SOC_DAPM_INPUT("MICN"),
+ SND_SOC_DAPM_INPUT("MICP"),
+ SND_SOC_DAPM_INPUT("AUX"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+ /* Mono output mixer */
+ {"Mono Mixer", "PCM Playback Switch", "DAC"},
+ {"Mono Mixer", "Aux Playback Switch", "Aux Input"},
+ {"Mono Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+ /* Speaker output mixer */
+ {"Speaker Mixer", "PCM Playback Switch", "DAC"},
+ {"Speaker Mixer", "Aux Playback Switch", "Aux Input"},
+ {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"},
+
+ /* Outputs */
+ {"Mono Out", NULL, "Mono Mixer"},
+ {"MONOOUT", NULL, "Mono Out"},
+ {"SpkN Out", NULL, "Speaker Mixer"},
+ {"SpkP Out", NULL, "Speaker Mixer"},
+ {"SPKOUTN", NULL, "SpkN Out"},
+ {"SPKOUTP", NULL, "SpkP Out"},
+
+ /* Microphone PGA */
+ {"Mic PGA", "MICN Switch", "MICN"},
+ {"Mic PGA", "MICP Switch", "MICP"},
+ {"Mic PGA", "AUX Switch", "AUX"},
+
+ /* Boost Mixer */
+ {"Boost Mixer", "Mic PGA Switch", "Mic PGA"},
+ {"Boost Mixer", "Mic Volume", "MICP"},
+ {"Boost Mixer", "Aux Volume", "Aux Input"},
+
+ {"ADC", NULL, "Boost Mixer"},
+};
+
+static int wm8940_add_widgets(struct snd_soc_codec *codec)
+{
+ int ret;
+
+ ret = snd_soc_dapm_new_controls(codec, wm8940_dapm_widgets,
+ ARRAY_SIZE(wm8940_dapm_widgets));
+ if (ret)
+ goto error_ret;
+ ret = snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ if (ret)
+ goto error_ret;
+ ret = snd_soc_dapm_new_widgets(codec);
+
+error_ret:
+ return ret;
+}
+
+#define wm8940_reset(c) wm8940_write(c, WM8940_SOFTRESET, 0);
+
+static int wm8940_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFE67;
+ u16 clk = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0x1fe;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ clk |= 1;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+ wm8940_write(codec, WM8940_CLOCK, clk);
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= (2 << 3);
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= (1 << 3);
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= (3 << 3);
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= (3 << 3) | (1 << 7);
+ break;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= (1 << 7);
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= (1 << 8);
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= (1 << 8) | (1 << 7);
+ break;
+ }
+
+ wm8940_write(codec, WM8940_IFACE, iface);
+
+ return 0;
+}
+
+static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ u16 iface = wm8940_read_reg_cache(codec, WM8940_IFACE) & 0xFD9F;
+ u16 addcntrl = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFF1;
+ u16 companding = wm8940_read_reg_cache(codec,
+ WM8940_COMPANDINGCTL) & 0xFFDF;
+ int ret;
+
+ /* LoutR control */
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE
+ && params_channels(params) == 2)
+ iface |= (1 << 9);
+
+ switch (params_rate(params)) {
+ case SNDRV_PCM_RATE_8000:
+ addcntrl |= (0x5 << 1);
+ break;
+ case SNDRV_PCM_RATE_11025:
+ addcntrl |= (0x4 << 1);
+ break;
+ case SNDRV_PCM_RATE_16000:
+ addcntrl |= (0x3 << 1);
+ break;
+ case SNDRV_PCM_RATE_22050:
+ addcntrl |= (0x2 << 1);
+ break;
+ case SNDRV_PCM_RATE_32000:
+ addcntrl |= (0x1 << 1);
+ break;
+ case SNDRV_PCM_RATE_44100:
+ case SNDRV_PCM_RATE_48000:
+ break;
+ }
+ ret = wm8940_write(codec, WM8940_ADDCNTRL, addcntrl);
+ if (ret)
+ goto error_ret;
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ companding = companding | (1 << 5);
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= (1 << 5);
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= (2 << 5);
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= (3 << 5);
+ break;
+ }
+ ret = wm8940_write(codec, WM8940_COMPANDINGCTL, companding);
+ if (ret)
+ goto error_ret;
+ ret = wm8940_write(codec, WM8940_IFACE, iface);
+
+error_ret:
+ return ret;
+}
+
+static int wm8940_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8940_read_reg_cache(codec, WM8940_DAC) & 0xffbf;
+
+ if (mute)
+ mute_reg |= 0x40;
+
+ return wm8940_write(codec, WM8940_DAC, mute_reg);
+}
+
+static int wm8940_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 val;
+ u16 pwr_reg = wm8940_read_reg_cache(codec, WM8940_POWER1) & 0x1F0;
+ int ret = 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ /* ensure bufioen and biasen */
+ pwr_reg |= (1 << 2) | (1 << 3);
+ /* Enable thermal shutdown */
+ val = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL);
+ ret = wm8940_write(codec, WM8940_OUTPUTCTL, val | 0x2);
+ if (ret)
+ break;
+ /* set vmid to 75k */
+ ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1);
+ break;
+ case SND_SOC_BIAS_PREPARE:
+ /* ensure bufioen and biasen */
+ pwr_reg |= (1 << 2) | (1 << 3);
+ ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x1);
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ /* ensure bufioen and biasen */
+ pwr_reg |= (1 << 2) | (1 << 3);
+ /* set vmid to 300k for standby */
+ ret = wm8940_write(codec, WM8940_POWER1, pwr_reg | 0x2);
+ break;
+ case SND_SOC_BIAS_OFF:
+ ret = wm8940_write(codec, WM8940_POWER1, pwr_reg);
+ break;
+ }
+
+ return ret;
+}
+
+struct pll_ {
+ unsigned int pre_scale:2;
+ unsigned int n:4;
+ unsigned int k;
+};
+
+static struct pll_ pll_div;
+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 24) * 10)
+static void pll_factors(unsigned int target, unsigned int source)
+{
+ unsigned long long Kpart;
+ unsigned int K, Ndiv, Nmod;
+ /* The left shift ist to avoid accuracy loss when right shifting */
+ Ndiv = target / source;
+
+ if (Ndiv > 12) {
+ source <<= 1;
+ /* Multiply by 2 */
+ pll_div.pre_scale = 0;
+ Ndiv = target / source;
+ } else if (Ndiv < 3) {
+ source >>= 2;
+ /* Divide by 4 */
+ pll_div.pre_scale = 3;
+ Ndiv = target / source;
+ } else if (Ndiv < 6) {
+ source >>= 1;
+ /* divide by 2 */
+ pll_div.pre_scale = 2;
+ Ndiv = target / source;
+ } else
+ pll_div.pre_scale = 1;
+
+ if ((Ndiv < 6) || (Ndiv > 12))
+ printk(KERN_WARNING
+ "WM8940 N value %d outwith recommended range!d\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;
+}
+
+/* Untested at the moment */
+static int wm8940_set_dai_pll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ /* Turn off PLL */
+ reg = wm8940_read_reg_cache(codec, WM8940_POWER1);
+ wm8940_write(codec, WM8940_POWER1, reg & 0x1df);
+
+ if (freq_in == 0 || freq_out == 0) {
+ /* Clock CODEC directly from MCLK */
+ reg = wm8940_read_reg_cache(codec, WM8940_CLOCK);
+ wm8940_write(codec, WM8940_CLOCK, reg & 0x0ff);
+ /* Pll power down */
+ wm8940_write(codec, WM8940_PLLN, (1 << 7));
+ return 0;
+ }
+
+ /* Pll is followed by a frequency divide by 4 */
+ pll_factors(freq_out*4, freq_in);
+ if (pll_div.k)
+ wm8940_write(codec, WM8940_PLLN,
+ (pll_div.pre_scale << 4) | pll_div.n | (1 << 6));
+ else /* No factional component */
+ wm8940_write(codec, WM8940_PLLN,
+ (pll_div.pre_scale << 4) | pll_div.n);
+ wm8940_write(codec, WM8940_PLLK1, pll_div.k >> 18);
+ wm8940_write(codec, WM8940_PLLK2, (pll_div.k >> 9) & 0x1ff);
+ wm8940_write(codec, WM8940_PLLK3, pll_div.k & 0x1ff);
+ /* Enable the PLL */
+ reg = wm8940_read_reg_cache(codec, WM8940_POWER1);
+ wm8940_write(codec, WM8940_POWER1, reg | 0x020);
+
+ /* Run CODEC from PLL instead of MCLK */
+ reg = wm8940_read_reg_cache(codec, WM8940_CLOCK);
+ wm8940_write(codec, WM8940_CLOCK, reg | 0x100);
+
+ return 0;
+}
+
+static int wm8940_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8940_priv *wm8940 = codec->private_data;
+
+ switch (freq) {
+ case 11289600:
+ case 12000000:
+ case 12288000:
+ case 16934400:
+ case 18432000:
+ wm8940->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
+static int wm8940_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+ int ret = 0;
+
+ switch (div_id) {
+ case WM8940_BCLKDIV:
+ reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFFEF3;
+ ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 2));
+ break;
+ case WM8940_MCLKDIV:
+ reg = wm8940_read_reg_cache(codec, WM8940_CLOCK) & 0xFF1F;
+ ret = wm8940_write(codec, WM8940_CLOCK, reg | (div << 5));
+ break;
+ case WM8940_OPCLKDIV:
+ reg = wm8940_read_reg_cache(codec, WM8940_ADDCNTRL) & 0xFFCF;
+ ret = wm8940_write(codec, WM8940_ADDCNTRL, reg | (div << 4));
+ break;
+ }
+ return ret;
+}
+
+#define WM8940_RATES SNDRV_PCM_RATE_8000_48000
+
+#define WM8940_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+ SNDRV_PCM_FMTBIT_S16_LE | \
+ SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE | \
+ SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops wm8940_dai_ops = {
+ .hw_params = wm8940_i2s_hw_params,
+ .set_sysclk = wm8940_set_dai_sysclk,
+ .digital_mute = wm8940_mute,
+ .set_fmt = wm8940_set_dai_fmt,
+ .set_clkdiv = wm8940_set_dai_clkdiv,
+ .set_pll = wm8940_set_dai_pll,
+};
+
+struct snd_soc_dai wm8940_dai = {
+ .name = "WM8940",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8940_RATES,
+ .formats = WM8940_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8940_RATES,
+ .formats = WM8940_FORMATS,
+ },
+ .ops = &wm8940_dai_ops,
+ .symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8940_dai);
+
+static int wm8940_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ return wm8940_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int wm8940_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ int i;
+ int ret;
+ u8 data[3];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware
+ * Could use auto incremented writes to speed this up
+ */
+ for (i = 0; i < ARRAY_SIZE(wm8940_reg_defaults); i++) {
+ data[0] = i;
+ data[1] = (cache[i] & 0xFF00) >> 8;
+ data[2] = cache[i] & 0x00FF;
+ ret = codec->hw_write(codec->control_data, data, 3);
+ if (ret < 0)
+ goto error_ret;
+ else if (ret != 3) {
+ ret = -EIO;
+ goto error_ret;
+ }
+ }
+ ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ if (ret)
+ goto error_ret;
+ ret = wm8940_set_bias_level(codec, codec->suspend_bias_level);
+
+error_ret:
+ return ret;
+}
+
+static struct snd_soc_codec *wm8940_codec;
+
+static int wm8940_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+
+ int ret = 0;
+
+ if (wm8940_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = wm8940_codec;
+ codec = wm8940_codec;
+
+ mutex_init(&codec->mutex);
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ ret = snd_soc_add_controls(codec, wm8940_snd_controls,
+ ARRAY_SIZE(wm8940_snd_controls));
+ if (ret)
+ goto error_free_pcms;
+ ret = wm8940_add_widgets(codec);
+ if (ret)
+ goto error_free_pcms;
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto error_free_pcms;
+ }
+
+ return ret;
+
+error_free_pcms:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+static int wm8940_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8940 = {
+ .probe = wm8940_probe,
+ .remove = wm8940_remove,
+ .suspend = wm8940_suspend,
+ .resume = wm8940_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8940);
+
+static int wm8940_register(struct wm8940_priv *wm8940)
+{
+ struct wm8940_setup_data *pdata = wm8940->codec.dev->platform_data;
+ struct snd_soc_codec *codec = &wm8940->codec;
+ int ret;
+ u16 reg;
+ if (wm8940_codec) {
+ dev_err(codec->dev, "Another WM8940 is registered\n");
+ return -EINVAL;
+ }
+
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = wm8940;
+ codec->name = "WM8940";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8940_read_reg_cache;
+ codec->write = wm8940_write;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8940_set_bias_level;
+ codec->dai = &wm8940_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults);
+ codec->reg_cache = &wm8940->reg_cache;
+
+ memcpy(codec->reg_cache, wm8940_reg_defaults,
+ sizeof(wm8940_reg_defaults));
+
+ ret = wm8940_reset(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset\n");
+ return ret;
+ }
+
+ wm8940_dai.dev = codec->dev;
+
+ wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ ret = wm8940_write(codec, WM8940_POWER1, 0x180);
+ if (ret < 0)
+ return ret;
+
+ if (!pdata)
+ dev_warn(codec->dev, "No platform data supplied\n");
+ else {
+ reg = wm8940_read_reg_cache(codec, WM8940_OUTPUTCTL);
+ ret = wm8940_write(codec, WM8940_OUTPUTCTL, reg | pdata->vroi);
+ if (ret < 0)
+ return ret;
+ }
+
+
+ wm8940_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_dai(&wm8940_dai);
+ if (ret) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ snd_soc_unregister_codec(codec);
+ return ret;
+ }
+
+ return 0;
+}
+
+static void wm8940_unregister(struct wm8940_priv *wm8940)
+{
+ wm8940_set_bias_level(&wm8940->codec, SND_SOC_BIAS_OFF);
+ snd_soc_unregister_dai(&wm8940_dai);
+ snd_soc_unregister_codec(&wm8940->codec);
+ kfree(wm8940);
+ wm8940_codec = NULL;
+}
+
+static int wm8940_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct wm8940_priv *wm8940;
+ struct snd_soc_codec *codec;
+
+ wm8940 = kzalloc(sizeof *wm8940, GFP_KERNEL);
+ if (wm8940 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8940->codec;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+ i2c_set_clientdata(i2c, wm8940);
+ codec->control_data = i2c;
+ codec->dev = &i2c->dev;
+
+ return wm8940_register(wm8940);
+}
+
+static int __devexit wm8940_i2c_remove(struct i2c_client *client)
+{
+ struct wm8940_priv *wm8940 = i2c_get_clientdata(client);
+
+ wm8940_unregister(wm8940);
+
+ return 0;
+}
+
+static const struct i2c_device_id wm8940_i2c_id[] = {
+ { "wm8940", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8940_i2c_id);
+
+static struct i2c_driver wm8940_i2c_driver = {
+ .driver = {
+ .name = "WM8940 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8940_i2c_probe,
+ .remove = __devexit_p(wm8940_i2c_remove),
+ .id_table = wm8940_i2c_id,
+};
+
+static int __init wm8940_modinit(void)
+{
+ int ret;
+
+ ret = i2c_add_driver(&wm8940_i2c_driver);
+ if (ret)
+ printk(KERN_ERR "Failed to register WM8940 I2C driver: %d\n",
+ ret);
+ return ret;
+}
+module_init(wm8940_modinit);
+
+static void __exit wm8940_exit(void)
+{
+ i2c_del_driver(&wm8940_i2c_driver);
+}
+module_exit(wm8940_exit);
+
+MODULE_DESCRIPTION("ASoC WM8940 driver");
+MODULE_AUTHOR("Jonathan Cameron");
+MODULE_LICENSE("GPL");
--- /dev/null
+/*
+ * wm8940.h -- WM8940 Soc Audio driver
+ *
+ * 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 _WM8940_H
+#define _WM8940_H
+
+struct wm8940_setup_data {
+ /* Vref to analogue output resistance */
+#define WM8940_VROI_1K 0
+#define WM8940_VROI_30K 1
+ unsigned int vroi:1;
+};
+extern struct snd_soc_dai wm8940_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8940;
+
+/* WM8940 register space */
+#define WM8940_SOFTRESET 0x00
+#define WM8940_POWER1 0x01
+#define WM8940_POWER2 0x02
+#define WM8940_POWER3 0x03
+#define WM8940_IFACE 0x04
+#define WM8940_COMPANDINGCTL 0x05
+#define WM8940_CLOCK 0x06
+#define WM8940_ADDCNTRL 0x07
+#define WM8940_GPIO 0x08
+#define WM8940_CTLINT 0x09
+#define WM8940_DAC 0x0A
+#define WM8940_DACVOL 0x0B
+
+#define WM8940_ADC 0x0E
+#define WM8940_ADCVOL 0x0F
+#define WM8940_NOTCH1 0x10
+#define WM8940_NOTCH2 0x11
+#define WM8940_NOTCH3 0x12
+#define WM8940_NOTCH4 0x13
+#define WM8940_NOTCH5 0x14
+#define WM8940_NOTCH6 0x15
+#define WM8940_NOTCH7 0x16
+#define WM8940_NOTCH8 0x17
+#define WM8940_DACLIM1 0x18
+#define WM8940_DACLIM2 0x19
+
+#define WM8940_ALC1 0x20
+#define WM8940_ALC2 0x21
+#define WM8940_ALC3 0x22
+#define WM8940_NOISEGATE 0x23
+#define WM8940_PLLN 0x24
+#define WM8940_PLLK1 0x25
+#define WM8940_PLLK2 0x26
+#define WM8940_PLLK3 0x27
+
+#define WM8940_ALC4 0x2A
+
+#define WM8940_INPUTCTL 0x2C
+#define WM8940_PGAGAIN 0x2D
+
+#define WM8940_ADCBOOST 0x2F
+
+#define WM8940_OUTPUTCTL 0x31
+#define WM8940_SPKMIX 0x32
+
+#define WM8940_SPKVOL 0x36
+
+#define WM8940_MONOMIX 0x38
+
+#define WM8940_CACHEREGNUM 0x57
+
+
+/* Clock divider Id's */
+#define WM8940_BCLKDIV 0
+#define WM8940_MCLKDIV 1
+#define WM8940_OPCLKDIV 2
+
+/* MCLK clock dividers */
+#define WM8940_MCLKDIV_1 0
+#define WM8940_MCLKDIV_1_5 1
+#define WM8940_MCLKDIV_2 2
+#define WM8940_MCLKDIV_3 3
+#define WM8940_MCLKDIV_4 4
+#define WM8940_MCLKDIV_6 5
+#define WM8940_MCLKDIV_8 6
+#define WM8940_MCLKDIV_12 7
+
+/* BCLK clock dividers */
+#define WM8940_BCLKDIV_1 0
+#define WM8940_BCLKDIV_2 1
+#define WM8940_BCLKDIV_4 2
+#define WM8940_BCLKDIV_8 3
+#define WM8940_BCLKDIV_16 4
+#define WM8940_BCLKDIV_32 5
+
+/* PLL Out Dividers */
+#define WM8940_OPCLKDIV_1 0
+#define WM8940_OPCLKDIV_2 1
+#define WM8940_OPCLKDIV_3 2
+#define WM8940_OPCLKDIV_4 3
+
+#endif /* _WM8940_H */
+
--- /dev/null
+/*
+ * wm8960.c -- WM8960 ALSA SoC Audio driver
+ *
+ * Author: Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8960.h"
+
+#define AUDIO_NAME "wm8960"
+
+struct snd_soc_codec_device soc_codec_dev_wm8960;
+
+/* R25 - Power 1 */
+#define WM8960_VREF 0x40
+
+/* R28 - Anti-pop 1 */
+#define WM8960_POBCTRL 0x80
+#define WM8960_BUFDCOPEN 0x10
+#define WM8960_BUFIOEN 0x08
+#define WM8960_SOFT_ST 0x04
+#define WM8960_HPSTBY 0x01
+
+/* R29 - Anti-pop 2 */
+#define WM8960_DISOP 0x40
+
+/*
+ * wm8960 register cache
+ * We can't read the WM8960 register space when we are
+ * using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8960_reg[WM8960_CACHEREGNUM] = {
+ 0x0097, 0x0097, 0x0000, 0x0000,
+ 0x0000, 0x0008, 0x0000, 0x000a,
+ 0x01c0, 0x0000, 0x00ff, 0x00ff,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x007b, 0x0100, 0x0032,
+ 0x0000, 0x00c3, 0x00c3, 0x01c0,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0000, 0x0000,
+ 0x0100, 0x0100, 0x0050, 0x0050,
+ 0x0050, 0x0050, 0x0000, 0x0000,
+ 0x0000, 0x0000, 0x0040, 0x0000,
+ 0x0000, 0x0050, 0x0050, 0x0000,
+ 0x0002, 0x0037, 0x004d, 0x0080,
+ 0x0008, 0x0031, 0x0026, 0x00e9,
+};
+
+struct wm8960_priv {
+ u16 reg_cache[WM8960_CACHEREGNUM];
+ struct snd_soc_codec codec;
+};
+
+/*
+ * read wm8960 register cache
+ */
+static inline unsigned int wm8960_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg == WM8960_RESET)
+ return 0;
+ if (reg >= WM8960_CACHEREGNUM)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write wm8960 register cache
+ */
+static inline void wm8960_write_reg_cache(struct snd_soc_codec *codec,
+ u16 reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg >= WM8960_CACHEREGNUM)
+ return;
+ cache[reg] = value;
+}
+
+static inline unsigned int wm8960_read(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ return wm8960_read_reg_cache(codec, reg);
+}
+
+/*
+ * write to the WM8960 register space
+ */
+static int wm8960_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 WM8960 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8960_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define wm8960_reset(c) wm8960_write(c, WM8960_RESET, 0)
+
+/* enumerated controls */
+static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted",
+ "Right Inverted", "Stereo Inversion"};
+static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"};
+static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"};
+static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"};
+static const char *wm8960_alcmode[] = {"ALC", "Limiter"};
+
+static const struct soc_enum wm8960_enum[] = {
+ SOC_ENUM_SINGLE(WM8960_DACCTL1, 1, 4, wm8960_deemph),
+ SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity),
+ SOC_ENUM_SINGLE(WM8960_DACCTL2, 5, 4, wm8960_polarity),
+ SOC_ENUM_SINGLE(WM8960_3D, 6, 2, wm8960_3d_upper_cutoff),
+ SOC_ENUM_SINGLE(WM8960_3D, 5, 2, wm8960_3d_lower_cutoff),
+ SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc),
+ SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode),
+};
+
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
+static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+
+static const struct snd_kcontrol_new wm8960_snd_controls[] = {
+SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
+ 0, 63, 0, adc_tlv),
+SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
+ 6, 1, 0),
+SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
+ 7, 1, 0),
+
+SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC,
+ 0, 255, 0, dac_tlv),
+
+SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8960_LOUT1, WM8960_ROUT1,
+ 0, 127, 0, out_tlv),
+SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8960_LOUT1, WM8960_ROUT1,
+ 7, 1, 0),
+
+SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8960_LOUT2, WM8960_ROUT2,
+ 0, 127, 0, out_tlv),
+SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8960_LOUT2, WM8960_ROUT2,
+ 7, 1, 0),
+SOC_SINGLE("Speaker DC Volume", WM8960_CLASSD3, 3, 5, 0),
+SOC_SINGLE("Speaker AC Volume", WM8960_CLASSD3, 0, 5, 0),
+
+SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0),
+SOC_ENUM("ADC Polarity", wm8960_enum[1]),
+SOC_ENUM("Playback De-emphasis", wm8960_enum[0]),
+SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0),
+
+SOC_ENUM("DAC Polarity", wm8960_enum[2]),
+
+SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[3]),
+SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[4]),
+SOC_SINGLE("3D Volume", WM8960_3D, 1, 15, 0),
+SOC_SINGLE("3D Switch", WM8960_3D, 0, 1, 0),
+
+SOC_ENUM("ALC Function", wm8960_enum[5]),
+SOC_SINGLE("ALC Max Gain", WM8960_ALC1, 4, 7, 0),
+SOC_SINGLE("ALC Target", WM8960_ALC1, 0, 15, 1),
+SOC_SINGLE("ALC Min Gain", WM8960_ALC2, 4, 7, 0),
+SOC_SINGLE("ALC Hold Time", WM8960_ALC2, 0, 15, 0),
+SOC_ENUM("ALC Mode", wm8960_enum[6]),
+SOC_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0),
+SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0),
+
+SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0),
+SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0),
+
+SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH,
+ 0, 127, 0),
+
+SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume",
+ WM8960_BYPASS1, 4, 7, 1, bypass_tlv),
+SOC_SINGLE_TLV("Left Output Mixer LINPUT3 Volume",
+ WM8960_LOUTMIX, 4, 7, 1, bypass_tlv),
+SOC_SINGLE_TLV("Right Output Mixer Boost Bypass Volume",
+ WM8960_BYPASS2, 4, 7, 1, bypass_tlv),
+SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume",
+ WM8960_ROUTMIX, 4, 7, 1, bypass_tlv),
+};
+
+static const struct snd_kcontrol_new wm8960_lin_boost[] = {
+SOC_DAPM_SINGLE("LINPUT2 Switch", WM8960_LINPATH, 6, 1, 0),
+SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LINPATH, 7, 1, 0),
+SOC_DAPM_SINGLE("LINPUT1 Switch", WM8960_LINPATH, 8, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_lin[] = {
+SOC_DAPM_SINGLE("Boost Switch", WM8960_LINPATH, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_rin_boost[] = {
+SOC_DAPM_SINGLE("RINPUT2 Switch", WM8960_RINPATH, 6, 1, 0),
+SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_RINPATH, 7, 1, 0),
+SOC_DAPM_SINGLE("RINPUT1 Switch", WM8960_RINPATH, 8, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_rin[] = {
+SOC_DAPM_SINGLE("Boost Switch", WM8960_RINPATH, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_loutput_mixer[] = {
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_LOUTMIX, 8, 1, 0),
+SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LOUTMIX, 7, 1, 0),
+SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS1, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_routput_mixer[] = {
+SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_ROUTMIX, 8, 1, 0),
+SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_ROUTMIX, 7, 1, 0),
+SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS2, 7, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8960_mono_out[] = {
+SOC_DAPM_SINGLE("Left Switch", WM8960_MONOMIX1, 7, 1, 0),
+SOC_DAPM_SINGLE("Right Switch", WM8960_MONOMIX2, 7, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8960_dapm_widgets[] = {
+SND_SOC_DAPM_INPUT("LINPUT1"),
+SND_SOC_DAPM_INPUT("RINPUT1"),
+SND_SOC_DAPM_INPUT("LINPUT2"),
+SND_SOC_DAPM_INPUT("RINPUT2"),
+SND_SOC_DAPM_INPUT("LINPUT3"),
+SND_SOC_DAPM_INPUT("RINPUT3"),
+
+SND_SOC_DAPM_MICBIAS("MICB", WM8960_POWER1, 1, 0),
+
+SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8960_POWER1, 5, 0,
+ wm8960_lin_boost, ARRAY_SIZE(wm8960_lin_boost)),
+SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8960_POWER1, 4, 0,
+ wm8960_rin_boost, ARRAY_SIZE(wm8960_rin_boost)),
+
+SND_SOC_DAPM_MIXER("Left Input Mixer", WM8960_POWER3, 5, 0,
+ wm8960_lin, ARRAY_SIZE(wm8960_lin)),
+SND_SOC_DAPM_MIXER("Right Input Mixer", WM8960_POWER3, 4, 0,
+ wm8960_rin, ARRAY_SIZE(wm8960_rin)),
+
+SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER2, 3, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER2, 2, 0),
+
+SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0),
+
+SND_SOC_DAPM_MIXER("Left Output Mixer", WM8960_POWER3, 3, 0,
+ &wm8960_loutput_mixer[0],
+ ARRAY_SIZE(wm8960_loutput_mixer)),
+SND_SOC_DAPM_MIXER("Right Output Mixer", WM8960_POWER3, 2, 0,
+ &wm8960_routput_mixer[0],
+ ARRAY_SIZE(wm8960_routput_mixer)),
+
+SND_SOC_DAPM_MIXER("Mono Output Mixer", WM8960_POWER2, 1, 0,
+ &wm8960_mono_out[0],
+ ARRAY_SIZE(wm8960_mono_out)),
+
+SND_SOC_DAPM_PGA("LOUT1 PGA", WM8960_POWER2, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA("ROUT1 PGA", WM8960_POWER2, 5, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Left Speaker PGA", WM8960_POWER2, 4, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Right Speaker PGA", WM8960_POWER2, 3, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Right Speaker Output", WM8960_CLASSD1, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Left Speaker Output", WM8960_CLASSD1, 6, 0, NULL, 0),
+
+SND_SOC_DAPM_OUTPUT("SPK_LP"),
+SND_SOC_DAPM_OUTPUT("SPK_LN"),
+SND_SOC_DAPM_OUTPUT("HP_L"),
+SND_SOC_DAPM_OUTPUT("HP_R"),
+SND_SOC_DAPM_OUTPUT("SPK_RP"),
+SND_SOC_DAPM_OUTPUT("SPK_RN"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+};
+
+static const struct snd_soc_dapm_route audio_paths[] = {
+ { "Left Boost Mixer", "LINPUT1 Switch", "LINPUT1" },
+ { "Left Boost Mixer", "LINPUT2 Switch", "LINPUT2" },
+ { "Left Boost Mixer", "LINPUT3 Switch", "LINPUT3" },
+
+ { "Left Input Mixer", "Boost Switch", "Left Boost Mixer", },
+ { "Left Input Mixer", NULL, "LINPUT1", }, /* Really Boost Switch */
+ { "Left Input Mixer", NULL, "LINPUT2" },
+ { "Left Input Mixer", NULL, "LINPUT3" },
+
+ { "Right Boost Mixer", "RINPUT1 Switch", "RINPUT1" },
+ { "Right Boost Mixer", "RINPUT2 Switch", "RINPUT2" },
+ { "Right Boost Mixer", "RINPUT3 Switch", "RINPUT3" },
+
+ { "Right Input Mixer", "Boost Switch", "Right Boost Mixer", },
+ { "Right Input Mixer", NULL, "RINPUT1", }, /* Really Boost Switch */
+ { "Right Input Mixer", NULL, "RINPUT2" },
+ { "Right Input Mixer", NULL, "LINPUT3" },
+
+ { "Left ADC", NULL, "Left Input Mixer" },
+ { "Right ADC", NULL, "Right Input Mixer" },
+
+ { "Left Output Mixer", "LINPUT3 Switch", "LINPUT3" },
+ { "Left Output Mixer", "Boost Bypass Switch", "Left Boost Mixer"} ,
+ { "Left Output Mixer", "PCM Playback Switch", "Left DAC" },
+
+ { "Right Output Mixer", "RINPUT3 Switch", "RINPUT3" },
+ { "Right Output Mixer", "Boost Bypass Switch", "Right Boost Mixer" } ,
+ { "Right Output Mixer", "PCM Playback Switch", "Right DAC" },
+
+ { "Mono Output Mixer", "Left Switch", "Left Output Mixer" },
+ { "Mono Output Mixer", "Right Switch", "Right Output Mixer" },
+
+ { "LOUT1 PGA", NULL, "Left Output Mixer" },
+ { "ROUT1 PGA", NULL, "Right Output Mixer" },
+
+ { "HP_L", NULL, "LOUT1 PGA" },
+ { "HP_R", NULL, "ROUT1 PGA" },
+
+ { "Left Speaker PGA", NULL, "Left Output Mixer" },
+ { "Right Speaker PGA", NULL, "Right Output Mixer" },
+
+ { "Left Speaker Output", NULL, "Left Speaker PGA" },
+ { "Right Speaker Output", NULL, "Right Speaker PGA" },
+
+ { "SPK_LN", NULL, "Left Speaker Output" },
+ { "SPK_LP", NULL, "Left Speaker Output" },
+ { "SPK_RN", NULL, "Right Speaker Output" },
+ { "SPK_RP", NULL, "Right Speaker Output" },
+
+ { "OUT3", NULL, "Mono Output Mixer", }
+};
+
+static int wm8960_add_widgets(struct snd_soc_codec *codec)
+{
+ snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets,
+ ARRAY_SIZE(wm8960_dapm_widgets));
+
+ snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
+}
+
+static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface |= 0x0040;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x0013;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* set iface */
+ wm8960_write(codec, WM8960_IFACE1, iface);
+ return 0;
+}
+
+static int wm8960_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ u16 iface = wm8960_read(codec, WM8960_IFACE1) & 0xfff3;
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0008;
+ break;
+ }
+
+ /* set iface */
+ wm8960_write(codec, WM8960_IFACE1, iface);
+ return 0;
+}
+
+static int wm8960_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8960_read(codec, WM8960_DACCTL1) & 0xfff7;
+
+ if (mute)
+ wm8960_write(codec, WM8960_DACCTL1, mute_reg | 0x8);
+ else
+ wm8960_write(codec, WM8960_DACCTL1, mute_reg);
+ return 0;
+}
+
+static int wm8960_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct wm8960_data *pdata = codec->dev->platform_data;
+ u16 reg;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ /* Set VMID to 2x50k */
+ reg = wm8960_read(codec, WM8960_POWER1);
+ reg &= ~0x180;
+ reg |= 0x80;
+ wm8960_write(codec, WM8960_POWER1, reg);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* Enable anti-pop features */
+ wm8960_write(codec, WM8960_APOP1,
+ WM8960_POBCTRL | WM8960_SOFT_ST |
+ WM8960_BUFDCOPEN | WM8960_BUFIOEN);
+
+ /* Discharge HP output */
+ reg = WM8960_DISOP;
+ if (pdata)
+ reg |= pdata->dres << 4;
+ wm8960_write(codec, WM8960_APOP2, reg);
+
+ msleep(400);
+
+ wm8960_write(codec, WM8960_APOP2, 0);
+
+ /* Enable & ramp VMID at 2x50k */
+ reg = wm8960_read(codec, WM8960_POWER1);
+ reg |= 0x80;
+ wm8960_write(codec, WM8960_POWER1, reg);
+ msleep(100);
+
+ /* Enable VREF */
+ wm8960_write(codec, WM8960_POWER1, reg | WM8960_VREF);
+
+ /* Disable anti-pop features */
+ wm8960_write(codec, WM8960_APOP1, WM8960_BUFIOEN);
+ }
+
+ /* Set VMID to 2x250k */
+ reg = wm8960_read(codec, WM8960_POWER1);
+ reg &= ~0x180;
+ reg |= 0x100;
+ wm8960_write(codec, WM8960_POWER1, reg);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ /* Enable anti-pop features */
+ wm8960_write(codec, WM8960_APOP1,
+ WM8960_POBCTRL | WM8960_SOFT_ST |
+ WM8960_BUFDCOPEN | WM8960_BUFIOEN);
+
+ /* Disable VMID and VREF, let them discharge */
+ wm8960_write(codec, WM8960_POWER1, 0);
+ msleep(600);
+
+ wm8960_write(codec, WM8960_APOP1, 0);
+ break;
+ }
+
+ codec->bias_level = level;
+
+ return 0;
+}
+
+/* PLL divisors */
+struct _pll_div {
+ u32 pre_div: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 << 24) * 10)
+
+static int pll_factors(unsigned int source, unsigned int target,
+ struct _pll_div *pll_div)
+{
+ unsigned long long Kpart;
+ unsigned int K, Ndiv, Nmod;
+
+ pr_debug("WM8960 PLL: setting %dHz->%dHz\n", source, target);
+
+ /* Scale up target to PLL operating frequency */
+ target *= 4;
+
+ Ndiv = target / source;
+ if (Ndiv < 6) {
+ source >>= 1;
+ pll_div->pre_div = 1;
+ Ndiv = target / source;
+ } else
+ pll_div->pre_div = 0;
+
+ if ((Ndiv < 6) || (Ndiv > 12)) {
+ pr_err("WM8960 PLL: Unsupported N=%d\n", Ndiv);
+ return -EINVAL;
+ }
+
+ 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;
+
+ pr_debug("WM8960 PLL: N=%x K=%x pre_div=%d\n",
+ pll_div->n, pll_div->k, pll_div->pre_div);
+
+ return 0;
+}
+
+static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai,
+ int pll_id, unsigned int freq_in, unsigned int freq_out)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+ static struct _pll_div pll_div;
+ int ret;
+
+ if (freq_in && freq_out) {
+ ret = pll_factors(freq_in, freq_out, &pll_div);
+ if (ret != 0)
+ return ret;
+ }
+
+ /* Disable the PLL: even if we are changing the frequency the
+ * PLL needs to be disabled while we do so. */
+ wm8960_write(codec, WM8960_CLOCK1,
+ wm8960_read(codec, WM8960_CLOCK1) & ~1);
+ wm8960_write(codec, WM8960_POWER2,
+ wm8960_read(codec, WM8960_POWER2) & ~1);
+
+ if (!freq_in || !freq_out)
+ return 0;
+
+ reg = wm8960_read(codec, WM8960_PLL1) & ~0x3f;
+ reg |= pll_div.pre_div << 4;
+ reg |= pll_div.n;
+
+ if (pll_div.k) {
+ reg |= 0x20;
+
+ wm8960_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f);
+ wm8960_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff);
+ wm8960_write(codec, WM8960_PLL4, pll_div.k & 0x1ff);
+ }
+ wm8960_write(codec, WM8960_PLL1, reg);
+
+ /* Turn it on */
+ wm8960_write(codec, WM8960_POWER2,
+ wm8960_read(codec, WM8960_POWER2) | 1);
+ msleep(250);
+ wm8960_write(codec, WM8960_CLOCK1,
+ wm8960_read(codec, WM8960_CLOCK1) | 1);
+
+ return 0;
+}
+
+static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+ int div_id, int div)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 reg;
+
+ switch (div_id) {
+ case WM8960_SYSCLKSEL:
+ reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1fe;
+ wm8960_write(codec, WM8960_CLOCK1, reg | div);
+ break;
+ case WM8960_SYSCLKDIV:
+ reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1f9;
+ wm8960_write(codec, WM8960_CLOCK1, reg | div);
+ break;
+ case WM8960_DACDIV:
+ reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1c7;
+ wm8960_write(codec, WM8960_CLOCK1, reg | div);
+ break;
+ case WM8960_OPCLKDIV:
+ reg = wm8960_read(codec, WM8960_PLL1) & 0x03f;
+ wm8960_write(codec, WM8960_PLL1, reg | div);
+ break;
+ case WM8960_DCLKDIV:
+ reg = wm8960_read(codec, WM8960_CLOCK2) & 0x03f;
+ wm8960_write(codec, WM8960_CLOCK2, reg | div);
+ break;
+ case WM8960_TOCLKSEL:
+ reg = wm8960_read(codec, WM8960_ADDCTL1) & 0x1fd;
+ wm8960_write(codec, WM8960_ADDCTL1, reg | div);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+#define WM8960_RATES SNDRV_PCM_RATE_8000_48000
+
+#define WM8960_FORMATS \
+ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8960_dai_ops = {
+ .hw_params = wm8960_hw_params,
+ .digital_mute = wm8960_mute,
+ .set_fmt = wm8960_set_dai_fmt,
+ .set_clkdiv = wm8960_set_dai_clkdiv,
+ .set_pll = wm8960_set_dai_pll,
+};
+
+struct snd_soc_dai wm8960_dai = {
+ .name = "WM8960",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8960_RATES,
+ .formats = WM8960_FORMATS,},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8960_RATES,
+ .formats = WM8960_FORMATS,},
+ .ops = &wm8960_dai_ops,
+ .symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8960_dai);
+
+static int wm8960_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ wm8960_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8960_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < ARRAY_SIZE(wm8960_reg); i++) {
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+
+ wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+ wm8960_set_bias_level(codec, codec->suspend_bias_level);
+ return 0;
+}
+
+static struct snd_soc_codec *wm8960_codec;
+
+static int wm8960_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (wm8960_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = wm8960_codec;
+ codec = wm8960_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ snd_soc_add_controls(codec, wm8960_snd_controls,
+ ARRAY_SIZE(wm8960_snd_controls));
+ wm8960_add_widgets(codec);
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+/* power down chip */
+static int wm8960_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8960 = {
+ .probe = wm8960_probe,
+ .remove = wm8960_remove,
+ .suspend = wm8960_suspend,
+ .resume = wm8960_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8960);
+
+static int wm8960_register(struct wm8960_priv *wm8960)
+{
+ struct wm8960_data *pdata = wm8960->codec.dev->platform_data;
+ struct snd_soc_codec *codec = &wm8960->codec;
+ int ret;
+ u16 reg;
+
+ if (wm8960_codec) {
+ dev_err(codec->dev, "Another WM8960 is registered\n");
+ return -EINVAL;
+ }
+
+ if (!pdata) {
+ dev_warn(codec->dev, "No platform data supplied\n");
+ } else {
+ if (pdata->dres > WM8960_DRES_MAX) {
+ dev_err(codec->dev, "Invalid DRES: %d\n", pdata->dres);
+ pdata->dres = 0;
+ }
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = wm8960;
+ codec->name = "WM8960";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8960_read_reg_cache;
+ codec->write = wm8960_write;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8960_set_bias_level;
+ codec->dai = &wm8960_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = WM8960_CACHEREGNUM;
+ codec->reg_cache = &wm8960->reg_cache;
+
+ memcpy(codec->reg_cache, wm8960_reg, sizeof(wm8960_reg));
+
+ ret = wm8960_reset(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset\n");
+ return ret;
+ }
+
+ wm8960_dai.dev = codec->dev;
+
+ wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ /* Latch the update bits */
+ reg = wm8960_read(codec, WM8960_LINVOL);
+ wm8960_write(codec, WM8960_LINVOL, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_RINVOL);
+ wm8960_write(codec, WM8960_RINVOL, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_LADC);
+ wm8960_write(codec, WM8960_LADC, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_RADC);
+ wm8960_write(codec, WM8960_RADC, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_LDAC);
+ wm8960_write(codec, WM8960_LDAC, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_RDAC);
+ wm8960_write(codec, WM8960_RDAC, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_LOUT1);
+ wm8960_write(codec, WM8960_LOUT1, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_ROUT1);
+ wm8960_write(codec, WM8960_ROUT1, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_LOUT2);
+ wm8960_write(codec, WM8960_LOUT2, reg | 0x100);
+ reg = wm8960_read(codec, WM8960_ROUT2);
+ wm8960_write(codec, WM8960_ROUT2, reg | 0x100);
+
+ wm8960_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_dai(&wm8960_dai);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ snd_soc_unregister_codec(codec);
+ return ret;
+ }
+
+ return 0;
+}
+
+static void wm8960_unregister(struct wm8960_priv *wm8960)
+{
+ wm8960_set_bias_level(&wm8960->codec, SND_SOC_BIAS_OFF);
+ snd_soc_unregister_dai(&wm8960_dai);
+ snd_soc_unregister_codec(&wm8960->codec);
+ kfree(wm8960);
+ wm8960_codec = NULL;
+}
+
+static __devinit int wm8960_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct wm8960_priv *wm8960;
+ struct snd_soc_codec *codec;
+
+ wm8960 = kzalloc(sizeof(struct wm8960_priv), GFP_KERNEL);
+ if (wm8960 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8960->codec;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+
+ i2c_set_clientdata(i2c, wm8960);
+ codec->control_data = i2c;
+
+ codec->dev = &i2c->dev;
+
+ return wm8960_register(wm8960);
+}
+
+static __devexit int wm8960_i2c_remove(struct i2c_client *client)
+{
+ struct wm8960_priv *wm8960 = i2c_get_clientdata(client);
+ wm8960_unregister(wm8960);
+ return 0;
+}
+
+static const struct i2c_device_id wm8960_i2c_id[] = {
+ { "wm8960", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id);
+
+static struct i2c_driver wm8960_i2c_driver = {
+ .driver = {
+ .name = "WM8960 I2C Codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8960_i2c_probe,
+ .remove = __devexit_p(wm8960_i2c_remove),
+ .id_table = wm8960_i2c_id,
+};
+
+static int __init wm8960_modinit(void)
+{
+ int ret;
+
+ ret = i2c_add_driver(&wm8960_i2c_driver);
+ if (ret != 0) {
+ printk(KERN_ERR "Failed to register WM8960 I2C driver: %d\n",
+ ret);
+ }
+
+ return ret;
+}
+module_init(wm8960_modinit);
+
+static void __exit wm8960_exit(void)
+{
+ i2c_del_driver(&wm8960_i2c_driver);
+}
+module_exit(wm8960_exit);
+
+
+MODULE_DESCRIPTION("ASoC WM8960 driver");
+MODULE_AUTHOR("Liam Girdwood");
+MODULE_LICENSE("GPL");
--- /dev/null
+/*
+ * wm8960.h -- WM8960 Soc Audio driver
+ *
+ * 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 _WM8960_H
+#define _WM8960_H
+
+/* WM8960 register space */
+
+
+#define WM8960_CACHEREGNUM 56
+
+#define WM8960_LINVOL 0x0
+#define WM8960_RINVOL 0x1
+#define WM8960_LOUT1 0x2
+#define WM8960_ROUT1 0x3
+#define WM8960_CLOCK1 0x4
+#define WM8960_DACCTL1 0x5
+#define WM8960_DACCTL2 0x6
+#define WM8960_IFACE1 0x7
+#define WM8960_CLOCK2 0x8
+#define WM8960_IFACE2 0x9
+#define WM8960_LDAC 0xa
+#define WM8960_RDAC 0xb
+
+#define WM8960_RESET 0xf
+#define WM8960_3D 0x10
+#define WM8960_ALC1 0x11
+#define WM8960_ALC2 0x12
+#define WM8960_ALC3 0x13
+#define WM8960_NOISEG 0x14
+#define WM8960_LADC 0x15
+#define WM8960_RADC 0x16
+#define WM8960_ADDCTL1 0x17
+#define WM8960_ADDCTL2 0x18
+#define WM8960_POWER1 0x19
+#define WM8960_POWER2 0x1a
+#define WM8960_ADDCTL3 0x1b
+#define WM8960_APOP1 0x1c
+#define WM8960_APOP2 0x1d
+
+#define WM8960_LINPATH 0x20
+#define WM8960_RINPATH 0x21
+#define WM8960_LOUTMIX 0x22
+
+#define WM8960_ROUTMIX 0x25
+#define WM8960_MONOMIX1 0x26
+#define WM8960_MONOMIX2 0x27
+#define WM8960_LOUT2 0x28
+#define WM8960_ROUT2 0x29
+#define WM8960_MONO 0x2a
+#define WM8960_INBMIX1 0x2b
+#define WM8960_INBMIX2 0x2c
+#define WM8960_BYPASS1 0x2d
+#define WM8960_BYPASS2 0x2e
+#define WM8960_POWER3 0x2f
+#define WM8960_ADDCTL4 0x30
+#define WM8960_CLASSD1 0x31
+
+#define WM8960_CLASSD3 0x33
+#define WM8960_PLL1 0x34
+#define WM8960_PLL2 0x35
+#define WM8960_PLL3 0x36
+#define WM8960_PLL4 0x37
+
+
+/*
+ * WM8960 Clock dividers
+ */
+#define WM8960_SYSCLKDIV 0
+#define WM8960_DACDIV 1
+#define WM8960_OPCLKDIV 2
+#define WM8960_DCLKDIV 3
+#define WM8960_TOCLKSEL 4
+#define WM8960_SYSCLKSEL 5
+
+#define WM8960_SYSCLK_DIV_1 (0 << 1)
+#define WM8960_SYSCLK_DIV_2 (2 << 1)
+
+#define WM8960_SYSCLK_MCLK (0 << 0)
+#define WM8960_SYSCLK_PLL (1 << 0)
+
+#define WM8960_DAC_DIV_1 (0 << 3)
+#define WM8960_DAC_DIV_1_5 (1 << 3)
+#define WM8960_DAC_DIV_2 (2 << 3)
+#define WM8960_DAC_DIV_3 (3 << 3)
+#define WM8960_DAC_DIV_4 (4 << 3)
+#define WM8960_DAC_DIV_5_5 (5 << 3)
+#define WM8960_DAC_DIV_6 (6 << 3)
+
+#define WM8960_DCLK_DIV_1_5 (0 << 6)
+#define WM8960_DCLK_DIV_2 (1 << 6)
+#define WM8960_DCLK_DIV_3 (2 << 6)
+#define WM8960_DCLK_DIV_4 (3 << 6)
+#define WM8960_DCLK_DIV_6 (4 << 6)
+#define WM8960_DCLK_DIV_8 (5 << 6)
+#define WM8960_DCLK_DIV_12 (6 << 6)
+#define WM8960_DCLK_DIV_16 (7 << 6)
+
+#define WM8960_TOCLK_F19 (0 << 1)
+#define WM8960_TOCLK_F21 (1 << 1)
+
+#define WM8960_OPCLK_DIV_1 (0 << 0)
+#define WM8960_OPCLK_DIV_2 (1 << 0)
+#define WM8960_OPCLK_DIV_3 (2 << 0)
+#define WM8960_OPCLK_DIV_4 (3 << 0)
+#define WM8960_OPCLK_DIV_5_5 (4 << 0)
+#define WM8960_OPCLK_DIV_6 (5 << 0)
+
+extern struct snd_soc_dai wm8960_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8960;
+
+#define WM8960_DRES_400R 0
+#define WM8960_DRES_200R 1
+#define WM8960_DRES_600R 2
+#define WM8960_DRES_150R 3
+#define WM8960_DRES_MAX 3
+
+struct wm8960_data {
+ int dres;
+};
+
+#endif
--- /dev/null
+/*
+ * wm8988.c -- WM8988 ALSA SoC audio driver
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wm8988.h"
+
+/*
+ * wm8988 register cache
+ * We can't read the WM8988 register space when we
+ * are using 2 wire for device control, so we cache them instead.
+ */
+static const u16 wm8988_reg[] = {
+ 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */
+ 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */
+ 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */
+ 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */
+ 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */
+ 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */
+ 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */
+ 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */
+ 0x0079, 0x0079, 0x0079, /* 40 */
+};
+
+/* codec private data */
+struct wm8988_priv {
+ unsigned int sysclk;
+ struct snd_soc_codec codec;
+ struct snd_pcm_hw_constraint_list *sysclk_constraints;
+ u16 reg_cache[WM8988_NUM_REG];
+};
+
+
+/*
+ * read wm8988 register cache
+ */
+static inline unsigned int wm8988_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg > WM8988_NUM_REG)
+ return -1;
+ return cache[reg];
+}
+
+/*
+ * write wm8988 register cache
+ */
+static inline void wm8988_write_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg, unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ if (reg > WM8988_NUM_REG)
+ return;
+ cache[reg] = value;
+}
+
+static int wm8988_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u8 data[2];
+
+ /* data is
+ * D15..D9 WM8753 register offset
+ * D8...D0 register data
+ */
+ data[0] = (reg << 1) | ((value >> 8) & 0x0001);
+ data[1] = value & 0x00ff;
+
+ wm8988_write_reg_cache(codec, reg, value);
+ if (codec->hw_write(codec->control_data, data, 2) == 2)
+ return 0;
+ else
+ return -EIO;
+}
+
+#define wm8988_reset(c) wm8988_write(c, WM8988_RESET, 0)
+
+/*
+ * WM8988 Controls
+ */
+
+static const char *bass_boost_txt[] = {"Linear Control", "Adaptive Boost"};
+static const struct soc_enum bass_boost =
+ SOC_ENUM_SINGLE(WM8988_BASS, 7, 2, bass_boost_txt);
+
+static const char *bass_filter_txt[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" };
+static const struct soc_enum bass_filter =
+ SOC_ENUM_SINGLE(WM8988_BASS, 6, 2, bass_filter_txt);
+
+static const char *treble_txt[] = {"8kHz", "4kHz"};
+static const struct soc_enum treble =
+ SOC_ENUM_SINGLE(WM8988_TREBLE, 6, 2, treble_txt);
+
+static const char *stereo_3d_lc_txt[] = {"200Hz", "500Hz"};
+static const struct soc_enum stereo_3d_lc =
+ SOC_ENUM_SINGLE(WM8988_3D, 5, 2, stereo_3d_lc_txt);
+
+static const char *stereo_3d_uc_txt[] = {"2.2kHz", "1.5kHz"};
+static const struct soc_enum stereo_3d_uc =
+ SOC_ENUM_SINGLE(WM8988_3D, 6, 2, stereo_3d_uc_txt);
+
+static const char *stereo_3d_func_txt[] = {"Capture", "Playback"};
+static const struct soc_enum stereo_3d_func =
+ SOC_ENUM_SINGLE(WM8988_3D, 7, 2, stereo_3d_func_txt);
+
+static const char *alc_func_txt[] = {"Off", "Right", "Left", "Stereo"};
+static const struct soc_enum alc_func =
+ SOC_ENUM_SINGLE(WM8988_ALC1, 7, 4, alc_func_txt);
+
+static const char *ng_type_txt[] = {"Constant PGA Gain",
+ "Mute ADC Output"};
+static const struct soc_enum ng_type =
+ SOC_ENUM_SINGLE(WM8988_NGATE, 1, 2, ng_type_txt);
+
+static const char *deemph_txt[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+static const struct soc_enum deemph =
+ SOC_ENUM_SINGLE(WM8988_ADCDAC, 1, 4, deemph_txt);
+
+static const char *adcpol_txt[] = {"Normal", "L Invert", "R Invert",
+ "L + R Invert"};
+static const struct soc_enum adcpol =
+ SOC_ENUM_SINGLE(WM8988_ADCDAC, 5, 4, adcpol_txt);
+
+static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
+static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0);
+
+static const struct snd_kcontrol_new wm8988_snd_controls[] = {
+
+SOC_ENUM("Bass Boost", bass_boost),
+SOC_ENUM("Bass Filter", bass_filter),
+SOC_SINGLE("Bass Volume", WM8988_BASS, 0, 15, 1),
+
+SOC_SINGLE("Treble Volume", WM8988_TREBLE, 0, 15, 0),
+SOC_ENUM("Treble Cut-off", treble),
+
+SOC_SINGLE("3D Switch", WM8988_3D, 0, 1, 0),
+SOC_SINGLE("3D Volume", WM8988_3D, 1, 15, 0),
+SOC_ENUM("3D Lower Cut-off", stereo_3d_lc),
+SOC_ENUM("3D Upper Cut-off", stereo_3d_uc),
+SOC_ENUM("3D Mode", stereo_3d_func),
+
+SOC_SINGLE("ALC Capture Target Volume", WM8988_ALC1, 0, 7, 0),
+SOC_SINGLE("ALC Capture Max Volume", WM8988_ALC1, 4, 7, 0),
+SOC_ENUM("ALC Capture Function", alc_func),
+SOC_SINGLE("ALC Capture ZC Switch", WM8988_ALC2, 7, 1, 0),
+SOC_SINGLE("ALC Capture Hold Time", WM8988_ALC2, 0, 15, 0),
+SOC_SINGLE("ALC Capture Decay Time", WM8988_ALC3, 4, 15, 0),
+SOC_SINGLE("ALC Capture Attack Time", WM8988_ALC3, 0, 15, 0),
+SOC_SINGLE("ALC Capture NG Threshold", WM8988_NGATE, 3, 31, 0),
+SOC_ENUM("ALC Capture NG Type", ng_type),
+SOC_SINGLE("ALC Capture NG Switch", WM8988_NGATE, 0, 1, 0),
+
+SOC_SINGLE("ZC Timeout Switch", WM8988_ADCTL1, 0, 1, 0),
+
+SOC_DOUBLE_R_TLV("Capture Digital Volume", WM8988_LADC, WM8988_RADC,
+ 0, 255, 0, adc_tlv),
+SOC_DOUBLE_R_TLV("Capture Volume", WM8988_LINVOL, WM8988_RINVOL,
+ 0, 63, 0, pga_tlv),
+SOC_DOUBLE_R("Capture ZC Switch", WM8988_LINVOL, WM8988_RINVOL, 6, 1, 0),
+SOC_DOUBLE_R("Capture Switch", WM8988_LINVOL, WM8988_RINVOL, 7, 1, 1),
+
+SOC_ENUM("Playback De-emphasis", deemph),
+
+SOC_ENUM("Capture Polarity", adcpol),
+SOC_SINGLE("Playback 6dB Attenuate", WM8988_ADCDAC, 7, 1, 0),
+SOC_SINGLE("Capture 6dB Attenuate", WM8988_ADCDAC, 8, 1, 0),
+
+SOC_DOUBLE_R_TLV("PCM Volume", WM8988_LDAC, WM8988_RDAC, 0, 255, 0, dac_tlv),
+
+SOC_SINGLE_TLV("Left Mixer Left Bypass Volume", WM8988_LOUTM1, 4, 7, 1,
+ bypass_tlv),
+SOC_SINGLE_TLV("Left Mixer Right Bypass Volume", WM8988_LOUTM2, 4, 7, 1,
+ bypass_tlv),
+SOC_SINGLE_TLV("Right Mixer Left Bypass Volume", WM8988_ROUTM1, 4, 7, 1,
+ bypass_tlv),
+SOC_SINGLE_TLV("Right Mixer Right Bypass Volume", WM8988_ROUTM2, 4, 7, 1,
+ bypass_tlv),
+
+SOC_DOUBLE_R("Output 1 Playback ZC Switch", WM8988_LOUT1V,
+ WM8988_ROUT1V, 7, 1, 0),
+SOC_DOUBLE_R_TLV("Output 1 Playback Volume", WM8988_LOUT1V, WM8988_ROUT1V,
+ 0, 127, 0, out_tlv),
+
+SOC_DOUBLE_R("Output 2 Playback ZC Switch", WM8988_LOUT2V,
+ WM8988_ROUT2V, 7, 1, 0),
+SOC_DOUBLE_R_TLV("Output 2 Playback Volume", WM8988_LOUT2V, WM8988_ROUT2V,
+ 0, 127, 0, out_tlv),
+
+};
+
+/*
+ * DAPM Controls
+ */
+
+static int wm8988_lrc_control(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ u16 adctl2 = wm8988_read_reg_cache(codec, WM8988_ADCTL2);
+
+ /* Use the DAC to gate LRC if active, otherwise use ADC */
+ if (wm8988_read_reg_cache(codec, WM8988_PWR2) & 0x180)
+ adctl2 &= ~0x4;
+ else
+ adctl2 |= 0x4;
+
+ return wm8988_write(codec, WM8988_ADCTL2, adctl2);
+}
+
+static const char *wm8988_line_texts[] = {
+ "Line 1", "Line 2", "PGA", "Differential"};
+
+static const unsigned int wm8988_line_values[] = {
+ 0, 1, 3, 4};
+
+static const struct soc_enum wm8988_lline_enum =
+ SOC_VALUE_ENUM_SINGLE(WM8988_LOUTM1, 0, 7,
+ ARRAY_SIZE(wm8988_line_texts),
+ wm8988_line_texts,
+ wm8988_line_values);
+static const struct snd_kcontrol_new wm8988_left_line_controls =
+ SOC_DAPM_VALUE_ENUM("Route", wm8988_lline_enum);
+
+static const struct soc_enum wm8988_rline_enum =
+ SOC_VALUE_ENUM_SINGLE(WM8988_ROUTM1, 0, 7,
+ ARRAY_SIZE(wm8988_line_texts),
+ wm8988_line_texts,
+ wm8988_line_values);
+static const struct snd_kcontrol_new wm8988_right_line_controls =
+ SOC_DAPM_VALUE_ENUM("Route", wm8988_lline_enum);
+
+/* Left Mixer */
+static const struct snd_kcontrol_new wm8988_left_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Playback Switch", WM8988_LOUTM1, 8, 1, 0),
+ SOC_DAPM_SINGLE("Left Bypass Switch", WM8988_LOUTM1, 7, 1, 0),
+ SOC_DAPM_SINGLE("Right Playback Switch", WM8988_LOUTM2, 8, 1, 0),
+ SOC_DAPM_SINGLE("Right Bypass Switch", WM8988_LOUTM2, 7, 1, 0),
+};
+
+/* Right Mixer */
+static const struct snd_kcontrol_new wm8988_right_mixer_controls[] = {
+ SOC_DAPM_SINGLE("Left Playback Switch", WM8988_ROUTM1, 8, 1, 0),
+ SOC_DAPM_SINGLE("Left Bypass Switch", WM8988_ROUTM1, 7, 1, 0),
+ SOC_DAPM_SINGLE("Playback Switch", WM8988_ROUTM2, 8, 1, 0),
+ SOC_DAPM_SINGLE("Right Bypass Switch", WM8988_ROUTM2, 7, 1, 0),
+};
+
+static const char *wm8988_pga_sel[] = {"Line 1", "Line 2", "Differential"};
+static const unsigned int wm8988_pga_val[] = { 0, 1, 3 };
+
+/* Left PGA Mux */
+static const struct soc_enum wm8988_lpga_enum =
+ SOC_VALUE_ENUM_SINGLE(WM8988_LADCIN, 6, 3,
+ ARRAY_SIZE(wm8988_pga_sel),
+ wm8988_pga_sel,
+ wm8988_pga_val);
+static const struct snd_kcontrol_new wm8988_left_pga_controls =
+ SOC_DAPM_VALUE_ENUM("Route", wm8988_lpga_enum);
+
+/* Right PGA Mux */
+static const struct soc_enum wm8988_rpga_enum =
+ SOC_VALUE_ENUM_SINGLE(WM8988_RADCIN, 6, 3,
+ ARRAY_SIZE(wm8988_pga_sel),
+ wm8988_pga_sel,
+ wm8988_pga_val);
+static const struct snd_kcontrol_new wm8988_right_pga_controls =
+ SOC_DAPM_VALUE_ENUM("Route", wm8988_rpga_enum);
+
+/* Differential Mux */
+static const char *wm8988_diff_sel[] = {"Line 1", "Line 2"};
+static const struct soc_enum diffmux =
+ SOC_ENUM_SINGLE(WM8988_ADCIN, 8, 2, wm8988_diff_sel);
+static const struct snd_kcontrol_new wm8988_diffmux_controls =
+ SOC_DAPM_ENUM("Route", diffmux);
+
+/* Mono ADC Mux */
+static const char *wm8988_mono_mux[] = {"Stereo", "Mono (Left)",
+ "Mono (Right)", "Digital Mono"};
+static const struct soc_enum monomux =
+ SOC_ENUM_SINGLE(WM8988_ADCIN, 6, 4, wm8988_mono_mux);
+static const struct snd_kcontrol_new wm8988_monomux_controls =
+ SOC_DAPM_ENUM("Route", monomux);
+
+static const struct snd_soc_dapm_widget wm8988_dapm_widgets[] = {
+ SND_SOC_DAPM_MICBIAS("Mic Bias", WM8988_PWR1, 1, 0),
+
+ SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0,
+ &wm8988_diffmux_controls),
+ SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0,
+ &wm8988_monomux_controls),
+ SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0,
+ &wm8988_monomux_controls),
+
+ SND_SOC_DAPM_MUX("Left PGA Mux", WM8988_PWR1, 5, 0,
+ &wm8988_left_pga_controls),
+ SND_SOC_DAPM_MUX("Right PGA Mux", WM8988_PWR1, 4, 0,
+ &wm8988_right_pga_controls),
+
+ SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0,
+ &wm8988_left_line_controls),
+ SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0,
+ &wm8988_right_line_controls),
+
+ SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8988_PWR1, 2, 0),
+ SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8988_PWR1, 3, 0),
+
+ SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8988_PWR2, 7, 0),
+ SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8988_PWR2, 8, 0),
+
+ SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0,
+ &wm8988_left_mixer_controls[0],
+ ARRAY_SIZE(wm8988_left_mixer_controls)),
+ SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0,
+ &wm8988_right_mixer_controls[0],
+ ARRAY_SIZE(wm8988_right_mixer_controls)),
+
+ SND_SOC_DAPM_PGA("Right Out 2", WM8988_PWR2, 3, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Left Out 2", WM8988_PWR2, 4, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Right Out 1", WM8988_PWR2, 5, 0, NULL, 0),
+ SND_SOC_DAPM_PGA("Left Out 1", WM8988_PWR2, 6, 0, NULL, 0),
+
+ SND_SOC_DAPM_POST("LRC control", wm8988_lrc_control),
+
+ SND_SOC_DAPM_OUTPUT("LOUT1"),
+ SND_SOC_DAPM_OUTPUT("ROUT1"),
+ SND_SOC_DAPM_OUTPUT("LOUT2"),
+ SND_SOC_DAPM_OUTPUT("ROUT2"),
+ SND_SOC_DAPM_OUTPUT("VREF"),
+
+ SND_SOC_DAPM_INPUT("LINPUT1"),
+ SND_SOC_DAPM_INPUT("LINPUT2"),
+ SND_SOC_DAPM_INPUT("RINPUT1"),
+ SND_SOC_DAPM_INPUT("RINPUT2"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+
+ { "Left Line Mux", "Line 1", "LINPUT1" },
+ { "Left Line Mux", "Line 2", "LINPUT2" },
+ { "Left Line Mux", "PGA", "Left PGA Mux" },
+ { "Left Line Mux", "Differential", "Differential Mux" },
+
+ { "Right Line Mux", "Line 1", "RINPUT1" },
+ { "Right Line Mux", "Line 2", "RINPUT2" },
+ { "Right Line Mux", "PGA", "Right PGA Mux" },
+ { "Right Line Mux", "Differential", "Differential Mux" },
+
+ { "Left PGA Mux", "Line 1", "LINPUT1" },
+ { "Left PGA Mux", "Line 2", "LINPUT2" },
+ { "Left PGA Mux", "Differential", "Differential Mux" },
+
+ { "Right PGA Mux", "Line 1", "RINPUT1" },
+ { "Right PGA Mux", "Line 2", "RINPUT2" },
+ { "Right PGA Mux", "Differential", "Differential Mux" },
+
+ { "Differential Mux", "Line 1", "LINPUT1" },
+ { "Differential Mux", "Line 1", "RINPUT1" },
+ { "Differential Mux", "Line 2", "LINPUT2" },
+ { "Differential Mux", "Line 2", "RINPUT2" },
+
+ { "Left ADC Mux", "Stereo", "Left PGA Mux" },
+ { "Left ADC Mux", "Mono (Left)", "Left PGA Mux" },
+ { "Left ADC Mux", "Digital Mono", "Left PGA Mux" },
+
+ { "Right ADC Mux", "Stereo", "Right PGA Mux" },
+ { "Right ADC Mux", "Mono (Right)", "Right PGA Mux" },
+ { "Right ADC Mux", "Digital Mono", "Right PGA Mux" },
+
+ { "Left ADC", NULL, "Left ADC Mux" },
+ { "Right ADC", NULL, "Right ADC Mux" },
+
+ { "Left Line Mux", "Line 1", "LINPUT1" },
+ { "Left Line Mux", "Line 2", "LINPUT2" },
+ { "Left Line Mux", "PGA", "Left PGA Mux" },
+ { "Left Line Mux", "Differential", "Differential Mux" },
+
+ { "Right Line Mux", "Line 1", "RINPUT1" },
+ { "Right Line Mux", "Line 2", "RINPUT2" },
+ { "Right Line Mux", "PGA", "Right PGA Mux" },
+ { "Right Line Mux", "Differential", "Differential Mux" },
+
+ { "Left Mixer", "Playback Switch", "Left DAC" },
+ { "Left Mixer", "Left Bypass Switch", "Left Line Mux" },
+ { "Left Mixer", "Right Playback Switch", "Right DAC" },
+ { "Left Mixer", "Right Bypass Switch", "Right Line Mux" },
+
+ { "Right Mixer", "Left Playback Switch", "Left DAC" },
+ { "Right Mixer", "Left Bypass Switch", "Left Line Mux" },
+ { "Right Mixer", "Playback Switch", "Right DAC" },
+ { "Right Mixer", "Right Bypass Switch", "Right Line Mux" },
+
+ { "Left Out 1", NULL, "Left Mixer" },
+ { "LOUT1", NULL, "Left Out 1" },
+ { "Right Out 1", NULL, "Right Mixer" },
+ { "ROUT1", NULL, "Right Out 1" },
+
+ { "Left Out 2", NULL, "Left Mixer" },
+ { "LOUT2", NULL, "Left Out 2" },
+ { "Right Out 2", NULL, "Right Mixer" },
+ { "ROUT2", NULL, "Right Out 2" },
+};
+
+struct _coeff_div {
+ u32 mclk;
+ u32 rate;
+ u16 fs;
+ u8 sr:5;
+ u8 usb:1;
+};
+
+/* codec hifi mclk clock divider coefficients */
+static const struct _coeff_div coeff_div[] = {
+ /* 8k */
+ {12288000, 8000, 1536, 0x6, 0x0},
+ {11289600, 8000, 1408, 0x16, 0x0},
+ {18432000, 8000, 2304, 0x7, 0x0},
+ {16934400, 8000, 2112, 0x17, 0x0},
+ {12000000, 8000, 1500, 0x6, 0x1},
+
+ /* 11.025k */
+ {11289600, 11025, 1024, 0x18, 0x0},
+ {16934400, 11025, 1536, 0x19, 0x0},
+ {12000000, 11025, 1088, 0x19, 0x1},
+
+ /* 16k */
+ {12288000, 16000, 768, 0xa, 0x0},
+ {18432000, 16000, 1152, 0xb, 0x0},
+ {12000000, 16000, 750, 0xa, 0x1},
+
+ /* 22.05k */
+ {11289600, 22050, 512, 0x1a, 0x0},
+ {16934400, 22050, 768, 0x1b, 0x0},
+ {12000000, 22050, 544, 0x1b, 0x1},
+
+ /* 32k */
+ {12288000, 32000, 384, 0xc, 0x0},
+ {18432000, 32000, 576, 0xd, 0x0},
+ {12000000, 32000, 375, 0xa, 0x1},
+
+ /* 44.1k */
+ {11289600, 44100, 256, 0x10, 0x0},
+ {16934400, 44100, 384, 0x11, 0x0},
+ {12000000, 44100, 272, 0x11, 0x1},
+
+ /* 48k */
+ {12288000, 48000, 256, 0x0, 0x0},
+ {18432000, 48000, 384, 0x1, 0x0},
+ {12000000, 48000, 250, 0x0, 0x1},
+
+ /* 88.2k */
+ {11289600, 88200, 128, 0x1e, 0x0},
+ {16934400, 88200, 192, 0x1f, 0x0},
+ {12000000, 88200, 136, 0x1f, 0x1},
+
+ /* 96k */
+ {12288000, 96000, 128, 0xe, 0x0},
+ {18432000, 96000, 192, 0xf, 0x0},
+ {12000000, 96000, 125, 0xe, 0x1},
+};
+
+static inline int get_coeff(int mclk, int rate)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
+ if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
+ return i;
+ }
+
+ return -EINVAL;
+}
+
+/* The set of rates we can generate from the above for each SYSCLK */
+
+static unsigned int rates_12288[] = {
+ 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_12288 = {
+ .count = ARRAY_SIZE(rates_12288),
+ .list = rates_12288,
+};
+
+static unsigned int rates_112896[] = {
+ 8000, 11025, 22050, 44100,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_112896 = {
+ .count = ARRAY_SIZE(rates_112896),
+ .list = rates_112896,
+};
+
+static unsigned int rates_12[] = {
+ 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000,
+ 48000, 88235, 96000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_12 = {
+ .count = ARRAY_SIZE(rates_12),
+ .list = rates_12,
+};
+
+/*
+ * Note that this should be called from init rather than from hw_params.
+ */
+static int wm8988_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct wm8988_priv *wm8988 = codec->private_data;
+
+ switch (freq) {
+ case 11289600:
+ case 18432000:
+ case 22579200:
+ case 36864000:
+ wm8988->sysclk_constraints = &constraints_112896;
+ wm8988->sysclk = freq;
+ return 0;
+
+ case 12288000:
+ case 16934400:
+ case 24576000:
+ case 33868800:
+ wm8988->sysclk_constraints = &constraints_12288;
+ wm8988->sysclk = freq;
+ return 0;
+
+ case 12000000:
+ case 24000000:
+ wm8988->sysclk_constraints = &constraints_12;
+ wm8988->sysclk = freq;
+ return 0;
+ }
+ return -EINVAL;
+}
+
+static int wm8988_set_dai_fmt(struct snd_soc_dai *codec_dai,
+ unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u16 iface = 0;
+
+ /* set master/slave audio interface */
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ iface = 0x0040;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* interface format */
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= 0x0002;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ iface |= 0x0001;
+ break;
+ case SND_SOC_DAIFMT_DSP_A:
+ iface |= 0x0003;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ iface |= 0x0013;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* clock inversion */
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ iface |= 0x0090;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ iface |= 0x0080;
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ iface |= 0x0010;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ wm8988_write(codec, WM8988_IFACE, iface);
+ return 0;
+}
+
+static int wm8988_pcm_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct wm8988_priv *wm8988 = codec->private_data;
+
+ /* The set of sample rates that can be supported depends on the
+ * MCLK supplied to the CODEC - enforce this.
+ */
+ if (!wm8988->sysclk) {
+ dev_err(codec->dev,
+ "No MCLK configured, call set_sysclk() on init\n");
+ return -EINVAL;
+ }
+
+ snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ wm8988->sysclk_constraints);
+
+ return 0;
+}
+
+static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_device *socdev = rtd->socdev;
+ struct snd_soc_codec *codec = socdev->card->codec;
+ struct wm8988_priv *wm8988 = codec->private_data;
+ u16 iface = wm8988_read_reg_cache(codec, WM8988_IFACE) & 0x1f3;
+ u16 srate = wm8988_read_reg_cache(codec, WM8988_SRATE) & 0x180;
+ int coeff;
+
+ coeff = get_coeff(wm8988->sysclk, params_rate(params));
+ if (coeff < 0) {
+ coeff = get_coeff(wm8988->sysclk / 2, params_rate(params));
+ srate |= 0x40;
+ }
+ if (coeff < 0) {
+ dev_err(codec->dev,
+ "Unable to configure sample rate %dHz with %dHz MCLK\n",
+ params_rate(params), wm8988->sysclk);
+ return coeff;
+ }
+
+ /* bit size */
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ break;
+ case SNDRV_PCM_FORMAT_S20_3LE:
+ iface |= 0x0004;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ iface |= 0x0008;
+ break;
+ case SNDRV_PCM_FORMAT_S32_LE:
+ iface |= 0x000c;
+ break;
+ }
+
+ /* set iface & srate */
+ wm8988_write(codec, WM8988_IFACE, iface);
+ if (coeff >= 0)
+ wm8988_write(codec, WM8988_SRATE, srate |
+ (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb);
+
+ return 0;
+}
+
+static int wm8988_mute(struct snd_soc_dai *dai, int mute)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u16 mute_reg = wm8988_read_reg_cache(codec, WM8988_ADCDAC) & 0xfff7;
+
+ if (mute)
+ wm8988_write(codec, WM8988_ADCDAC, mute_reg | 0x8);
+ else
+ wm8988_write(codec, WM8988_ADCDAC, mute_reg);
+ return 0;
+}
+
+static int wm8988_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u16 pwr_reg = wm8988_read_reg_cache(codec, WM8988_PWR1) & ~0x1c1;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ break;
+
+ case SND_SOC_BIAS_PREPARE:
+ /* VREF, VMID=2x50k, digital enabled */
+ wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x00c0);
+ break;
+
+ case SND_SOC_BIAS_STANDBY:
+ if (codec->bias_level == SND_SOC_BIAS_OFF) {
+ /* VREF, VMID=2x5k */
+ wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x1c1);
+
+ /* Charge caps */
+ msleep(100);
+ }
+
+ /* VREF, VMID=2*500k, digital stopped */
+ wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x0141);
+ break;
+
+ case SND_SOC_BIAS_OFF:
+ wm8988_write(codec, WM8988_PWR1, 0x0000);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+#define WM8988_RATES SNDRV_PCM_RATE_8000_96000
+
+#define WM8988_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8988_ops = {
+ .startup = wm8988_pcm_startup,
+ .hw_params = wm8988_pcm_hw_params,
+ .set_fmt = wm8988_set_dai_fmt,
+ .set_sysclk = wm8988_set_dai_sysclk,
+ .digital_mute = wm8988_mute,
+};
+
+struct snd_soc_dai wm8988_dai = {
+ .name = "WM8988",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8988_RATES,
+ .formats = WM8988_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = WM8988_RATES,
+ .formats = WM8988_FORMATS,
+ },
+ .ops = &wm8988_ops,
+ .symmetric_rates = 1,
+};
+EXPORT_SYMBOL_GPL(wm8988_dai);
+
+static int wm8988_suspend(struct platform_device *pdev, pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int wm8988_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ int i;
+ u8 data[2];
+ u16 *cache = codec->reg_cache;
+
+ /* Sync reg_cache with the hardware */
+ for (i = 0; i < WM8988_NUM_REG; i++) {
+ if (i == WM8988_RESET)
+ continue;
+ data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
+ data[1] = cache[i] & 0x00ff;
+ codec->hw_write(codec->control_data, data, 2);
+ }
+
+ wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+
+static struct snd_soc_codec *wm8988_codec;
+
+static int wm8988_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ if (wm8988_codec == NULL) {
+ dev_err(&pdev->dev, "Codec device not registered\n");
+ return -ENODEV;
+ }
+
+ socdev->card->codec = wm8988_codec;
+ codec = wm8988_codec;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to create pcms: %d\n", ret);
+ goto pcm_err;
+ }
+
+ snd_soc_add_controls(codec, wm8988_snd_controls,
+ ARRAY_SIZE(wm8988_snd_controls));
+ snd_soc_dapm_new_controls(codec, wm8988_dapm_widgets,
+ ARRAY_SIZE(wm8988_dapm_widgets));
+ snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+ snd_soc_dapm_new_widgets(codec);
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0) {
+ dev_err(codec->dev, "failed to register card: %d\n", ret);
+ goto card_err;
+ }
+
+ return ret;
+
+card_err:
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+pcm_err:
+ return ret;
+}
+
+static int wm8988_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8988 = {
+ .probe = wm8988_probe,
+ .remove = wm8988_remove,
+ .suspend = wm8988_suspend,
+ .resume = wm8988_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8988);
+
+static int wm8988_register(struct wm8988_priv *wm8988)
+{
+ struct snd_soc_codec *codec = &wm8988->codec;
+ int ret;
+ u16 reg;
+
+ if (wm8988_codec) {
+ dev_err(codec->dev, "Another WM8988 is registered\n");
+ ret = -EINVAL;
+ goto err;
+ }
+
+ mutex_init(&codec->mutex);
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ codec->private_data = wm8988;
+ codec->name = "WM8988";
+ codec->owner = THIS_MODULE;
+ codec->read = wm8988_read_reg_cache;
+ codec->write = wm8988_write;
+ codec->dai = &wm8988_dai;
+ codec->num_dai = 1;
+ codec->reg_cache_size = ARRAY_SIZE(wm8988->reg_cache);
+ codec->reg_cache = &wm8988->reg_cache;
+ codec->bias_level = SND_SOC_BIAS_OFF;
+ codec->set_bias_level = wm8988_set_bias_level;
+
+ memcpy(codec->reg_cache, wm8988_reg,
+ sizeof(wm8988_reg));
+
+ ret = wm8988_reset(codec);
+ if (ret < 0) {
+ dev_err(codec->dev, "Failed to issue reset\n");
+ return ret;
+ }
+
+ /* set the update bits (we always update left then right) */
+ reg = wm8988_read_reg_cache(codec, WM8988_RADC);
+ wm8988_write(codec, WM8988_RADC, reg | 0x100);
+ reg = wm8988_read_reg_cache(codec, WM8988_RDAC);
+ wm8988_write(codec, WM8988_RDAC, reg | 0x0100);
+ reg = wm8988_read_reg_cache(codec, WM8988_ROUT1V);
+ wm8988_write(codec, WM8988_ROUT1V, reg | 0x0100);
+ reg = wm8988_read_reg_cache(codec, WM8988_ROUT2V);
+ wm8988_write(codec, WM8988_ROUT2V, reg | 0x0100);
+ reg = wm8988_read_reg_cache(codec, WM8988_RINVOL);
+ wm8988_write(codec, WM8988_RINVOL, reg | 0x0100);
+
+ wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_STANDBY);
+
+ wm8988_dai.dev = codec->dev;
+
+ wm8988_codec = codec;
+
+ ret = snd_soc_register_codec(codec);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_dai(&wm8988_dai);
+ if (ret != 0) {
+ dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+ snd_soc_unregister_codec(codec);
+ return ret;
+ }
+
+ return 0;
+
+err:
+ kfree(wm8988);
+ return ret;
+}
+
+static void wm8988_unregister(struct wm8988_priv *wm8988)
+{
+ wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_OFF);
+ snd_soc_unregister_dai(&wm8988_dai);
+ snd_soc_unregister_codec(&wm8988->codec);
+ kfree(wm8988);
+ wm8988_codec = NULL;
+}
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static int wm8988_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct wm8988_priv *wm8988;
+ struct snd_soc_codec *codec;
+
+ wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL);
+ if (wm8988 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8988->codec;
+ codec->hw_write = (hw_write_t)i2c_master_send;
+
+ i2c_set_clientdata(i2c, wm8988);
+ codec->control_data = i2c;
+
+ codec->dev = &i2c->dev;
+
+ return wm8988_register(wm8988);
+}
+
+static int wm8988_i2c_remove(struct i2c_client *client)
+{
+ struct wm8988_priv *wm8988 = i2c_get_clientdata(client);
+ wm8988_unregister(wm8988);
+ return 0;
+}
+
+static const struct i2c_device_id wm8988_i2c_id[] = {
+ { "wm8988", 0 },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, wm8988_i2c_id);
+
+static struct i2c_driver wm8988_i2c_driver = {
+ .driver = {
+ .name = "WM8988",
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8988_i2c_probe,
+ .remove = wm8988_i2c_remove,
+ .id_table = wm8988_i2c_id,
+};
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+static int wm8988_spi_write(struct spi_device *spi, const char *data, int len)
+{
+ struct spi_transfer t;
+ struct spi_message m;
+ u8 msg[2];
+
+ if (len <= 0)
+ return 0;
+
+ msg[0] = data[0];
+ msg[1] = data[1];
+
+ spi_message_init(&m);
+ memset(&t, 0, (sizeof t));
+
+ t.tx_buf = &msg[0];
+ t.len = len;
+
+ spi_message_add_tail(&t, &m);
+ spi_sync(spi, &m);
+
+ return len;
+}
+
+static int __devinit wm8988_spi_probe(struct spi_device *spi)
+{
+ struct wm8988_priv *wm8988;
+ struct snd_soc_codec *codec;
+
+ wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL);
+ if (wm8988 == NULL)
+ return -ENOMEM;
+
+ codec = &wm8988->codec;
+ codec->hw_write = (hw_write_t)wm8988_spi_write;
+ codec->control_data = spi;
+ codec->dev = &spi->dev;
+
+ spi->dev.driver_data = wm8988;
+
+ return wm8988_register(wm8988);
+}
+
+static int __devexit wm8988_spi_remove(struct spi_device *spi)
+{
+ struct wm8988_priv *wm8988 = spi->dev.driver_data;
+
+ wm8988_unregister(wm8988);
+
+ return 0;
+}
+
+static struct spi_driver wm8988_spi_driver = {
+ .driver = {
+ .name = "wm8988",
+ .bus = &spi_bus_type,
+ .owner = THIS_MODULE,
+ },
+ .probe = wm8988_spi_probe,
+ .remove = __devexit_p(wm8988_spi_remove),
+};
+#endif
+
+static int __init wm8988_modinit(void)
+{
+ int ret;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ ret = i2c_add_driver(&wm8988_i2c_driver);
+ if (ret != 0)
+ pr_err("WM8988: Unable to register I2C driver: %d\n", ret);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ ret = spi_register_driver(&wm8988_spi_driver);
+ if (ret != 0)
+ pr_err("WM8988: Unable to register SPI driver: %d\n", ret);
+#endif
+ return ret;
+}
+module_init(wm8988_modinit);
+
+static void __exit wm8988_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+ i2c_del_driver(&wm8988_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+ spi_unregister_driver(&wm8988_spi_driver);
+#endif
+}
+module_exit(wm8988_exit);
+
+
+MODULE_DESCRIPTION("ASoC WM8988 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
--- /dev/null
+/*
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * Author: Richard Purdie <richard@openedhand.com>
+ *
+ * Based on WM8753.h
+ *
+ * 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 _WM8988_H
+#define _WM8988_H
+
+/* WM8988 register space */
+
+#define WM8988_LINVOL 0x00
+#define WM8988_RINVOL 0x01
+#define WM8988_LOUT1V 0x02
+#define WM8988_ROUT1V 0x03
+#define WM8988_ADCDAC 0x05
+#define WM8988_IFACE 0x07
+#define WM8988_SRATE 0x08
+#define WM8988_LDAC 0x0a
+#define WM8988_RDAC 0x0b
+#define WM8988_BASS 0x0c
+#define WM8988_TREBLE 0x0d
+#define WM8988_RESET 0x0f
+#define WM8988_3D 0x10
+#define WM8988_ALC1 0x11
+#define WM8988_ALC2 0x12
+#define WM8988_ALC3 0x13
+#define WM8988_NGATE 0x14
+#define WM8988_LADC 0x15
+#define WM8988_RADC 0x16
+#define WM8988_ADCTL1 0x17
+#define WM8988_ADCTL2 0x18
+#define WM8988_PWR1 0x19
+#define WM8988_PWR2 0x1a
+#define WM8988_ADCTL3 0x1b
+#define WM8988_ADCIN 0x1f
+#define WM8988_LADCIN 0x20
+#define WM8988_RADCIN 0x21
+#define WM8988_LOUTM1 0x22
+#define WM8988_LOUTM2 0x23
+#define WM8988_ROUTM1 0x24
+#define WM8988_ROUTM2 0x25
+#define WM8988_LOUT2V 0x28
+#define WM8988_ROUT2V 0x29
+#define WM8988_LPPB 0x43
+#define WM8988_NUM_REG 0x44
+
+#define WM8988_SYSCLK 0
+
+extern struct snd_soc_dai wm8988_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8988;
+
+#endif
if ((Ndiv < 6) || (Ndiv > 12))
printk(KERN_WARNING
- "WM8990 N value outwith recommended range! N = %d\n", Ndiv);
+ "WM8990 N value outwith recommended range! N = %u\n", Ndiv);
pll_div->n = Ndiv;
Nmod = target % source;
--- /dev/null
+/*
+ * wm9081.c -- WM9081 ALSA SoC Audio driver
+ *
+ * Author: Mark Brown
+ *
+ * Copyright 2009 Wolfson Microelectronics plc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include <sound/wm9081.h>
+#include "wm9081.h"
+
+static u16 wm9081_reg_defaults[] = {
+ 0x0000, /* R0 - Software Reset */
+ 0x0000, /* R1 */
+ 0x00B9, /* R2 - Analogue Lineout */
+ 0x00B9, /* R3 - Analogue Speaker PGA */
+ 0x0001, /* R4 - VMID Control */
+ 0x0068, /* R5 - Bias Control 1 */
+ 0x0000, /* R6 */
+ 0x0000, /* R7 - Analogue Mixer */
+ 0x0000, /* R8 - Anti Pop Control */
+ 0x01DB, /* R9 - Analogue Speaker 1 */
+ 0x0018, /* R10 - Analogue Speaker 2 */
+ 0x0180, /* R11 - Power Management */
+ 0x0000, /* R12 - Clock Control 1 */
+ 0x0038, /* R13 - Clock Control 2 */
+ 0x4000, /* R14 - Clock Control 3 */
+ 0x0000, /* R15 */
+ 0x0000, /* R16 - FLL Control 1 */
+ 0x0200, /* R17 - FLL Control 2 */
+ 0x0000, /* R18 - FLL Control 3 */
+ 0x0204, /* R19 - FLL Control 4 */
+ 0x0000, /* R20 - FLL Control 5 */
+ 0x0000, /* R21 */
+ 0x0000, /* R22 - Audio Interface 1 */
+ 0x0002, /* R23 - Audio Interface 2 */
+ 0x0008, /* R24 - Audio Interface 3 */
+ 0x0022, /* R25 - Audio Interface 4 */
+ 0x0000, /* R26 - Interrupt Status */
+ 0x0006, /* R27 - Interrupt Status Mask */
+ 0x0000, /* R28 - Interrupt Polarity */
+ 0x0000, /* R29 - Interrupt Control */
+ 0x00C0, /* R30 - DAC Digital 1 */
+ 0x0008, /* R31 - DAC Digital 2 */
+ 0x09AF, /* R32 - DRC 1 */
+ 0x4201, /* R33 - DRC 2 */
+ 0x0000, /* R34 - DRC 3 */
+ 0x0000, /* R35 - DRC 4 */
+ 0x0000, /* R36 */
+ 0x0000, /* R37 */
+ 0x0000, /* R38 - Write Sequencer 1 */
+ 0x0000, /* R39 - Write Sequencer 2 */
+ 0x0002, /* R40 - MW Slave 1 */
+ 0x0000, /* R41 */
+ 0x0000, /* R42 - EQ 1 */
+ 0x0000, /* R43 - EQ 2 */
+ 0x0FCA, /* R44 - EQ 3 */
+ 0x0400, /* R45 - EQ 4 */
+ 0x00B8, /* R46 - EQ 5 */
+ 0x1EB5, /* R47 - EQ 6 */
+ 0xF145, /* R48 - EQ 7 */
+ 0x0B75, /* R49 - EQ 8 */
+ 0x01C5, /* R50 - EQ 9 */
+ 0x169E, /* R51 - EQ 10 */
+ 0xF829, /* R52 - EQ 11 */
+ 0x07AD, /* R53 - EQ 12 */
+ 0x1103, /* R54 - EQ 13 */
+ 0x1C58, /* R55 - EQ 14 */
+ 0xF373, /* R56 - EQ 15 */
+ 0x0A54, /* R57 - EQ 16 */
+ 0x0558, /* R58 - EQ 17 */
+ 0x0564, /* R59 - EQ 18 */
+ 0x0559, /* R60 - EQ 19 */
+ 0x4000, /* R61 - EQ 20 */
+};
+
+static struct {
+ int ratio;
+ int clk_sys_rate;
+} clk_sys_rates[] = {
+ { 64, 0 },
+ { 128, 1 },
+ { 192, 2 },
+ { 256, 3 },
+ { 384, 4 },
+ { 512, 5 },
+ { 768, 6 },
+ { 1024, 7 },
+ { 1408, 8 },
+ { 1536, 9 },
+};
+
+static struct {
+ int rate;
+ int sample_rate;
+} sample_rates[] = {
+ { 8000, 0 },
+ { 11025, 1 },
+ { 12000, 2 },
+ { 16000, 3 },
+ { 22050, 4 },
+ { 24000, 5 },
+ { 32000, 6 },
+ { 44100, 7 },
+ { 48000, 8 },
+ { 88200, 9 },
+ { 96000, 10 },
+};
+
+static struct {
+ int div; /* *10 due to .5s */
+ int bclk_div;
+} bclk_divs[] = {
+ { 10, 0 },
+ { 15, 1 },
+ { 20, 2 },
+ { 30, 3 },
+ { 40, 4 },
+ { 50, 5 },
+ { 55, 6 },
+ { 60, 7 },
+ { 80, 8 },
+ { 100, 9 },
+ { 110, 10 },
+ { 120, 11 },
+ { 160, 12 },
+ { 200, 13 },
+ { 220, 14 },
+ { 240, 15 },
+ { 250, 16 },
+ { 300, 17 },
+ { 320, 18 },
+ { 440, 19 },
+ { 480, 20 },
+};
+
+struct wm9081_priv {
+ struct snd_soc_codec codec;
+ u16 reg_cache[WM9081_MAX_REGISTER + 1];
+ int sysclk_source;
+ int mclk_rate;
+ int sysclk_rate;
+ int fs;
+ int bclk;
+ int master;
+ int fll_fref;
+ int fll_fout;
+ struct wm9081_retune_mobile_config *retune;
+};
+
+static int wm9081_reg_is_volatile(int reg)
+{
+ switch (reg) {
+ default:
+ return 0;
+ }
+}
+
+static unsigned int wm9081_read_reg_cache(struct snd_soc_codec *codec,
+ unsigned int reg)
+{
+ u16 *cache = codec->reg_cache;
+ BUG_ON(reg > WM9081_MAX_REGISTER);
+ return cache[reg];
+}
+
+static unsigned int wm9081_read_hw(struct snd_soc_codec *codec, u8 reg)
+{
+ struct i2c_msg xfer[2];
+ u16 data;
+ int ret;
+ struct i2c_client *client = codec->control_data;
+
+ BUG_ON(reg > WM9081_MAX_REGISTER);
+
+ /* Write register */
+ xfer[0].addr = client->addr;
+ xfer[0].flags = 0;
+ xfer[0].len = 1;
+ xfer[0].buf = ®
+
+ /* Read data */
+ xfer[1].addr = client->addr;
+ xfer[1].flags = I2C_M_RD;
+ xfer[1].len = 2;
+ xfer[1].buf = (u8 *)&data;
+
+ ret = i2c_transfer(client->adapter, xfer, 2);
+ if (ret != 2) {
+ dev_err(&client->dev, "i2c_transfer() returned %d\n", ret);
+ return 0;
+ }
+
+ return (data >> 8) | ((data & 0xff) << 8);
+}
+
+static unsigned int wm9081_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+ if (wm9081_reg_is_volatile(reg))
+ return wm9081_read_hw(codec, reg);
+ else
+ return wm9081_read_reg_cache(codec, reg);
+}
+
+static int wm9081_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int value)
+{
+ u16 *cache = codec->reg_cache;
+ u8 data[3];
+
+ BUG_ON(reg > WM9081_MAX_REGISTER);
+
+ if (!wm9081_reg_is_volatile(reg))
+ cache[reg] = value;
+
+ data[0] = reg;
+ data[1] = value >> 8;
+ data[2] = value & 0x00ff;
+
+ if (codec->hw_write(codec->control_data, data, 3) == 3)
+ return 0;
+ else
+ return -EIO;
+}
+
+static int wm9081_reset(struct snd_soc_codec *codec)
+{
+ return wm9081_write(codec, WM9081_SOFTWARE_RESET, 0);
+}
+
+static const DECLARE_TLV_DB_SCALE(drc_in_tlv, -4500, 75, 0);
+static const DECLARE_TLV_DB_SCALE(drc_out_tlv, -2250, 75, 0);
+static const DECLARE_TLV_DB_SCALE(drc_min_tlv, -1800, 600, 0);
+static unsigned int drc_max_tlv[] = {
+ TLV_DB_RANGE_HEAD(4),
+ 0, 0, TLV_DB_SCALE_ITEM(1200, 0, 0),
+ 1, 1, TLV_DB_SCALE_ITEM(1800, 0, 0),
+ 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0),
+ 3, 3, TLV_DB_SCALE_ITEM(3600, 0, 0),
+};
+static const DECLARE_TLV_DB_SCALE(drc_qr_tlv, 1200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(drc_startup_tlv, -300, 50, 0);
+
+static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+
+static const DECLARE_TLV_DB_SCALE(in_tlv, -600, 600, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -7200, 75, 1);
+static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
+
+static const char *drc_high_text[] = {
+ "1",
+ "1/2",
+ "1/4",
+ "1/8",
+ "1/16",
+ "0",
+};
+
+static const struct soc_enum drc_high =
+ SOC_ENUM_SINGLE(WM9081_DRC_3, 3, 6, drc_high_text);
+
+static const char *drc_low_text[] = {
+ "1",
+ "1/2",
+ "1/4",
+ "1/8",
+ "0",
+};
+
+static const struct soc_enum drc_low =
+ SOC_ENUM_SINGLE(WM9081_DRC_3, 0, 5, drc_low_text);
+
+static const char *drc_atk_text[] = {
+ "181us",
+ "181us",
+ "363us",
+ "726us",
+ "1.45ms",
+ "2.9ms",
+ "5.8ms",
+ "11.6ms",
+ "23.2ms",
+ "46.4ms",
+ "92.8ms",
+ "185.6ms",
+};
+
+static const struct soc_enum drc_atk =
+ SOC_ENUM_SINGLE(WM9081_DRC_2, 12, 12, drc_atk_text);
+
+static const char *drc_dcy_text[] = {
+ "186ms",
+ "372ms",
+ "743ms",
+ "1.49s",
+ "2.97s",
+ "5.94s",
+ "11.89s",
+ "23.78s",
+ "47.56s",
+};
+
+static const struct soc_enum drc_dcy =
+ SOC_ENUM_SINGLE(WM9081_DRC_2, 8, 9, drc_dcy_text);
+
+static const char *drc_qr_dcy_text[] = {
+ "0.725ms",
+ "1.45ms",
+ "5.8ms",
+};
+
+static const struct soc_enum drc_qr_dcy =
+ SOC_ENUM_SINGLE(WM9081_DRC_2, 4, 3, drc_qr_dcy_text);
+
+static const char *dac_deemph_text[] = {
+ "None",
+ "32kHz",
+ "44.1kHz",
+ "48kHz",
+};
+
+static const struct soc_enum dac_deemph =
+ SOC_ENUM_SINGLE(WM9081_DAC_DIGITAL_2, 1, 4, dac_deemph_text);
+
+static const char *speaker_mode_text[] = {
+ "Class D",
+ "Class AB",
+};
+
+static const struct soc_enum speaker_mode =
+ SOC_ENUM_SINGLE(WM9081_ANALOGUE_SPEAKER_2, 6, 2, speaker_mode_text);
+
+static int speaker_mode_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg;
+
+ reg = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2);
+ if (reg & WM9081_SPK_MODE)
+ ucontrol->value.integer.value[0] = 1;
+ else
+ ucontrol->value.integer.value[0] = 0;
+
+ return 0;
+}
+
+/*
+ * Stop any attempts to change speaker mode while the speaker is enabled.
+ *
+ * We also have some special anti-pop controls dependant on speaker
+ * mode which must be changed along with the mode.
+ */
+static int speaker_mode_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ unsigned int reg_pwr = wm9081_read(codec, WM9081_POWER_MANAGEMENT);
+ unsigned int reg2 = wm9081_read(codec, WM9081_ANALOGUE_SPEAKER_2);
+
+ /* Are we changing anything? */
+ if (ucontrol->value.integer.value[0] ==
+ ((reg2 & WM9081_SPK_MODE) != 0))
+ return 0;
+
+ /* Don't try to change modes while enabled */
+ if (reg_pwr & WM9081_SPK_ENA)
+ return -EINVAL;
+
+ if (ucontrol->value.integer.value[0]) {
+ /* Class AB */
+ reg2 &= ~(WM9081_SPK_INV_MUTE | WM9081_OUT_SPK_CTRL);
+ reg2 |= WM9081_SPK_MODE;
+ } else {
+ /* Class D */
+ reg2 |= WM9081_SPK_INV_MUTE | WM9081_OUT_SPK_CTRL;
+ reg2 &= ~WM9081_SPK_MODE;
+ }
+
+ wm9081_write(codec, WM9081_ANALOGUE_SPEAKER_2, reg2);
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new wm9081_snd_controls[] = {
+SOC_SINGLE_TLV("IN1 Volume", WM9081_ANALOGUE_MIXER, 1, 1, 1, in_tlv),
+SOC_SINGLE_TLV("IN2 Volume", WM9081_ANALOGUE_MIXER, 3, 1, 1, in_tlv),
+
+SOC_SINGLE_TLV("Playback Volume", WM9081_DAC_DIGITAL_1, 1, 96, 0, dac_tlv),
+
+SOC_SINGLE("LINEOUT Switch", WM9081_ANALOGUE_LINEOUT, 7, 1, 1),
+SOC_SINGLE("LINEOUT ZC Switch", WM9081_ANALOGUE_LINEOUT, 6, 1, 0),
+SOC_SINGLE_TLV("LINEOUT Volume", WM9081_ANALOGUE_LINEOUT, 0, 63, 0, out_tlv),
+
+SOC_SINGLE("DRC Switch", WM9081_DRC_1, 15, 1, 0),
+SOC_ENUM("DRC High Slope", drc_high),
+SOC_ENUM("DRC Low Slope", drc_low),
+SOC_SINGLE_TLV("DRC Input Volume", WM9081_DRC_4, 5, 60, 1, drc_in_tlv),
+SOC_SINGLE_TLV("DRC Output Volume", WM9081_DRC_4, 0, 30, 1, drc_out_tlv),
+SOC_SINGLE_TLV("DRC Minimum Volume", WM9081_DRC_2, 2, 3, 1, drc_min_tlv),
+SOC_SINGLE_TLV("DRC Maximum Volume", WM9081_DRC_2, 0, 3, 0, drc_max_tlv),
+SOC_ENUM("DRC Attack", drc_atk),
+SOC_ENUM("DRC Decay", drc_dcy),
+SOC_SINGLE("DRC Quick Release Switch", WM9081_DRC_1, 2, 1, 0),
+SOC_SINGLE_TLV("DRC Quick Release Volume", WM9081_DRC_2, 6, 3, 0, drc_qr_tlv),
+SOC_ENUM("DRC Quick Release Decay", drc_qr_dcy),
+SOC_SINGLE_TLV("DRC Startup Volume", WM9081_DRC_1, 6, 18, 0, drc_startup_tlv),
+
+SOC_SINGLE("EQ Switch", WM9081_EQ_1, 0, 1, 0),
+
+SOC_SINGLE("Speaker DC Volume", WM9081_ANALOGUE_SPEAKER_1, 3, 5, 0),
+SOC_SINGLE("Speaker AC Volume", WM9081_ANALOGUE_SPEAKER_1, 0, 5, 0),
+SOC_SINGLE("Speaker Switch", WM9081_ANALOGUE_SPEAKER_PGA, 7, 1, 1),
+SOC_SINGLE("Speaker ZC Switch", WM9081_ANALOGUE_SPEAKER_PGA, 6, 1, 0),
+SOC_SINGLE_TLV("Speaker Volume", WM9081_ANALOGUE_SPEAKER_PGA, 0, 63, 0,
+ out_tlv),
+SOC_ENUM("DAC Deemphasis", dac_deemph),
+SOC_ENUM_EXT("Speaker Mode", speaker_mode, speaker_mode_get, speaker_mode_put),
+};
+
+static const struct snd_kcontrol_new wm9081_eq_controls[] = {
+SOC_SINGLE_TLV("EQ1 Volume", WM9081_EQ_1, 11, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 Volume", WM9081_EQ_1, 6, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 Volume", WM9081_EQ_1, 1, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 Volume", WM9081_EQ_2, 11, 24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ5 Volume", WM9081_EQ_2, 6, 24, 0, eq_tlv),
+};
+
+static const struct snd_kcontrol_new mixer[] = {
+SOC_DAPM_SINGLE("IN1 Switch", WM9081_ANALOGUE_MIXER, 0, 1, 0),
+SOC_DAPM_SINGLE("IN2 Switch", WM9081_ANALOGUE_MIXER, 2, 1, 0),
+SOC_DAPM_SINGLE("Playback Switch", WM9081_ANALOGUE_MIXER, 4, 1, 0),
+};
+
+static int speaker_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = w->codec;
+ unsigned int reg = wm9081_read(codec, WM9081_POWER_MANAGEMENT);
+
+ switch (event) {
+ case SND_SOC_DAPM_POST_PMU:
+ reg |= WM9081_SPK_ENA;
+ break;
+
+ case SND_SOC_DAPM_PRE_PMD:
+ reg &= ~WM9081_SPK_ENA;
+ break;
+ }
+
+ wm9081_write(codec, WM9081_POWER_MANAGEMENT, reg);
+
+ return 0;
+}
+
+struct _fll_div {
+ u16 fll_fratio;
+ u16 fll_outdiv;
+ u16 fll_clk_ref_div;
+ u16 n;
+ u16 k;
+};
+
+/* The size in bits of the FLL divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_FLL_SIZE ((1 << 16) * 10)
+
+static struct {
+ unsigned int min;
+ unsigned int max;
+ u16 fll_fratio;
+ int ratio;
+} fll_fratios[] = {
+ { 0, 64000, 4, 16 },
+ { 64000, 128000, 3, 8 },
+ { 128000, 256000, 2, 4 },
+ { 256000, 1000000, 1, 2 },
+ { 1000000, 13500000, 0, 1 },
+};
+
+static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
+ unsigned int Fout)
+{
+ u64 Kpart;
+ unsigned int K, Ndiv, Nmod, target;
+ unsigned int div;
+ int i;
+
+ /* Fref must be <=13.5MHz */
+ div = 1;
+ while ((Fref / div) > 13500000) {
+ div *= 2;
+
+ if (div > 8) {
+ pr_err("Can't scale %dMHz input down to <=13.5MHz\n",
+ Fref);
+ return -EINVAL;
+ }
+ }
+ fll_div->fll_clk_ref_div = div / 2;
+
+ pr_debug("Fref=%u Fout=%u\n", Fref, Fout);
+
+ /* Apply the division for our remaining calculations */
+ Fref /= div;
+
+ /* Fvco should be 90-100MHz; don't check the upper bound */
+ div = 0;
+ target = Fout * 2;
+ while (target < 90000000) {
+ div++;
+ target *= 2;
+ if (div > 7) {
+ pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n",
+ Fout);
+ return -EINVAL;
+ }
+ }
+ fll_div->fll_outdiv = div;
+
+ pr_debug("Fvco=%dHz\n", target);
+
+ /* Find an appropraite FLL_FRATIO and factor it out of the target */
+ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
+ if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
+ fll_div->fll_fratio = fll_fratios[i].fll_fratio;
+ target /= fll_fratios[i].ratio;
+ break;
+ }
+ }
+ if (i == ARRAY_SIZE(fll_fratios)) {
+ pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref);
+ return -EINVAL;
+ }
+
+ /* Now, calculate N.K */
+ Ndiv = target / Fref;
+
+ fll_div->n = Ndiv;
+ Nmod = target % Fref;
+ pr_debug("Nmod=%d\n", Nmod);
+
+ /* Calculate fractional part - scale up so we can round. */
+ Kpart = FIXED_FLL_SIZE * (long long)Nmod;
+
+ do_div(Kpart, Fref);
+
+ K = Kpart & 0xFFFFFFFF;
+
+ if ((K % 10) >= 5)
+ K += 5;
+
+ /* Move down to proper range now rounding is done */
+ fll_div->k = K / 10;
+
+ pr_debug("N=%x K=%x FLL_FRATIO=%x FLL_OUTDIV=%x FLL_CLK_REF_DIV=%x\n",
+ fll_div->n, fll_div->k,
+ fll_div->fll_fratio, fll_div->fll_outdiv,
+ fll_div->fll_clk_ref_div);
+
+ return 0;
+}
+
+static int wm9081_set_fll(struct snd_soc_codec *codec, int fll_id,
+ unsigned int Fref, unsigned int Fout)
+{
+ struct wm9081_priv *wm9081 = codec->private_data;
+ u16 reg1, reg4, reg5;
+ struct _fll_div fll_div;
+ int ret;
+ int clk_sys_reg;
+
+ /* Any change? */
+ if (Fref == wm9081->fll_fref && Fout == wm9081->fll_fout)
+ return 0;
+
+ /* Disable the FLL */
+ if (Fout == 0) {
+ dev_dbg(codec->dev, "FLL disabled\n");
+ wm9081->fll_fref = 0;
+ wm9081->fll_fout = 0;
+
+ return 0;
+ }
+
+ ret = fll_factors(&fll_div, Fref, Fout);
+ if (ret != 0)
+ return ret;
+
+ reg5 = wm9081_read(codec, WM9081_FLL_CONTROL_5);
+ reg5 &= ~WM9081_FLL_CLK_SRC_MASK;
+
+ switch (fll_id) {
+ case WM9081_SYSCLK_FLL_MCLK:
+ reg5 |= 0x1;
+ break;
+
+ default:
+ dev_err(codec->dev, "Unknown FLL ID %d\n", fll_id);
+ return -EINVAL;
+ }
+
+ /* Disable CLK_SYS while we reconfigure */
+ clk_sys_reg = wm9081_read(codec, WM9081_CLOCK_CONTROL_3);
+ if (clk_sys_reg & WM9081_CLK_SYS_ENA)
+ wm9081_write(code
Code Review / linux-2.6.git / commitdiff