OpenCloudOS-Kernel/sound/soc/codecs/wm8974.c

740 lines
21 KiB
C
Raw Permalink Normal View History

// SPDX-License-Identifier: GPL-2.0-only
/*
* wm8974.c -- WM8974 ALSA Soc Audio driver
*
* Copyright 2006-2009 Wolfson Microelectronics PLC.
*
* Author: Liam Girdwood <Liam.Girdwood@wolfsonmicro.com>
*/
#include <linux/module.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/regmap.h>
include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo <tj@kernel.org> Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org> Cc: Ingo Molnar <mingo@redhat.com> Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
2010-03-24 16:04:11 +08:00
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include "wm8974.h"
struct wm8974_priv {
unsigned int mclk;
unsigned int fs;
};
static const struct reg_default wm8974_reg_defaults[] = {
{ 0, 0x0000 }, { 1, 0x0000 }, { 2, 0x0000 }, { 3, 0x0000 },
{ 4, 0x0050 }, { 5, 0x0000 }, { 6, 0x0140 }, { 7, 0x0000 },
{ 8, 0x0000 }, { 9, 0x0000 }, { 10, 0x0000 }, { 11, 0x00ff },
{ 12, 0x0000 }, { 13, 0x0000 }, { 14, 0x0100 }, { 15, 0x00ff },
{ 16, 0x0000 }, { 17, 0x0000 }, { 18, 0x012c }, { 19, 0x002c },
{ 20, 0x002c }, { 21, 0x002c }, { 22, 0x002c }, { 23, 0x0000 },
{ 24, 0x0032 }, { 25, 0x0000 }, { 26, 0x0000 }, { 27, 0x0000 },
{ 28, 0x0000 }, { 29, 0x0000 }, { 30, 0x0000 }, { 31, 0x0000 },
{ 32, 0x0038 }, { 33, 0x000b }, { 34, 0x0032 }, { 35, 0x0000 },
{ 36, 0x0008 }, { 37, 0x000c }, { 38, 0x0093 }, { 39, 0x00e9 },
{ 40, 0x0000 }, { 41, 0x0000 }, { 42, 0x0000 }, { 43, 0x0000 },
{ 44, 0x0003 }, { 45, 0x0010 }, { 46, 0x0000 }, { 47, 0x0000 },
{ 48, 0x0000 }, { 49, 0x0002 }, { 50, 0x0000 }, { 51, 0x0000 },
{ 52, 0x0000 }, { 53, 0x0000 }, { 54, 0x0039 }, { 55, 0x0000 },
{ 56, 0x0000 },
};
#define WM8974_POWER1_BIASEN 0x08
#define WM8974_POWER1_BUFIOEN 0x04
#define wm8974_reset(c) snd_soc_component_write(c, WM8974_RESET, 0)
static const char *wm8974_companding[] = {"Off", "NC", "u-law", "A-law" };
static const char *wm8974_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz" };
static const char *wm8974_eqmode[] = {"Capture", "Playback" };
static const char *wm8974_bw[] = {"Narrow", "Wide" };
static const char *wm8974_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz" };
static const char *wm8974_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz" };
static const char *wm8974_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz" };
static const char *wm8974_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" };
static const char *wm8974_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz" };
static const char *wm8974_alc[] = {"ALC", "Limiter" };
static const struct soc_enum wm8974_enum[] = {
SOC_ENUM_SINGLE(WM8974_COMP, 1, 4, wm8974_companding), /* adc */
SOC_ENUM_SINGLE(WM8974_COMP, 3, 4, wm8974_companding), /* dac */
SOC_ENUM_SINGLE(WM8974_DAC, 4, 4, wm8974_deemp),
SOC_ENUM_SINGLE(WM8974_EQ1, 8, 2, wm8974_eqmode),
SOC_ENUM_SINGLE(WM8974_EQ1, 5, 4, wm8974_eq1),
SOC_ENUM_SINGLE(WM8974_EQ2, 8, 2, wm8974_bw),
SOC_ENUM_SINGLE(WM8974_EQ2, 5, 4, wm8974_eq2),
SOC_ENUM_SINGLE(WM8974_EQ3, 8, 2, wm8974_bw),
SOC_ENUM_SINGLE(WM8974_EQ3, 5, 4, wm8974_eq3),
SOC_ENUM_SINGLE(WM8974_EQ4, 8, 2, wm8974_bw),
SOC_ENUM_SINGLE(WM8974_EQ4, 5, 4, wm8974_eq4),
SOC_ENUM_SINGLE(WM8974_EQ5, 8, 2, wm8974_bw),
SOC_ENUM_SINGLE(WM8974_EQ5, 5, 4, wm8974_eq5),
SOC_ENUM_SINGLE(WM8974_ALC3, 8, 2, wm8974_alc),
};
static const char *wm8974_auxmode_text[] = { "Buffer", "Mixer" };
static SOC_ENUM_SINGLE_DECL(wm8974_auxmode,
WM8974_INPUT, 3, wm8974_auxmode_text);
static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1);
static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0);
static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0);
static const struct snd_kcontrol_new wm8974_snd_controls[] = {
SOC_SINGLE("Digital Loopback Switch", WM8974_COMP, 0, 1, 0),
SOC_ENUM("DAC Companding", wm8974_enum[1]),
SOC_ENUM("ADC Companding", wm8974_enum[0]),
SOC_ENUM("Playback De-emphasis", wm8974_enum[2]),
SOC_SINGLE("DAC Inversion Switch", WM8974_DAC, 0, 1, 0),
SOC_SINGLE_TLV("PCM Volume", WM8974_DACVOL, 0, 255, 0, digital_tlv),
SOC_SINGLE("High Pass Filter Switch", WM8974_ADC, 8, 1, 0),
SOC_SINGLE("High Pass Cut Off", WM8974_ADC, 4, 7, 0),
SOC_SINGLE("ADC Inversion Switch", WM8974_ADC, 0, 1, 0),
SOC_SINGLE_TLV("Capture Volume", WM8974_ADCVOL, 0, 255, 0, digital_tlv),
SOC_ENUM("Equaliser Function", wm8974_enum[3]),
SOC_ENUM("EQ1 Cut Off", wm8974_enum[4]),
SOC_SINGLE_TLV("EQ1 Volume", WM8974_EQ1, 0, 24, 1, eq_tlv),
SOC_ENUM("Equaliser EQ2 Bandwidth", wm8974_enum[5]),
SOC_ENUM("EQ2 Cut Off", wm8974_enum[6]),
SOC_SINGLE_TLV("EQ2 Volume", WM8974_EQ2, 0, 24, 1, eq_tlv),
SOC_ENUM("Equaliser EQ3 Bandwidth", wm8974_enum[7]),
SOC_ENUM("EQ3 Cut Off", wm8974_enum[8]),
SOC_SINGLE_TLV("EQ3 Volume", WM8974_EQ3, 0, 24, 1, eq_tlv),
SOC_ENUM("Equaliser EQ4 Bandwidth", wm8974_enum[9]),
SOC_ENUM("EQ4 Cut Off", wm8974_enum[10]),
SOC_SINGLE_TLV("EQ4 Volume", WM8974_EQ4, 0, 24, 1, eq_tlv),
SOC_ENUM("Equaliser EQ5 Bandwidth", wm8974_enum[11]),
SOC_ENUM("EQ5 Cut Off", wm8974_enum[12]),
SOC_SINGLE_TLV("EQ5 Volume", WM8974_EQ5, 0, 24, 1, eq_tlv),
SOC_SINGLE("DAC Playback Limiter Switch", WM8974_DACLIM1, 8, 1, 0),
SOC_SINGLE("DAC Playback Limiter Decay", WM8974_DACLIM1, 4, 15, 0),
SOC_SINGLE("DAC Playback Limiter Attack", WM8974_DACLIM1, 0, 15, 0),
SOC_SINGLE("DAC Playback Limiter Threshold", WM8974_DACLIM2, 4, 7, 0),
SOC_SINGLE("DAC Playback Limiter Boost", WM8974_DACLIM2, 0, 15, 0),
SOC_SINGLE("ALC Enable Switch", WM8974_ALC1, 8, 1, 0),
SOC_SINGLE("ALC Capture Max Gain", WM8974_ALC1, 3, 7, 0),
SOC_SINGLE("ALC Capture Min Gain", WM8974_ALC1, 0, 7, 0),
SOC_SINGLE("ALC Capture ZC Switch", WM8974_ALC2, 8, 1, 0),
SOC_SINGLE("ALC Capture Hold", WM8974_ALC2, 4, 7, 0),
SOC_SINGLE("ALC Capture Target", WM8974_ALC2, 0, 15, 0),
SOC_ENUM("ALC Capture Mode", wm8974_enum[13]),
SOC_SINGLE("ALC Capture Decay", WM8974_ALC3, 4, 15, 0),
SOC_SINGLE("ALC Capture Attack", WM8974_ALC3, 0, 15, 0),
SOC_SINGLE("ALC Capture Noise Gate Switch", WM8974_NGATE, 3, 1, 0),
SOC_SINGLE("ALC Capture Noise Gate Threshold", WM8974_NGATE, 0, 7, 0),
SOC_SINGLE("Capture PGA ZC Switch", WM8974_INPPGA, 7, 1, 0),
SOC_SINGLE_TLV("Capture PGA Volume", WM8974_INPPGA, 0, 63, 0, inpga_tlv),
SOC_SINGLE("Speaker Playback ZC Switch", WM8974_SPKVOL, 7, 1, 0),
SOC_SINGLE("Speaker Playback Switch", WM8974_SPKVOL, 6, 1, 1),
SOC_SINGLE_TLV("Speaker Playback Volume", WM8974_SPKVOL, 0, 63, 0, spk_tlv),
SOC_ENUM("Aux Mode", wm8974_auxmode),
SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST, 8, 1, 0),
SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 1),
/* DAC / ADC oversampling */
SOC_SINGLE("DAC 128x Oversampling Switch", WM8974_DAC, 8, 1, 0),
SOC_SINGLE("ADC 128x Oversampling Switch", WM8974_ADC, 8, 1, 0),
};
/* Speaker Output Mixer */
static const struct snd_kcontrol_new wm8974_speaker_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_SPKMIX, 1, 1, 0),
SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_SPKMIX, 5, 1, 0),
SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_SPKMIX, 0, 1, 0),
};
/* Mono Output Mixer */
static const struct snd_kcontrol_new wm8974_mono_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", WM8974_MONOMIX, 1, 1, 0),
SOC_DAPM_SINGLE("Aux Playback Switch", WM8974_MONOMIX, 2, 1, 0),
SOC_DAPM_SINGLE("PCM Playback Switch", WM8974_MONOMIX, 0, 1, 0),
};
/* Boost mixer */
static const struct snd_kcontrol_new wm8974_boost_mixer[] = {
SOC_DAPM_SINGLE("Aux Switch", WM8974_INPPGA, 6, 1, 0),
};
/* Input PGA */
static const struct snd_kcontrol_new wm8974_inpga[] = {
SOC_DAPM_SINGLE("Aux Switch", WM8974_INPUT, 2, 1, 0),
SOC_DAPM_SINGLE("MicN Switch", WM8974_INPUT, 1, 1, 0),
SOC_DAPM_SINGLE("MicP Switch", WM8974_INPUT, 0, 1, 0),
};
/* AUX Input boost vol */
static const struct snd_kcontrol_new wm8974_aux_boost_controls =
SOC_DAPM_SINGLE("Aux Volume", WM8974_ADCBOOST, 0, 7, 0);
/* Mic Input boost vol */
static const struct snd_kcontrol_new wm8974_mic_boost_controls =
SOC_DAPM_SINGLE("Mic Volume", WM8974_ADCBOOST, 4, 7, 0);
static const struct snd_soc_dapm_widget wm8974_dapm_widgets[] = {
SND_SOC_DAPM_MIXER("Speaker Mixer", WM8974_POWER3, 2, 0,
&wm8974_speaker_mixer_controls[0],
ARRAY_SIZE(wm8974_speaker_mixer_controls)),
SND_SOC_DAPM_MIXER("Mono Mixer", WM8974_POWER3, 3, 0,
&wm8974_mono_mixer_controls[0],
ARRAY_SIZE(wm8974_mono_mixer_controls)),
SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8974_POWER3, 0, 0),
SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8974_POWER2, 0, 0),
SND_SOC_DAPM_PGA("Aux Input", WM8974_POWER1, 6, 0, NULL, 0),
SND_SOC_DAPM_PGA("SpkN Out", WM8974_POWER3, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("SpkP Out", WM8974_POWER3, 6, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mono Out", WM8974_POWER3, 7, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Input PGA", WM8974_POWER2, 2, 0, wm8974_inpga,
ARRAY_SIZE(wm8974_inpga)),
SND_SOC_DAPM_MIXER("Boost Mixer", WM8974_POWER2, 4, 0,
wm8974_boost_mixer, ARRAY_SIZE(wm8974_boost_mixer)),
SND_SOC_DAPM_SUPPLY("Mic Bias", WM8974_POWER1, 4, 0, NULL, 0),
SND_SOC_DAPM_INPUT("MICN"),
SND_SOC_DAPM_INPUT("MICP"),
SND_SOC_DAPM_INPUT("AUX"),
SND_SOC_DAPM_OUTPUT("MONOOUT"),
SND_SOC_DAPM_OUTPUT("SPKOUTP"),
SND_SOC_DAPM_OUTPUT("SPKOUTN"),
};
static const struct snd_soc_dapm_route wm8974_dapm_routes[] = {
/* 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"},
/* Boost Mixer */
{"ADC", NULL, "Boost Mixer"},
{"Boost Mixer", "Aux Switch", "Aux Input"},
{"Boost Mixer", NULL, "Input PGA"},
{"Boost Mixer", NULL, "MICP"},
/* Input PGA */
{"Input PGA", "Aux Switch", "Aux Input"},
{"Input PGA", "MicN Switch", "MICN"},
{"Input PGA", "MicP Switch", "MICP"},
/* Inputs */
{"Aux Input", NULL, "AUX"},
};
struct pll_ {
unsigned int pre_div:1;
unsigned int n:4;
unsigned int k;
};
/* 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(struct pll_ *pll_div,
unsigned int target, unsigned int source)
{
unsigned long long Kpart;
unsigned int K, Ndiv, Nmod;
/* There is a fixed divide by 4 in the output path */
target *= 4;
Ndiv = target / source;
if (Ndiv < 6) {
source /= 2;
pll_div->pre_div = 1;
Ndiv = target / source;
} else
pll_div->pre_div = 0;
if ((Ndiv < 6) || (Ndiv > 12))
printk(KERN_WARNING
"WM8974 N value %u outwith recommended range!\n",
Ndiv);
pll_div->n = Ndiv;
Nmod = target % source;
Kpart = FIXED_PLL_SIZE * (long long)Nmod;
do_div(Kpart, source);
K = Kpart & 0xFFFFFFFF;
/* Check if we need to round */
if ((K % 10) >= 5)
K += 5;
/* Move down to proper range now rounding is done */
K /= 10;
pll_div->k = K;
}
static int wm8974_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
int source, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_component *component = codec_dai->component;
struct pll_ pll_div;
u16 reg;
if (freq_in == 0 || freq_out == 0) {
/* Clock CODEC directly from MCLK */
reg = snd_soc_component_read32(component, WM8974_CLOCK);
snd_soc_component_write(component, WM8974_CLOCK, reg & 0x0ff);
/* Turn off PLL */
reg = snd_soc_component_read32(component, WM8974_POWER1);
snd_soc_component_write(component, WM8974_POWER1, reg & 0x1df);
return 0;
}
pll_factors(&pll_div, freq_out, freq_in);
snd_soc_component_write(component, WM8974_PLLN, (pll_div.pre_div << 4) | pll_div.n);
snd_soc_component_write(component, WM8974_PLLK1, pll_div.k >> 18);
snd_soc_component_write(component, WM8974_PLLK2, (pll_div.k >> 9) & 0x1ff);
snd_soc_component_write(component, WM8974_PLLK3, pll_div.k & 0x1ff);
reg = snd_soc_component_read32(component, WM8974_POWER1);
snd_soc_component_write(component, WM8974_POWER1, reg | 0x020);
/* Run CODEC from PLL instead of MCLK */
reg = snd_soc_component_read32(component, WM8974_CLOCK);
snd_soc_component_write(component, WM8974_CLOCK, reg | 0x100);
return 0;
}
/*
* Configure WM8974 clock dividers.
*/
static int wm8974_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
int div_id, int div)
{
struct snd_soc_component *component = codec_dai->component;
u16 reg;
switch (div_id) {
case WM8974_OPCLKDIV:
reg = snd_soc_component_read32(component, WM8974_GPIO) & 0x1cf;
snd_soc_component_write(component, WM8974_GPIO, reg | div);
break;
case WM8974_MCLKDIV:
reg = snd_soc_component_read32(component, WM8974_CLOCK) & 0x11f;
snd_soc_component_write(component, WM8974_CLOCK, reg | div);
break;
case WM8974_BCLKDIV:
reg = snd_soc_component_read32(component, WM8974_CLOCK) & 0x1e3;
snd_soc_component_write(component, WM8974_CLOCK, reg | div);
break;
default:
return -EINVAL;
}
return 0;
}
static unsigned int wm8974_get_mclkdiv(unsigned int f_in, unsigned int f_out,
int *mclkdiv)
{
unsigned int ratio = 2 * f_in / f_out;
if (ratio <= 2) {
*mclkdiv = WM8974_MCLKDIV_1;
ratio = 2;
} else if (ratio == 3) {
*mclkdiv = WM8974_MCLKDIV_1_5;
} else if (ratio == 4) {
*mclkdiv = WM8974_MCLKDIV_2;
} else if (ratio <= 6) {
*mclkdiv = WM8974_MCLKDIV_3;
ratio = 6;
} else if (ratio <= 8) {
*mclkdiv = WM8974_MCLKDIV_4;
ratio = 8;
} else if (ratio <= 12) {
*mclkdiv = WM8974_MCLKDIV_6;
ratio = 12;
} else if (ratio <= 16) {
*mclkdiv = WM8974_MCLKDIV_8;
ratio = 16;
} else {
*mclkdiv = WM8974_MCLKDIV_12;
ratio = 24;
}
return f_out * ratio / 2;
}
static int wm8974_update_clocks(struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct wm8974_priv *priv = snd_soc_component_get_drvdata(component);
unsigned int fs256;
unsigned int fpll = 0;
unsigned int f;
int mclkdiv;
if (!priv->mclk || !priv->fs)
return 0;
fs256 = 256 * priv->fs;
f = wm8974_get_mclkdiv(priv->mclk, fs256, &mclkdiv);
if (f != priv->mclk) {
/* The PLL performs best around 90MHz */
fpll = wm8974_get_mclkdiv(22500000, fs256, &mclkdiv);
}
wm8974_set_dai_pll(dai, 0, 0, priv->mclk, fpll);
wm8974_set_dai_clkdiv(dai, WM8974_MCLKDIV, mclkdiv);
return 0;
}
static int wm8974_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
struct snd_soc_component *component = dai->component;
struct wm8974_priv *priv = snd_soc_component_get_drvdata(component);
if (dir != SND_SOC_CLOCK_IN)
return -EINVAL;
priv->mclk = freq;
return wm8974_update_clocks(dai);
}
static int wm8974_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_component *component = codec_dai->component;
u16 iface = 0;
u16 clk = snd_soc_component_read32(component, WM8974_CLOCK) & 0x1fe;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
clk |= 0x0001;
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 |= 0x0010;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
iface |= 0x0008;
break;
case SND_SOC_DAIFMT_DSP_A:
iface |= 0x00018;
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 |= 0x0180;
break;
case SND_SOC_DAIFMT_IB_NF:
iface |= 0x0100;
break;
case SND_SOC_DAIFMT_NB_IF:
iface |= 0x0080;
break;
default:
return -EINVAL;
}
snd_soc_component_write(component, WM8974_IFACE, iface);
snd_soc_component_write(component, WM8974_CLOCK, clk);
return 0;
}
static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct wm8974_priv *priv = snd_soc_component_get_drvdata(component);
u16 iface = snd_soc_component_read32(component, WM8974_IFACE) & 0x19f;
u16 adn = snd_soc_component_read32(component, WM8974_ADD) & 0x1f1;
int err;
priv->fs = params_rate(params);
err = wm8974_update_clocks(dai);
if (err)
return err;
/* bit size */
switch (params_width(params)) {
case 16:
break;
case 20:
iface |= 0x0020;
break;
case 24:
iface |= 0x0040;
break;
case 32:
iface |= 0x0060;
break;
}
/* filter coefficient */
switch (params_rate(params)) {
case 8000:
adn |= 0x5 << 1;
break;
case 11025:
adn |= 0x4 << 1;
break;
case 16000:
adn |= 0x3 << 1;
break;
case 22050:
adn |= 0x2 << 1;
break;
case 32000:
adn |= 0x1 << 1;
break;
case 44100:
case 48000:
break;
}
snd_soc_component_write(component, WM8974_IFACE, iface);
snd_soc_component_write(component, WM8974_ADD, adn);
return 0;
}
static int wm8974_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_component *component = dai->component;
u16 mute_reg = snd_soc_component_read32(component, WM8974_DAC) & 0xffbf;
if (mute)
snd_soc_component_write(component, WM8974_DAC, mute_reg | 0x40);
else
snd_soc_component_write(component, WM8974_DAC, mute_reg);
return 0;
}
/* liam need to make this lower power with dapm */
static int wm8974_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
u16 power1 = snd_soc_component_read32(component, WM8974_POWER1) & ~0x3;
switch (level) {
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
power1 |= 0x1; /* VMID 50k */
snd_soc_component_write(component, WM8974_POWER1, power1);
break;
case SND_SOC_BIAS_STANDBY:
power1 |= WM8974_POWER1_BIASEN | WM8974_POWER1_BUFIOEN;
if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) {
regcache_sync(dev_get_regmap(component->dev, NULL));
/* Initial cap charge at VMID 5k */
snd_soc_component_write(component, WM8974_POWER1, power1 | 0x3);
mdelay(100);
}
power1 |= 0x2; /* VMID 500k */
snd_soc_component_write(component, WM8974_POWER1, power1);
break;
case SND_SOC_BIAS_OFF:
snd_soc_component_write(component, WM8974_POWER1, 0);
snd_soc_component_write(component, WM8974_POWER2, 0);
snd_soc_component_write(component, WM8974_POWER3, 0);
break;
}
return 0;
}
#define WM8974_RATES (SNDRV_PCM_RATE_8000_48000)
#define WM8974_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
static const struct snd_soc_dai_ops wm8974_ops = {
.hw_params = wm8974_pcm_hw_params,
.digital_mute = wm8974_mute,
.set_fmt = wm8974_set_dai_fmt,
.set_clkdiv = wm8974_set_dai_clkdiv,
.set_pll = wm8974_set_dai_pll,
.set_sysclk = wm8974_set_dai_sysclk,
};
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static struct snd_soc_dai_driver wm8974_dai = {
.name = "wm8974-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2, /* Only 1 channel of data */
.rates = WM8974_RATES,
.formats = WM8974_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2, /* Only 1 channel of data */
.rates = WM8974_RATES,
.formats = WM8974_FORMATS,},
.ops = &wm8974_ops,
.symmetric_rates = 1,
};
static const struct regmap_config wm8974_regmap = {
.reg_bits = 7,
.val_bits = 9,
.max_register = WM8974_MONOMIX,
.reg_defaults = wm8974_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(wm8974_reg_defaults),
.cache_type = REGCACHE_FLAT,
};
static int wm8974_probe(struct snd_soc_component *component)
{
int ret = 0;
ret = wm8974_reset(component);
if (ret < 0) {
dev_err(component->dev, "Failed to issue reset\n");
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
return ret;
}
return 0;
}
static const struct snd_soc_component_driver soc_component_dev_wm8974 = {
.probe = wm8974_probe,
.set_bias_level = wm8974_set_bias_level,
.controls = wm8974_snd_controls,
.num_controls = ARRAY_SIZE(wm8974_snd_controls),
.dapm_widgets = wm8974_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm8974_dapm_widgets),
.dapm_routes = wm8974_dapm_routes,
.num_dapm_routes = ARRAY_SIZE(wm8974_dapm_routes),
.suspend_bias_off = 1,
.idle_bias_on = 1,
.use_pmdown_time = 1,
.endianness = 1,
.non_legacy_dai_naming = 1,
};
static int wm8974_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct wm8974_priv *priv;
struct regmap *regmap;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
int ret;
priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
return -ENOMEM;
i2c_set_clientdata(i2c, priv);
regmap = devm_regmap_init_i2c(i2c, &wm8974_regmap);
if (IS_ERR(regmap))
return PTR_ERR(regmap);
ret = devm_snd_soc_register_component(&i2c->dev,
&soc_component_dev_wm8974, &wm8974_dai, 1);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
return ret;
}
static const struct i2c_device_id wm8974_i2c_id[] = {
{ "wm8974", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8974_i2c_id);
static const struct of_device_id wm8974_of_match[] = {
{ .compatible = "wlf,wm8974", },
{ }
};
MODULE_DEVICE_TABLE(of, wm8974_of_match);
static struct i2c_driver wm8974_i2c_driver = {
.driver = {
.name = "wm8974",
.of_match_table = wm8974_of_match,
},
.probe = wm8974_i2c_probe,
.id_table = wm8974_i2c_id,
};
module_i2c_driver(wm8974_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8974 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");