linux-stable/sound/soc/codecs/wm8350.c

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/*
* wm8350.c -- WM8350 ALSA SoC audio driver
*
* Copyright (C) 2007-12 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.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.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 08:04:11 +00:00
#include <linux/slab.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/platform_device.h>
#include <linux/mfd/wm8350/audio.h>
#include <linux/mfd/wm8350/core.h>
#include <linux/regulator/consumer.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 <trace/events/asoc.h>
#include "wm8350.h"
#define WM8350_OUTn_0dB 0x39
#define WM8350_RAMP_NONE 0
#define WM8350_RAMP_UP 1
#define WM8350_RAMP_DOWN 2
/* We only include the analogue supplies here; the digital supplies
* need to be available well before this driver can be probed.
*/
static const char *supply_names[] = {
"AVDD",
"HPVDD",
};
struct wm8350_output {
u16 active;
u16 left_vol;
u16 right_vol;
u16 ramp;
u16 mute;
};
struct wm8350_jack_data {
struct snd_soc_jack *jack;
struct delayed_work work;
int report;
int short_report;
};
struct wm8350_data {
struct wm8350 *wm8350;
struct wm8350_output out1;
struct wm8350_output out2;
struct wm8350_jack_data hpl;
struct wm8350_jack_data hpr;
struct wm8350_jack_data mic;
struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
int fll_freq_out;
int fll_freq_in;
struct delayed_work pga_work;
};
/*
* Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown.
*/
static inline int wm8350_out1_ramp_step(struct wm8350_data *wm8350_data)
{
struct wm8350_output *out1 = &wm8350_data->out1;
struct wm8350 *wm8350 = wm8350_data->wm8350;
int left_complete = 0, right_complete = 0;
u16 reg, val;
/* left channel */
reg = wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME);
val = (reg & WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
if (out1->ramp == WM8350_RAMP_UP) {
/* ramp step up */
if (val < out1->left_vol) {
val++;
reg &= ~WM8350_OUT1L_VOL_MASK;
wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME,
reg | (val << WM8350_OUT1L_VOL_SHIFT));
} else
left_complete = 1;
} else if (out1->ramp == WM8350_RAMP_DOWN) {
/* ramp step down */
if (val > 0) {
val--;
reg &= ~WM8350_OUT1L_VOL_MASK;
wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME,
reg | (val << WM8350_OUT1L_VOL_SHIFT));
} else
left_complete = 1;
} else
return 1;
/* right channel */
reg = wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME);
val = (reg & WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
if (out1->ramp == WM8350_RAMP_UP) {
/* ramp step up */
if (val < out1->right_vol) {
val++;
reg &= ~WM8350_OUT1R_VOL_MASK;
wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME,
reg | (val << WM8350_OUT1R_VOL_SHIFT));
} else
right_complete = 1;
} else if (out1->ramp == WM8350_RAMP_DOWN) {
/* ramp step down */
if (val > 0) {
val--;
reg &= ~WM8350_OUT1R_VOL_MASK;
wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME,
reg | (val << WM8350_OUT1R_VOL_SHIFT));
} else
right_complete = 1;
}
/* only hit the update bit if either volume has changed this step */
if (!left_complete || !right_complete)
wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME, WM8350_OUT1_VU);
return left_complete & right_complete;
}
/*
* Ramp OUT2 PGA volume to minimise pops at stream startup and shutdown.
*/
static inline int wm8350_out2_ramp_step(struct wm8350_data *wm8350_data)
{
struct wm8350_output *out2 = &wm8350_data->out2;
struct wm8350 *wm8350 = wm8350_data->wm8350;
int left_complete = 0, right_complete = 0;
u16 reg, val;
/* left channel */
reg = wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME);
val = (reg & WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
if (out2->ramp == WM8350_RAMP_UP) {
/* ramp step up */
if (val < out2->left_vol) {
val++;
reg &= ~WM8350_OUT2L_VOL_MASK;
wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME,
reg | (val << WM8350_OUT1L_VOL_SHIFT));
} else
left_complete = 1;
} else if (out2->ramp == WM8350_RAMP_DOWN) {
/* ramp step down */
if (val > 0) {
val--;
reg &= ~WM8350_OUT2L_VOL_MASK;
wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME,
reg | (val << WM8350_OUT1L_VOL_SHIFT));
} else
left_complete = 1;
} else
return 1;
/* right channel */
reg = wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME);
val = (reg & WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
if (out2->ramp == WM8350_RAMP_UP) {
/* ramp step up */
if (val < out2->right_vol) {
val++;
reg &= ~WM8350_OUT2R_VOL_MASK;
wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME,
reg | (val << WM8350_OUT1R_VOL_SHIFT));
} else
right_complete = 1;
} else if (out2->ramp == WM8350_RAMP_DOWN) {
/* ramp step down */
if (val > 0) {
val--;
reg &= ~WM8350_OUT2R_VOL_MASK;
wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME,
reg | (val << WM8350_OUT1R_VOL_SHIFT));
} else
right_complete = 1;
}
/* only hit the update bit if either volume has changed this step */
if (!left_complete || !right_complete)
wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME, WM8350_OUT2_VU);
return left_complete & right_complete;
}
/*
* This work ramps both output PGAs at stream start/stop time to
* minimise pop associated with DAPM power switching.
* It's best to enable Zero Cross when ramping occurs to minimise any
* zipper noises.
*/
static void wm8350_pga_work(struct work_struct *work)
{
struct wm8350_data *wm8350_data =
container_of(work, struct wm8350_data, pga_work.work);
struct wm8350_output *out1 = &wm8350_data->out1,
*out2 = &wm8350_data->out2;
int i, out1_complete, out2_complete;
/* do we need to ramp at all ? */
if (out1->ramp == WM8350_RAMP_NONE && out2->ramp == WM8350_RAMP_NONE)
return;
/* PGA volumes have 6 bits of resolution to ramp */
for (i = 0; i <= 63; i++) {
out1_complete = 1, out2_complete = 1;
if (out1->ramp != WM8350_RAMP_NONE)
out1_complete = wm8350_out1_ramp_step(wm8350_data);
if (out2->ramp != WM8350_RAMP_NONE)
out2_complete = wm8350_out2_ramp_step(wm8350_data);
/* ramp finished ? */
if (out1_complete && out2_complete)
break;
/* we need to delay longer on the up ramp */
if (out1->ramp == WM8350_RAMP_UP ||
out2->ramp == WM8350_RAMP_UP) {
/* delay is longer over 0dB as increases are larger */
if (i >= WM8350_OUTn_0dB)
schedule_timeout_interruptible(msecs_to_jiffies
(2));
else
schedule_timeout_interruptible(msecs_to_jiffies
(1));
} else
udelay(50); /* doesn't matter if we delay longer */
}
out1->ramp = WM8350_RAMP_NONE;
out2->ramp = WM8350_RAMP_NONE;
}
/*
* WM8350 Controls
*/
static int pga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
struct wm8350_data *wm8350_data = snd_soc_component_get_drvdata(component);
struct wm8350_output *out;
switch (w->shift) {
case 0:
case 1:
out = &wm8350_data->out1;
break;
case 2:
case 3:
out = &wm8350_data->out2;
break;
default:
WARN(1, "Invalid shift %d\n", w->shift);
return -1;
}
switch (event) {
case SND_SOC_DAPM_POST_PMU:
out->ramp = WM8350_RAMP_UP;
out->active = 1;
schedule_delayed_work(&wm8350_data->pga_work,
msecs_to_jiffies(1));
break;
case SND_SOC_DAPM_PRE_PMD:
out->ramp = WM8350_RAMP_DOWN;
out->active = 0;
schedule_delayed_work(&wm8350_data->pga_work,
msecs_to_jiffies(1));
break;
}
return 0;
}
static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
struct wm8350_data *wm8350_priv = snd_soc_component_get_drvdata(component);
struct wm8350_output *out = NULL;
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
int ret;
unsigned int reg = mc->reg;
u16 val;
/* For OUT1 and OUT2 we shadow the values and only actually write
* them out when active in order to ensure the amplifier comes on
* as quietly as possible. */
switch (reg) {
case WM8350_LOUT1_VOLUME:
out = &wm8350_priv->out1;
break;
case WM8350_LOUT2_VOLUME:
out = &wm8350_priv->out2;
break;
default:
break;
}
if (out) {
out->left_vol = ucontrol->value.integer.value[0];
out->right_vol = ucontrol->value.integer.value[1];
if (!out->active)
return 1;
}
ret = snd_soc_put_volsw(kcontrol, ucontrol);
if (ret < 0)
return ret;
/* now hit the volume update bits (always bit 8) */
val = snd_soc_component_read32(component, reg);
snd_soc_component_write(component, reg, val | WM8350_OUT1_VU);
return 1;
}
static int wm8350_get_volsw_2r(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
struct wm8350_data *wm8350_priv = snd_soc_component_get_drvdata(component);
struct wm8350_output *out1 = &wm8350_priv->out1;
struct wm8350_output *out2 = &wm8350_priv->out2;
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
unsigned int reg = mc->reg;
/* If these are cached registers use the cache */
switch (reg) {
case WM8350_LOUT1_VOLUME:
ucontrol->value.integer.value[0] = out1->left_vol;
ucontrol->value.integer.value[1] = out1->right_vol;
return 0;
case WM8350_LOUT2_VOLUME:
ucontrol->value.integer.value[0] = out2->left_vol;
ucontrol->value.integer.value[1] = out2->right_vol;
return 0;
default:
break;
}
return snd_soc_get_volsw(kcontrol, ucontrol);
}
static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" };
static const char *wm8350_dacmutem[] = { "Normal", "Soft" };
static const char *wm8350_dacmutes[] = { "Fast", "Slow" };
static const char *wm8350_adcfilter[] = { "None", "High Pass" };
static const char *wm8350_adchp[] = { "44.1kHz", "8kHz", "16kHz", "32kHz" };
static const char *wm8350_lr[] = { "Left", "Right" };
static const struct soc_enum wm8350_enum[] = {
SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 4, 4, wm8350_deemp),
SOC_ENUM_SINGLE(WM8350_DAC_CONTROL, 0, 4, wm8350_pol),
SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 14, 2, wm8350_dacmutem),
SOC_ENUM_SINGLE(WM8350_DAC_MUTE_VOLUME, 13, 2, wm8350_dacmutes),
SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 15, 2, wm8350_adcfilter),
SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 8, 4, wm8350_adchp),
SOC_ENUM_SINGLE(WM8350_ADC_CONTROL, 0, 4, wm8350_pol),
SOC_ENUM_SINGLE(WM8350_INPUT_MIXER_VOLUME, 15, 2, wm8350_lr),
};
static DECLARE_TLV_DB_SCALE(pre_amp_tlv, -1200, 3525, 0);
static DECLARE_TLV_DB_SCALE(out_pga_tlv, -5700, 600, 0);
static DECLARE_TLV_DB_SCALE(dac_pcm_tlv, -7163, 36, 1);
static DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -12700, 50, 1);
static DECLARE_TLV_DB_SCALE(out_mix_tlv, -1500, 300, 1);
static const DECLARE_TLV_DB_RANGE(capture_sd_tlv,
0, 12, TLV_DB_SCALE_ITEM(-3600, 300, 1),
13, 15, TLV_DB_SCALE_ITEM(0, 0, 0)
);
static const struct snd_kcontrol_new wm8350_snd_controls[] = {
SOC_ENUM("Playback Deemphasis", wm8350_enum[0]),
SOC_ENUM("Playback DAC Inversion", wm8350_enum[1]),
SOC_DOUBLE_R_EXT_TLV("Playback PCM Volume",
WM8350_DAC_DIGITAL_VOLUME_L,
WM8350_DAC_DIGITAL_VOLUME_R,
0, 255, 0, wm8350_get_volsw_2r,
wm8350_put_volsw_2r_vu, dac_pcm_tlv),
SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]),
SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]),
SOC_ENUM("Capture PCM Filter", wm8350_enum[4]),
SOC_ENUM("Capture PCM HP Filter", wm8350_enum[5]),
SOC_ENUM("Capture ADC Inversion", wm8350_enum[6]),
SOC_DOUBLE_R_EXT_TLV("Capture PCM Volume",
WM8350_ADC_DIGITAL_VOLUME_L,
WM8350_ADC_DIGITAL_VOLUME_R,
0, 255, 0, wm8350_get_volsw_2r,
wm8350_put_volsw_2r_vu, adc_pcm_tlv),
SOC_DOUBLE_TLV("Capture Sidetone Volume",
WM8350_ADC_DIVIDER,
8, 4, 15, 1, capture_sd_tlv),
SOC_DOUBLE_R_EXT_TLV("Capture Volume",
WM8350_LEFT_INPUT_VOLUME,
WM8350_RIGHT_INPUT_VOLUME,
2, 63, 0, wm8350_get_volsw_2r,
wm8350_put_volsw_2r_vu, pre_amp_tlv),
SOC_DOUBLE_R("Capture ZC Switch",
WM8350_LEFT_INPUT_VOLUME,
WM8350_RIGHT_INPUT_VOLUME, 13, 1, 0),
SOC_SINGLE_TLV("Left Input Left Sidetone Volume",
WM8350_OUTPUT_LEFT_MIXER_VOLUME, 1, 7, 0, out_mix_tlv),
SOC_SINGLE_TLV("Left Input Right Sidetone Volume",
WM8350_OUTPUT_LEFT_MIXER_VOLUME,
5, 7, 0, out_mix_tlv),
SOC_SINGLE_TLV("Left Input Bypass Volume",
WM8350_OUTPUT_LEFT_MIXER_VOLUME,
9, 7, 0, out_mix_tlv),
SOC_SINGLE_TLV("Right Input Left Sidetone Volume",
WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
1, 7, 0, out_mix_tlv),
SOC_SINGLE_TLV("Right Input Right Sidetone Volume",
WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
5, 7, 0, out_mix_tlv),
SOC_SINGLE_TLV("Right Input Bypass Volume",
WM8350_OUTPUT_RIGHT_MIXER_VOLUME,
13, 7, 0, out_mix_tlv),
SOC_SINGLE("Left Input Mixer +20dB Switch",
WM8350_INPUT_MIXER_VOLUME_L, 0, 1, 0),
SOC_SINGLE("Right Input Mixer +20dB Switch",
WM8350_INPUT_MIXER_VOLUME_R, 0, 1, 0),
SOC_SINGLE_TLV("Out4 Capture Volume",
WM8350_INPUT_MIXER_VOLUME,
1, 7, 0, out_mix_tlv),
SOC_DOUBLE_R_EXT_TLV("Out1 Playback Volume",
WM8350_LOUT1_VOLUME,
WM8350_ROUT1_VOLUME,
2, 63, 0, wm8350_get_volsw_2r,
wm8350_put_volsw_2r_vu, out_pga_tlv),
SOC_DOUBLE_R("Out1 Playback ZC Switch",
WM8350_LOUT1_VOLUME,
WM8350_ROUT1_VOLUME, 13, 1, 0),
SOC_DOUBLE_R_EXT_TLV("Out2 Playback Volume",
WM8350_LOUT2_VOLUME,
WM8350_ROUT2_VOLUME,
2, 63, 0, wm8350_get_volsw_2r,
wm8350_put_volsw_2r_vu, out_pga_tlv),
SOC_DOUBLE_R("Out2 Playback ZC Switch", WM8350_LOUT2_VOLUME,
WM8350_ROUT2_VOLUME, 13, 1, 0),
SOC_SINGLE("Out2 Right Invert Switch", WM8350_ROUT2_VOLUME, 10, 1, 0),
SOC_SINGLE_TLV("Out2 Beep Volume", WM8350_BEEP_VOLUME,
5, 7, 0, out_mix_tlv),
SOC_DOUBLE_R("Out1 Playback Switch",
WM8350_LOUT1_VOLUME,
WM8350_ROUT1_VOLUME,
14, 1, 1),
SOC_DOUBLE_R("Out2 Playback Switch",
WM8350_LOUT2_VOLUME,
WM8350_ROUT2_VOLUME,
14, 1, 1),
};
/*
* DAPM Controls
*/
/* Left Playback Mixer */
static const struct snd_kcontrol_new wm8350_left_play_mixer_controls[] = {
SOC_DAPM_SINGLE("Playback Switch",
WM8350_LEFT_MIXER_CONTROL, 11, 1, 0),
SOC_DAPM_SINGLE("Left Bypass Switch",
WM8350_LEFT_MIXER_CONTROL, 2, 1, 0),
SOC_DAPM_SINGLE("Right Playback Switch",
WM8350_LEFT_MIXER_CONTROL, 12, 1, 0),
SOC_DAPM_SINGLE("Left Sidetone Switch",
WM8350_LEFT_MIXER_CONTROL, 0, 1, 0),
SOC_DAPM_SINGLE("Right Sidetone Switch",
WM8350_LEFT_MIXER_CONTROL, 1, 1, 0),
};
/* Right Playback Mixer */
static const struct snd_kcontrol_new wm8350_right_play_mixer_controls[] = {
SOC_DAPM_SINGLE("Playback Switch",
WM8350_RIGHT_MIXER_CONTROL, 12, 1, 0),
SOC_DAPM_SINGLE("Right Bypass Switch",
WM8350_RIGHT_MIXER_CONTROL, 3, 1, 0),
SOC_DAPM_SINGLE("Left Playback Switch",
WM8350_RIGHT_MIXER_CONTROL, 11, 1, 0),
SOC_DAPM_SINGLE("Left Sidetone Switch",
WM8350_RIGHT_MIXER_CONTROL, 0, 1, 0),
SOC_DAPM_SINGLE("Right Sidetone Switch",
WM8350_RIGHT_MIXER_CONTROL, 1, 1, 0),
};
/* Out4 Mixer */
static const struct snd_kcontrol_new wm8350_out4_mixer_controls[] = {
SOC_DAPM_SINGLE("Right Playback Switch",
WM8350_OUT4_MIXER_CONTROL, 12, 1, 0),
SOC_DAPM_SINGLE("Left Playback Switch",
WM8350_OUT4_MIXER_CONTROL, 11, 1, 0),
SOC_DAPM_SINGLE("Right Capture Switch",
WM8350_OUT4_MIXER_CONTROL, 9, 1, 0),
SOC_DAPM_SINGLE("Out3 Playback Switch",
WM8350_OUT4_MIXER_CONTROL, 2, 1, 0),
SOC_DAPM_SINGLE("Right Mixer Switch",
WM8350_OUT4_MIXER_CONTROL, 1, 1, 0),
SOC_DAPM_SINGLE("Left Mixer Switch",
WM8350_OUT4_MIXER_CONTROL, 0, 1, 0),
};
/* Out3 Mixer */
static const struct snd_kcontrol_new wm8350_out3_mixer_controls[] = {
SOC_DAPM_SINGLE("Left Playback Switch",
WM8350_OUT3_MIXER_CONTROL, 11, 1, 0),
SOC_DAPM_SINGLE("Left Capture Switch",
WM8350_OUT3_MIXER_CONTROL, 8, 1, 0),
SOC_DAPM_SINGLE("Out4 Playback Switch",
WM8350_OUT3_MIXER_CONTROL, 3, 1, 0),
SOC_DAPM_SINGLE("Left Mixer Switch",
WM8350_OUT3_MIXER_CONTROL, 0, 1, 0),
};
/* Left Input Mixer */
static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = {
SOC_DAPM_SINGLE_TLV("L2 Capture Volume",
WM8350_INPUT_MIXER_VOLUME_L, 1, 7, 0, out_mix_tlv),
SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv),
SOC_DAPM_SINGLE("PGA Capture Switch",
WM8350_LEFT_INPUT_VOLUME, 14, 1, 1),
};
/* Right Input Mixer */
static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = {
SOC_DAPM_SINGLE_TLV("L2 Capture Volume",
WM8350_INPUT_MIXER_VOLUME_R, 5, 7, 0, out_mix_tlv),
SOC_DAPM_SINGLE_TLV("L3 Capture Volume",
WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv),
SOC_DAPM_SINGLE("PGA Capture Switch",
WM8350_RIGHT_INPUT_VOLUME, 14, 1, 1),
};
/* Left Mic Mixer */
static const struct snd_kcontrol_new wm8350_left_mic_mixer_controls[] = {
SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 1, 1, 0),
SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 0, 1, 0),
SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 2, 1, 0),
};
/* Right Mic Mixer */
static const struct snd_kcontrol_new wm8350_right_mic_mixer_controls[] = {
SOC_DAPM_SINGLE("INN Capture Switch", WM8350_INPUT_CONTROL, 9, 1, 0),
SOC_DAPM_SINGLE("INP Capture Switch", WM8350_INPUT_CONTROL, 8, 1, 0),
SOC_DAPM_SINGLE("IN2 Capture Switch", WM8350_INPUT_CONTROL, 10, 1, 0),
};
/* Beep Switch */
static const struct snd_kcontrol_new wm8350_beep_switch_controls =
SOC_DAPM_SINGLE("Switch", WM8350_BEEP_VOLUME, 15, 1, 1);
/* Out4 Capture Mux */
static const struct snd_kcontrol_new wm8350_out4_capture_controls =
SOC_DAPM_ENUM("Route", wm8350_enum[7]);
static const struct snd_soc_dapm_widget wm8350_dapm_widgets[] = {
SND_SOC_DAPM_PGA("IN3R PGA", WM8350_POWER_MGMT_2, 11, 0, NULL, 0),
SND_SOC_DAPM_PGA("IN3L PGA", WM8350_POWER_MGMT_2, 10, 0, NULL, 0),
SND_SOC_DAPM_PGA_E("Right Out2 PGA", WM8350_POWER_MGMT_3, 3, 0, NULL,
0, pga_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Left Out2 PGA", WM8350_POWER_MGMT_3, 2, 0, NULL, 0,
pga_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Right Out1 PGA", WM8350_POWER_MGMT_3, 1, 0, NULL,
0, pga_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Left Out1 PGA", WM8350_POWER_MGMT_3, 0, 0, NULL, 0,
pga_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_MIXER("Right Capture Mixer", WM8350_POWER_MGMT_2,
7, 0, &wm8350_right_capt_mixer_controls[0],
ARRAY_SIZE(wm8350_right_capt_mixer_controls)),
SND_SOC_DAPM_MIXER("Left Capture Mixer", WM8350_POWER_MGMT_2,
6, 0, &wm8350_left_capt_mixer_controls[0],
ARRAY_SIZE(wm8350_left_capt_mixer_controls)),
SND_SOC_DAPM_MIXER("Out4 Mixer", WM8350_POWER_MGMT_2, 5, 0,
&wm8350_out4_mixer_controls[0],
ARRAY_SIZE(wm8350_out4_mixer_controls)),
SND_SOC_DAPM_MIXER("Out3 Mixer", WM8350_POWER_MGMT_2, 4, 0,
&wm8350_out3_mixer_controls[0],
ARRAY_SIZE(wm8350_out3_mixer_controls)),
SND_SOC_DAPM_MIXER("Right Playback Mixer", WM8350_POWER_MGMT_2, 1, 0,
&wm8350_right_play_mixer_controls[0],
ARRAY_SIZE(wm8350_right_play_mixer_controls)),
SND_SOC_DAPM_MIXER("Left Playback Mixer", WM8350_POWER_MGMT_2, 0, 0,
&wm8350_left_play_mixer_controls[0],
ARRAY_SIZE(wm8350_left_play_mixer_controls)),
SND_SOC_DAPM_MIXER("Left Mic Mixer", WM8350_POWER_MGMT_2, 8, 0,
&wm8350_left_mic_mixer_controls[0],
ARRAY_SIZE(wm8350_left_mic_mixer_controls)),
SND_SOC_DAPM_MIXER("Right Mic Mixer", WM8350_POWER_MGMT_2, 9, 0,
&wm8350_right_mic_mixer_controls[0],
ARRAY_SIZE(wm8350_right_mic_mixer_controls)),
/* virtual mixer for Beep and Out2R */
SND_SOC_DAPM_MIXER("Out2 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_SWITCH("Beep", WM8350_POWER_MGMT_3, 7, 0,
&wm8350_beep_switch_controls),
SND_SOC_DAPM_ADC("Right ADC", "Right Capture",
WM8350_POWER_MGMT_4, 3, 0),
SND_SOC_DAPM_ADC("Left ADC", "Left Capture",
WM8350_POWER_MGMT_4, 2, 0),
SND_SOC_DAPM_DAC("Right DAC", "Right Playback",
WM8350_POWER_MGMT_4, 5, 0),
SND_SOC_DAPM_DAC("Left DAC", "Left Playback",
WM8350_POWER_MGMT_4, 4, 0),
SND_SOC_DAPM_MICBIAS("Mic Bias", WM8350_POWER_MGMT_1, 4, 0),
SND_SOC_DAPM_MUX("Out4 Capture Channel", SND_SOC_NOPM, 0, 0,
&wm8350_out4_capture_controls),
SND_SOC_DAPM_OUTPUT("OUT1R"),
SND_SOC_DAPM_OUTPUT("OUT1L"),
SND_SOC_DAPM_OUTPUT("OUT2R"),
SND_SOC_DAPM_OUTPUT("OUT2L"),
SND_SOC_DAPM_OUTPUT("OUT3"),
SND_SOC_DAPM_OUTPUT("OUT4"),
SND_SOC_DAPM_INPUT("IN1RN"),
SND_SOC_DAPM_INPUT("IN1RP"),
SND_SOC_DAPM_INPUT("IN2R"),
SND_SOC_DAPM_INPUT("IN1LP"),
SND_SOC_DAPM_INPUT("IN1LN"),
SND_SOC_DAPM_INPUT("IN2L"),
SND_SOC_DAPM_INPUT("IN3R"),
SND_SOC_DAPM_INPUT("IN3L"),
};
static const struct snd_soc_dapm_route wm8350_dapm_routes[] = {
/* left playback mixer */
{"Left Playback Mixer", "Playback Switch", "Left DAC"},
{"Left Playback Mixer", "Left Bypass Switch", "IN3L PGA"},
{"Left Playback Mixer", "Right Playback Switch", "Right DAC"},
{"Left Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"},
{"Left Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"},
/* right playback mixer */
{"Right Playback Mixer", "Playback Switch", "Right DAC"},
{"Right Playback Mixer", "Right Bypass Switch", "IN3R PGA"},
{"Right Playback Mixer", "Left Playback Switch", "Left DAC"},
{"Right Playback Mixer", "Left Sidetone Switch", "Left Mic Mixer"},
{"Right Playback Mixer", "Right Sidetone Switch", "Right Mic Mixer"},
/* out4 playback mixer */
{"Out4 Mixer", "Right Playback Switch", "Right DAC"},
{"Out4 Mixer", "Left Playback Switch", "Left DAC"},
{"Out4 Mixer", "Right Capture Switch", "Right Capture Mixer"},
{"Out4 Mixer", "Out3 Playback Switch", "Out3 Mixer"},
{"Out4 Mixer", "Right Mixer Switch", "Right Playback Mixer"},
{"Out4 Mixer", "Left Mixer Switch", "Left Playback Mixer"},
{"OUT4", NULL, "Out4 Mixer"},
/* out3 playback mixer */
{"Out3 Mixer", "Left Playback Switch", "Left DAC"},
{"Out3 Mixer", "Left Capture Switch", "Left Capture Mixer"},
{"Out3 Mixer", "Left Mixer Switch", "Left Playback Mixer"},
{"Out3 Mixer", "Out4 Playback Switch", "Out4 Mixer"},
{"OUT3", NULL, "Out3 Mixer"},
/* out2 */
{"Right Out2 PGA", NULL, "Right Playback Mixer"},
{"Left Out2 PGA", NULL, "Left Playback Mixer"},
{"OUT2L", NULL, "Left Out2 PGA"},
{"OUT2R", NULL, "Right Out2 PGA"},
/* out1 */
{"Right Out1 PGA", NULL, "Right Playback Mixer"},
{"Left Out1 PGA", NULL, "Left Playback Mixer"},
{"OUT1L", NULL, "Left Out1 PGA"},
{"OUT1R", NULL, "Right Out1 PGA"},
/* ADCs */
{"Left ADC", NULL, "Left Capture Mixer"},
{"Right ADC", NULL, "Right Capture Mixer"},
/* Left capture mixer */
{"Left Capture Mixer", "L2 Capture Volume", "IN2L"},
{"Left Capture Mixer", "L3 Capture Volume", "IN3L PGA"},
{"Left Capture Mixer", "PGA Capture Switch", "Left Mic Mixer"},
{"Left Capture Mixer", NULL, "Out4 Capture Channel"},
/* Right capture mixer */
{"Right Capture Mixer", "L2 Capture Volume", "IN2R"},
{"Right Capture Mixer", "L3 Capture Volume", "IN3R PGA"},
{"Right Capture Mixer", "PGA Capture Switch", "Right Mic Mixer"},
{"Right Capture Mixer", NULL, "Out4 Capture Channel"},
/* L3 Inputs */
{"IN3L PGA", NULL, "IN3L"},
{"IN3R PGA", NULL, "IN3R"},
/* Left Mic mixer */
{"Left Mic Mixer", "INN Capture Switch", "IN1LN"},
{"Left Mic Mixer", "INP Capture Switch", "IN1LP"},
{"Left Mic Mixer", "IN2 Capture Switch", "IN2L"},
/* Right Mic mixer */
{"Right Mic Mixer", "INN Capture Switch", "IN1RN"},
{"Right Mic Mixer", "INP Capture Switch", "IN1RP"},
{"Right Mic Mixer", "IN2 Capture Switch", "IN2R"},
/* out 4 capture */
{"Out4 Capture Channel", NULL, "Out4 Mixer"},
/* Beep */
{"Beep", NULL, "IN3R PGA"},
};
static int wm8350_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_component *component = codec_dai->component;
struct wm8350_data *wm8350_data = snd_soc_component_get_drvdata(component);
struct wm8350 *wm8350 = wm8350_data->wm8350;
u16 fll_4;
switch (clk_id) {
case WM8350_MCLK_SEL_MCLK:
wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_1,
WM8350_MCLK_SEL);
break;
case WM8350_MCLK_SEL_PLL_MCLK:
case WM8350_MCLK_SEL_PLL_DAC:
case WM8350_MCLK_SEL_PLL_ADC:
case WM8350_MCLK_SEL_PLL_32K:
wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_1,
WM8350_MCLK_SEL);
fll_4 = snd_soc_component_read32(component, WM8350_FLL_CONTROL_4) &
~WM8350_FLL_CLK_SRC_MASK;
snd_soc_component_write(component, WM8350_FLL_CONTROL_4, fll_4 | clk_id);
break;
}
/* MCLK direction */
if (dir == SND_SOC_CLOCK_OUT)
wm8350_set_bits(wm8350, WM8350_CLOCK_CONTROL_2,
WM8350_MCLK_DIR);
else
wm8350_clear_bits(wm8350, WM8350_CLOCK_CONTROL_2,
WM8350_MCLK_DIR);
return 0;
}
static int wm8350_set_clkdiv(struct snd_soc_dai *codec_dai, int div_id, int div)
{
struct snd_soc_component *component = codec_dai->component;
u16 val;
switch (div_id) {
case WM8350_ADC_CLKDIV:
val = snd_soc_component_read32(component, WM8350_ADC_DIVIDER) &
~WM8350_ADC_CLKDIV_MASK;
snd_soc_component_write(component, WM8350_ADC_DIVIDER, val | div);
break;
case WM8350_DAC_CLKDIV:
val = snd_soc_component_read32(component, WM8350_DAC_CLOCK_CONTROL) &
~WM8350_DAC_CLKDIV_MASK;
snd_soc_component_write(component, WM8350_DAC_CLOCK_CONTROL, val | div);
break;
case WM8350_BCLK_CLKDIV:
val = snd_soc_component_read32(component, WM8350_CLOCK_CONTROL_1) &
~WM8350_BCLK_DIV_MASK;
snd_soc_component_write(component, WM8350_CLOCK_CONTROL_1, val | div);
break;
case WM8350_OPCLK_CLKDIV:
val = snd_soc_component_read32(component, WM8350_CLOCK_CONTROL_1) &
~WM8350_OPCLK_DIV_MASK;
snd_soc_component_write(component, WM8350_CLOCK_CONTROL_1, val | div);
break;
case WM8350_SYS_CLKDIV:
val = snd_soc_component_read32(component, WM8350_CLOCK_CONTROL_1) &
~WM8350_MCLK_DIV_MASK;
snd_soc_component_write(component, WM8350_CLOCK_CONTROL_1, val | div);
break;
case WM8350_DACLR_CLKDIV:
val = snd_soc_component_read32(component, WM8350_DAC_LR_RATE) &
~WM8350_DACLRC_RATE_MASK;
snd_soc_component_write(component, WM8350_DAC_LR_RATE, val | div);
break;
case WM8350_ADCLR_CLKDIV:
val = snd_soc_component_read32(component, WM8350_ADC_LR_RATE) &
~WM8350_ADCLRC_RATE_MASK;
snd_soc_component_write(component, WM8350_ADC_LR_RATE, val | div);
break;
default:
return -EINVAL;
}
return 0;
}
static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
{
struct snd_soc_component *component = codec_dai->component;
u16 iface = snd_soc_component_read32(component, WM8350_AI_FORMATING) &
~(WM8350_AIF_BCLK_INV | WM8350_AIF_LRCLK_INV | WM8350_AIF_FMT_MASK);
u16 master = snd_soc_component_read32(component, WM8350_AI_DAC_CONTROL) &
~WM8350_BCLK_MSTR;
u16 dac_lrc = snd_soc_component_read32(component, WM8350_DAC_LR_RATE) &
~WM8350_DACLRC_ENA;
u16 adc_lrc = snd_soc_component_read32(component, WM8350_ADC_LR_RATE) &
~WM8350_ADCLRC_ENA;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
master |= WM8350_BCLK_MSTR;
dac_lrc |= WM8350_DACLRC_ENA;
adc_lrc |= WM8350_ADCLRC_ENA;
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 |= 0x2 << 8;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
iface |= 0x1 << 8;
break;
case SND_SOC_DAIFMT_DSP_A:
iface |= 0x3 << 8;
break;
case SND_SOC_DAIFMT_DSP_B:
iface |= 0x3 << 8 | WM8350_AIF_LRCLK_INV;
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 |= WM8350_AIF_LRCLK_INV | WM8350_AIF_BCLK_INV;
break;
case SND_SOC_DAIFMT_IB_NF:
iface |= WM8350_AIF_BCLK_INV;
break;
case SND_SOC_DAIFMT_NB_IF:
iface |= WM8350_AIF_LRCLK_INV;
break;
default:
return -EINVAL;
}
snd_soc_component_write(component, WM8350_AI_FORMATING, iface);
snd_soc_component_write(component, WM8350_AI_DAC_CONTROL, master);
snd_soc_component_write(component, WM8350_DAC_LR_RATE, dac_lrc);
snd_soc_component_write(component, WM8350_ADC_LR_RATE, adc_lrc);
return 0;
}
static int wm8350_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *codec_dai)
{
struct snd_soc_component *component = codec_dai->component;
struct wm8350_data *wm8350_data = snd_soc_component_get_drvdata(component);
struct wm8350 *wm8350 = wm8350_data->wm8350;
u16 iface = snd_soc_component_read32(component, WM8350_AI_FORMATING) &
~WM8350_AIF_WL_MASK;
/* bit size */
switch (params_width(params)) {
case 16:
break;
case 20:
iface |= 0x1 << 10;
break;
case 24:
iface |= 0x2 << 10;
break;
case 32:
iface |= 0x3 << 10;
break;
}
snd_soc_component_write(component, WM8350_AI_FORMATING, iface);
/* The sloping stopband filter is recommended for use with
* lower sample rates to improve performance.
*/
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
if (params_rate(params) < 24000)
wm8350_set_bits(wm8350, WM8350_DAC_MUTE_VOLUME,
WM8350_DAC_SB_FILT);
else
wm8350_clear_bits(wm8350, WM8350_DAC_MUTE_VOLUME,
WM8350_DAC_SB_FILT);
}
return 0;
}
static int wm8350_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_component *component = dai->component;
unsigned int val;
if (mute)
val = WM8350_DAC_MUTE_ENA;
else
val = 0;
snd_soc_component_update_bits(component, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA, val);
return 0;
}
/* FLL divisors */
struct _fll_div {
int div; /* FLL_OUTDIV */
int n;
int k;
int ratio; /* FLL_FRATIO */
};
/* The size in bits of the fll divide multiplied by 10
* to allow rounding later */
#define FIXED_FLL_SIZE ((1 << 16) * 10)
static inline int fll_factors(struct _fll_div *fll_div, unsigned int input,
unsigned int output)
{
u64 Kpart;
unsigned int t1, t2, K, Nmod;
if (output >= 2815250 && output <= 3125000)
fll_div->div = 0x4;
else if (output >= 5625000 && output <= 6250000)
fll_div->div = 0x3;
else if (output >= 11250000 && output <= 12500000)
fll_div->div = 0x2;
else if (output >= 22500000 && output <= 25000000)
fll_div->div = 0x1;
else {
printk(KERN_ERR "wm8350: fll freq %d out of range\n", output);
return -EINVAL;
}
if (input > 48000)
fll_div->ratio = 1;
else
fll_div->ratio = 8;
t1 = output * (1 << (fll_div->div + 1));
t2 = input * fll_div->ratio;
fll_div->n = t1 / t2;
Nmod = t1 % t2;
if (Nmod) {
Kpart = FIXED_FLL_SIZE * (long long)Nmod;
do_div(Kpart, t2);
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;
fll_div->k = K;
} else
fll_div->k = 0;
return 0;
}
static int wm8350_set_fll(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 wm8350_data *priv = snd_soc_component_get_drvdata(component);
struct wm8350 *wm8350 = priv->wm8350;
struct _fll_div fll_div;
int ret = 0;
u16 fll_1, fll_4;
if (freq_in == priv->fll_freq_in && freq_out == priv->fll_freq_out)
return 0;
/* power down FLL - we need to do this for reconfiguration */
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4,
WM8350_FLL_ENA | WM8350_FLL_OSC_ENA);
if (freq_out == 0 || freq_in == 0)
return ret;
ret = fll_factors(&fll_div, freq_in, freq_out);
if (ret < 0)
return ret;
dev_dbg(wm8350->dev,
"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);
/* set up N.K & dividers */
fll_1 = snd_soc_component_read32(component, WM8350_FLL_CONTROL_1) &
~(WM8350_FLL_OUTDIV_MASK | WM8350_FLL_RSP_RATE_MASK | 0xc000);
snd_soc_component_write(component, WM8350_FLL_CONTROL_1,
fll_1 | (fll_div.div << 8) | 0x50);
snd_soc_component_write(component, WM8350_FLL_CONTROL_2,
(fll_div.ratio << 11) | (fll_div.
n & WM8350_FLL_N_MASK));
snd_soc_component_write(component, WM8350_FLL_CONTROL_3, fll_div.k);
fll_4 = snd_soc_component_read32(component, WM8350_FLL_CONTROL_4) &
~(WM8350_FLL_FRAC | WM8350_FLL_SLOW_LOCK_REF);
snd_soc_component_write(component, WM8350_FLL_CONTROL_4,
fll_4 | (fll_div.k ? WM8350_FLL_FRAC : 0) |
(fll_div.ratio == 8 ? WM8350_FLL_SLOW_LOCK_REF : 0));
/* power FLL on */
wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_OSC_ENA);
wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_FLL_ENA);
priv->fll_freq_out = freq_out;
priv->fll_freq_in = freq_in;
return 0;
}
static int wm8350_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
struct wm8350_data *priv = snd_soc_component_get_drvdata(component);
struct wm8350 *wm8350 = priv->wm8350;
struct wm8350_audio_platform_data *platform =
wm8350->codec.platform_data;
u16 pm1;
int ret;
switch (level) {
case SND_SOC_BIAS_ON:
pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
pm1 | WM8350_VMID_50K |
platform->codec_current_on << 14);
break;
case SND_SOC_BIAS_PREPARE:
pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1);
pm1 &= ~WM8350_VMID_MASK;
wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
pm1 | WM8350_VMID_50K);
break;
case SND_SOC_BIAS_STANDBY:
if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) {
ret = regulator_bulk_enable(ARRAY_SIZE(priv->supplies),
priv->supplies);
if (ret != 0)
return ret;
/* Enable the system clock */
wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4,
WM8350_SYSCLK_ENA);
/* mute DAC & outputs */
wm8350_set_bits(wm8350, WM8350_DAC_MUTE,
WM8350_DAC_MUTE_ENA);
/* discharge cap memory */
wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
platform->dis_out1 |
(platform->dis_out2 << 2) |
(platform->dis_out3 << 4) |
(platform->dis_out4 << 6));
/* wait for discharge */
schedule_timeout_interruptible(msecs_to_jiffies
(platform->
cap_discharge_msecs));
/* enable antipop */
wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
(platform->vmid_s_curve << 8));
/* ramp up vmid */
wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
(platform->
codec_current_charge << 14) |
WM8350_VMID_5K | WM8350_VMIDEN |
WM8350_VBUFEN);
/* wait for vmid */
schedule_timeout_interruptible(msecs_to_jiffies
(platform->
vmid_charge_msecs));
/* turn on vmid 300k */
pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
pm1 |= WM8350_VMID_300K |
(platform->codec_current_standby << 14);
wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
pm1);
/* enable analogue bias */
pm1 |= WM8350_BIASEN;
wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
/* disable antipop */
wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0);
} else {
/* turn on vmid 300k and reduce current */
pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
~(WM8350_VMID_MASK | WM8350_CODEC_ISEL_MASK);
wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
pm1 | WM8350_VMID_300K |
(platform->
codec_current_standby << 14));
}
break;
case SND_SOC_BIAS_OFF:
/* mute DAC & enable outputs */
wm8350_set_bits(wm8350, WM8350_DAC_MUTE, WM8350_DAC_MUTE_ENA);
wm8350_set_bits(wm8350, WM8350_POWER_MGMT_3,
WM8350_OUT1L_ENA | WM8350_OUT1R_ENA |
WM8350_OUT2L_ENA | WM8350_OUT2R_ENA);
/* enable anti pop S curve */
wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
(platform->vmid_s_curve << 8));
/* turn off vmid */
pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
~WM8350_VMIDEN;
wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
/* wait */
schedule_timeout_interruptible(msecs_to_jiffies
(platform->
vmid_discharge_msecs));
wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL,
(platform->vmid_s_curve << 8) |
platform->dis_out1 |
(platform->dis_out2 << 2) |
(platform->dis_out3 << 4) |
(platform->dis_out4 << 6));
/* turn off VBuf and drain */
pm1 = wm8350_reg_read(wm8350, WM8350_POWER_MGMT_1) &
~(WM8350_VBUFEN | WM8350_VMID_MASK);
wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1,
pm1 | WM8350_OUTPUT_DRAIN_EN);
/* wait */
schedule_timeout_interruptible(msecs_to_jiffies
(platform->drain_msecs));
pm1 &= ~WM8350_BIASEN;
wm8350_reg_write(wm8350, WM8350_POWER_MGMT_1, pm1);
/* disable anti-pop */
wm8350_reg_write(wm8350, WM8350_ANTI_POP_CONTROL, 0);
wm8350_clear_bits(wm8350, WM8350_LOUT1_VOLUME,
WM8350_OUT1L_ENA);
wm8350_clear_bits(wm8350, WM8350_ROUT1_VOLUME,
WM8350_OUT1R_ENA);
wm8350_clear_bits(wm8350, WM8350_LOUT2_VOLUME,
WM8350_OUT2L_ENA);
wm8350_clear_bits(wm8350, WM8350_ROUT2_VOLUME,
WM8350_OUT2R_ENA);
/* disable clock gen */
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4,
WM8350_SYSCLK_ENA);
regulator_bulk_disable(ARRAY_SIZE(priv->supplies),
priv->supplies);
break;
}
return 0;
}
static void wm8350_hp_work(struct wm8350_data *priv,
struct wm8350_jack_data *jack,
u16 mask)
{
struct wm8350 *wm8350 = priv->wm8350;
u16 reg;
int report;
reg = wm8350_reg_read(wm8350, WM8350_JACK_PIN_STATUS);
if (reg & mask)
report = jack->report;
else
report = 0;
snd_soc_jack_report(jack->jack, report, jack->report);
}
static void wm8350_hpl_work(struct work_struct *work)
{
struct wm8350_data *priv =
container_of(work, struct wm8350_data, hpl.work.work);
wm8350_hp_work(priv, &priv->hpl, WM8350_JACK_L_LVL);
}
static void wm8350_hpr_work(struct work_struct *work)
{
struct wm8350_data *priv =
container_of(work, struct wm8350_data, hpr.work.work);
wm8350_hp_work(priv, &priv->hpr, WM8350_JACK_R_LVL);
}
static irqreturn_t wm8350_hpl_jack_handler(int irq, void *data)
{
struct wm8350_data *priv = data;
struct wm8350 *wm8350 = priv->wm8350;
#ifndef CONFIG_SND_SOC_WM8350_MODULE
trace_snd_soc_jack_irq("WM8350 HPL");
#endif
if (device_may_wakeup(wm8350->dev))
pm_wakeup_event(wm8350->dev, 250);
queue_delayed_work(system_power_efficient_wq,
&priv->hpl.work, msecs_to_jiffies(200));
return IRQ_HANDLED;
}
static irqreturn_t wm8350_hpr_jack_handler(int irq, void *data)
{
struct wm8350_data *priv = data;
struct wm8350 *wm8350 = priv->wm8350;
#ifndef CONFIG_SND_SOC_WM8350_MODULE
trace_snd_soc_jack_irq("WM8350 HPR");
#endif
if (device_may_wakeup(wm8350->dev))
pm_wakeup_event(wm8350->dev, 250);
queue_delayed_work(system_power_efficient_wq,
&priv->hpr.work, msecs_to_jiffies(200));
return IRQ_HANDLED;
}
/**
* wm8350_hp_jack_detect - Enable headphone jack detection.
*
* @component: WM8350 component
* @which: left or right jack detect signal
* @jack: jack to report detection events on
* @report: value to report
*
* Enables the headphone jack detection of the WM8350. If no report
* is specified then detection is disabled.
*/
int wm8350_hp_jack_detect(struct snd_soc_component *component, enum wm8350_jack which,
struct snd_soc_jack *jack, int report)
{
struct wm8350_data *priv = snd_soc_component_get_drvdata(component);
struct wm8350 *wm8350 = priv->wm8350;
int ena;
switch (which) {
case WM8350_JDL:
priv->hpl.jack = jack;
priv->hpl.report = report;
ena = WM8350_JDL_ENA;
break;
case WM8350_JDR:
priv->hpr.jack = jack;
priv->hpr.report = report;
ena = WM8350_JDR_ENA;
break;
default:
return -EINVAL;
}
if (report) {
wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena);
} else {
wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, ena);
}
/* Sync status */
switch (which) {
case WM8350_JDL:
wm8350_hpl_jack_handler(0, priv);
break;
case WM8350_JDR:
wm8350_hpr_jack_handler(0, priv);
break;
}
return 0;
}
EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect);
static irqreturn_t wm8350_mic_handler(int irq, void *data)
{
struct wm8350_data *priv = data;
struct wm8350 *wm8350 = priv->wm8350;
u16 reg;
int report = 0;
#ifndef CONFIG_SND_SOC_WM8350_MODULE
trace_snd_soc_jack_irq("WM8350 mic");
#endif
reg = wm8350_reg_read(wm8350, WM8350_JACK_PIN_STATUS);
if (reg & WM8350_JACK_MICSCD_LVL)
report |= priv->mic.short_report;
if (reg & WM8350_JACK_MICSD_LVL)
report |= priv->mic.report;
snd_soc_jack_report(priv->mic.jack, report,
priv->mic.report | priv->mic.short_report);
return IRQ_HANDLED;
}
/**
* wm8350_mic_jack_detect - Enable microphone jack detection.
*
* @component: WM8350 component
* @jack: jack to report detection events on
* @detect_report: value to report when presence detected
* @short_report: value to report when microphone short detected
*
* Enables the microphone jack detection of the WM8350. If both reports
* are specified as zero then detection is disabled.
*/
int wm8350_mic_jack_detect(struct snd_soc_component *component,
struct snd_soc_jack *jack,
int detect_report, int short_report)
{
struct wm8350_data *priv = snd_soc_component_get_drvdata(component);
struct wm8350 *wm8350 = priv->wm8350;
priv->mic.jack = jack;
priv->mic.report = detect_report;
priv->mic.short_report = short_report;
if (detect_report || short_report) {
wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
wm8350_set_bits(wm8350, WM8350_POWER_MGMT_1,
WM8350_MIC_DET_ENA);
} else {
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_1,
WM8350_MIC_DET_ENA);
}
return 0;
}
EXPORT_SYMBOL_GPL(wm8350_mic_jack_detect);
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>
2010-03-17 20:15:21 +00:00
#define WM8350_RATES (SNDRV_PCM_RATE_8000_96000)
#define WM8350_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\
SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
static const struct snd_soc_dai_ops wm8350_dai_ops = {
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>
2010-03-17 20:15:21 +00:00
.hw_params = wm8350_pcm_hw_params,
.digital_mute = wm8350_mute,
.set_fmt = wm8350_set_dai_fmt,
.set_sysclk = wm8350_set_dai_sysclk,
.set_pll = wm8350_set_fll,
.set_clkdiv = wm8350_set_clkdiv,
};
static struct snd_soc_dai_driver wm8350_dai = {
.name = "wm8350-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = WM8350_RATES,
.formats = WM8350_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8350_RATES,
.formats = WM8350_FORMATS,
},
.ops = &wm8350_dai_ops,
};
static int wm8350_component_probe(struct snd_soc_component *component)
{
struct wm8350 *wm8350 = dev_get_platdata(component->dev);
struct wm8350_data *priv;
struct wm8350_output *out1;
struct wm8350_output *out2;
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>
2010-03-17 20:15:21 +00:00
int ret, i;
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>
2010-03-17 20:15:21 +00:00
if (wm8350->codec.platform_data == NULL) {
dev_err(component->dev, "No audio platform data supplied\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>
2010-03-17 20:15:21 +00:00
return -EINVAL;
}
priv = devm_kzalloc(component->dev, sizeof(struct wm8350_data),
GFP_KERNEL);
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>
2010-03-17 20:15:21 +00:00
if (priv == NULL)
return -ENOMEM;
snd_soc_component_init_regmap(component, wm8350->regmap);
snd_soc_component_set_drvdata(component, priv);
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>
2010-03-17 20:15:21 +00:00
priv->wm8350 = wm8350;
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>
2010-03-17 20:15:21 +00:00
for (i = 0; i < ARRAY_SIZE(supply_names); i++)
priv->supplies[i].supply = supply_names[i];
ret = devm_regulator_bulk_get(wm8350->dev, ARRAY_SIZE(priv->supplies),
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>
2010-03-17 20:15:21 +00:00
priv->supplies);
if (ret != 0)
return ret;
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>
2010-03-17 20:15:21 +00:00
/* Put the codec into reset if it wasn't already */
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
INIT_DELAYED_WORK(&priv->pga_work, wm8350_pga_work);
INIT_DELAYED_WORK(&priv->hpl.work, wm8350_hpl_work);
INIT_DELAYED_WORK(&priv->hpr.work, wm8350_hpr_work);
/* Enable the codec */
wm8350_set_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
/* Enable robust clocking mode in ADC */
snd_soc_component_write(component, WM8350_SECURITY, 0xa7);
snd_soc_component_write(component, 0xde, 0x13);
snd_soc_component_write(component, WM8350_SECURITY, 0);
/* read OUT1 & OUT2 volumes */
out1 = &priv->out1;
out2 = &priv->out2;
out1->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT1_VOLUME) &
WM8350_OUT1L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
out1->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT1_VOLUME) &
WM8350_OUT1R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
out2->left_vol = (wm8350_reg_read(wm8350, WM8350_LOUT2_VOLUME) &
WM8350_OUT2L_VOL_MASK) >> WM8350_OUT1L_VOL_SHIFT;
out2->right_vol = (wm8350_reg_read(wm8350, WM8350_ROUT2_VOLUME) &
WM8350_OUT2R_VOL_MASK) >> WM8350_OUT1R_VOL_SHIFT;
wm8350_reg_write(wm8350, WM8350_LOUT1_VOLUME, 0);
wm8350_reg_write(wm8350, WM8350_ROUT1_VOLUME, 0);
wm8350_reg_write(wm8350, WM8350_LOUT2_VOLUME, 0);
wm8350_reg_write(wm8350, WM8350_ROUT2_VOLUME, 0);
/* Latch VU bits & mute */
wm8350_set_bits(wm8350, WM8350_LOUT1_VOLUME,
WM8350_OUT1_VU | WM8350_OUT1L_MUTE);
wm8350_set_bits(wm8350, WM8350_LOUT2_VOLUME,
WM8350_OUT2_VU | WM8350_OUT2L_MUTE);
wm8350_set_bits(wm8350, WM8350_ROUT1_VOLUME,
WM8350_OUT1_VU | WM8350_OUT1R_MUTE);
wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
/* Make sure AIF tristating is disabled by default */
wm8350_clear_bits(wm8350, WM8350_AI_FORMATING, WM8350_AIF_TRI);
/* Make sure we've got a sane companding setup too */
wm8350_clear_bits(wm8350, WM8350_ADC_DAC_COMP,
WM8350_DAC_COMP | WM8350_LOOPBACK);
/* Make sure jack detect is disabled to start off with */
wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
WM8350_JDL_ENA | WM8350_JDR_ENA);
wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L,
wm8350_hpl_jack_handler, 0, "Left jack detect",
priv);
wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R,
wm8350_hpr_jack_handler, 0, "Right jack detect",
priv);
wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD,
wm8350_mic_handler, 0, "Microphone short", priv);
wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD,
wm8350_mic_handler, 0, "Microphone detect", priv);
return 0;
}
static void wm8350_component_remove(struct snd_soc_component *component)
{
struct wm8350_data *priv = snd_soc_component_get_drvdata(component);
struct wm8350 *wm8350 = dev_get_platdata(component->dev);
wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
WM8350_JDL_ENA | WM8350_JDR_ENA);
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_MICD, priv);
wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, priv);
wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, priv);
wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, priv);
priv->hpl.jack = NULL;
priv->hpr.jack = NULL;
priv->mic.jack = NULL;
cancel_delayed_work_sync(&priv->hpl.work);
cancel_delayed_work_sync(&priv->hpr.work);
/* if there was any work waiting then we run it now and
* wait for its completion */
flush_delayed_work(&priv->pga_work);
wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
}
static const struct snd_soc_component_driver soc_component_dev_wm8350 = {
.probe = wm8350_component_probe,
.remove = wm8350_component_remove,
.set_bias_level = wm8350_set_bias_level,
.controls = wm8350_snd_controls,
.num_controls = ARRAY_SIZE(wm8350_snd_controls),
.dapm_widgets = wm8350_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm8350_dapm_widgets),
.dapm_routes = wm8350_dapm_routes,
.num_dapm_routes = ARRAY_SIZE(wm8350_dapm_routes),
.suspend_bias_off = 1,
.idle_bias_on = 1,
.use_pmdown_time = 1,
.endianness = 1,
.non_legacy_dai_naming = 1,
};
static int wm8350_probe(struct platform_device *pdev)
{
return devm_snd_soc_register_component(&pdev->dev,
&soc_component_dev_wm8350,
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>
2010-03-17 20:15:21 +00:00
&wm8350_dai, 1);
}
static struct platform_driver wm8350_codec_driver = {
.driver = {
.name = "wm8350-codec",
},
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>
2010-03-17 20:15:21 +00:00
.probe = wm8350_probe,
};
module_platform_driver(wm8350_codec_driver);
MODULE_DESCRIPTION("ASoC WM8350 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:wm8350-codec");