OpenCloudOS-Kernel/sound/soc/codecs/wm8994.c

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/*
* wm8994.c -- WM8994 ALSA SoC Audio driver
*
* Copyright 2009-12 Wolfson Microelectronics plc
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/gcd.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/pm_runtime.h>
#include <linux/regulator/consumer.h>
include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo <tj@kernel.org> Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org> Cc: Ingo Molnar <mingo@redhat.com> Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
2010-03-24 16:04:11 +08:00
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/jack.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 <linux/mfd/wm8994/core.h>
#include <linux/mfd/wm8994/registers.h>
#include <linux/mfd/wm8994/pdata.h>
#include <linux/mfd/wm8994/gpio.h>
#include "wm8994.h"
#include "wm_hubs.h"
#define WM1811_JACKDET_MODE_NONE 0x0000
#define WM1811_JACKDET_MODE_JACK 0x0100
#define WM1811_JACKDET_MODE_MIC 0x0080
#define WM1811_JACKDET_MODE_AUDIO 0x0180
#define WM8994_NUM_DRC 3
#define WM8994_NUM_EQ 3
static struct {
unsigned int reg;
unsigned int mask;
} wm8994_vu_bits[] = {
{ WM8994_LEFT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU },
{ WM8994_RIGHT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU },
{ WM8994_LEFT_LINE_INPUT_3_4_VOLUME, WM8994_IN2_VU },
{ WM8994_RIGHT_LINE_INPUT_3_4_VOLUME, WM8994_IN2_VU },
{ WM8994_SPEAKER_VOLUME_LEFT, WM8994_SPKOUT_VU },
{ WM8994_SPEAKER_VOLUME_RIGHT, WM8994_SPKOUT_VU },
{ WM8994_LEFT_OUTPUT_VOLUME, WM8994_HPOUT1_VU },
{ WM8994_RIGHT_OUTPUT_VOLUME, WM8994_HPOUT1_VU },
{ WM8994_LEFT_OPGA_VOLUME, WM8994_MIXOUT_VU },
{ WM8994_RIGHT_OPGA_VOLUME, WM8994_MIXOUT_VU },
{ WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1DAC1_VU },
{ WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU },
{ WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU },
{ WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU },
{ WM8994_AIF2_DAC_LEFT_VOLUME, WM8994_AIF2DAC_VU },
{ WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU },
{ WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1ADC1_VU },
{ WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU },
{ WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU },
{ WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU },
{ WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2ADC_VU },
{ WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF1ADC2_VU },
{ WM8994_DAC1_LEFT_VOLUME, WM8994_DAC1_VU },
{ WM8994_DAC1_RIGHT_VOLUME, WM8994_DAC1_VU },
{ WM8994_DAC2_LEFT_VOLUME, WM8994_DAC2_VU },
{ WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU },
};
static int wm8994_drc_base[] = {
WM8994_AIF1_DRC1_1,
WM8994_AIF1_DRC2_1,
WM8994_AIF2_DRC_1,
};
static int wm8994_retune_mobile_base[] = {
WM8994_AIF1_DAC1_EQ_GAINS_1,
WM8994_AIF1_DAC2_EQ_GAINS_1,
WM8994_AIF2_EQ_GAINS_1,
};
static const struct wm8958_micd_rate micdet_rates[] = {
{ 32768, true, 1, 4 },
{ 32768, false, 1, 1 },
{ 44100 * 256, true, 7, 10 },
{ 44100 * 256, false, 7, 10 },
};
static const struct wm8958_micd_rate jackdet_rates[] = {
{ 32768, true, 0, 1 },
{ 32768, false, 0, 1 },
{ 44100 * 256, true, 10, 10 },
{ 44100 * 256, false, 7, 8 },
};
static void wm8958_micd_set_rate(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
int best, i, sysclk, val;
bool idle;
const struct wm8958_micd_rate *rates;
int num_rates;
idle = !wm8994->jack_mic;
sysclk = snd_soc_read(codec, WM8994_CLOCKING_1);
if (sysclk & WM8994_SYSCLK_SRC)
sysclk = wm8994->aifclk[1];
else
sysclk = wm8994->aifclk[0];
if (control->pdata.micd_rates) {
rates = control->pdata.micd_rates;
num_rates = control->pdata.num_micd_rates;
} else if (wm8994->jackdet) {
rates = jackdet_rates;
num_rates = ARRAY_SIZE(jackdet_rates);
} else {
rates = micdet_rates;
num_rates = ARRAY_SIZE(micdet_rates);
}
best = 0;
for (i = 0; i < num_rates; i++) {
if (rates[i].idle != idle)
continue;
if (abs(rates[i].sysclk - sysclk) <
abs(rates[best].sysclk - sysclk))
best = i;
else if (rates[best].idle != idle)
best = i;
}
val = rates[best].start << WM8958_MICD_BIAS_STARTTIME_SHIFT
| rates[best].rate << WM8958_MICD_RATE_SHIFT;
dev_dbg(codec->dev, "MICD rate %d,%d for %dHz %s\n",
rates[best].start, rates[best].rate, sysclk,
idle ? "idle" : "active");
snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
WM8958_MICD_BIAS_STARTTIME_MASK |
WM8958_MICD_RATE_MASK, val);
}
static int configure_aif_clock(struct snd_soc_codec *codec, int aif)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int rate;
int reg1 = 0;
int offset;
if (aif)
offset = 4;
else
offset = 0;
switch (wm8994->sysclk[aif]) {
case WM8994_SYSCLK_MCLK1:
rate = wm8994->mclk[0];
break;
case WM8994_SYSCLK_MCLK2:
reg1 |= 0x8;
rate = wm8994->mclk[1];
break;
case WM8994_SYSCLK_FLL1:
reg1 |= 0x10;
rate = wm8994->fll[0].out;
break;
case WM8994_SYSCLK_FLL2:
reg1 |= 0x18;
rate = wm8994->fll[1].out;
break;
default:
return -EINVAL;
}
if (rate >= 13500000) {
rate /= 2;
reg1 |= WM8994_AIF1CLK_DIV;
dev_dbg(codec->dev, "Dividing AIF%d clock to %dHz\n",
aif + 1, rate);
}
wm8994->aifclk[aif] = rate;
snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1 + offset,
WM8994_AIF1CLK_SRC_MASK | WM8994_AIF1CLK_DIV,
reg1);
return 0;
}
static int configure_clock(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int change, new;
/* Bring up the AIF clocks first */
configure_aif_clock(codec, 0);
configure_aif_clock(codec, 1);
/* Then switch CLK_SYS over to the higher of them; a change
* can only happen as a result of a clocking change which can
* only be made outside of DAPM so we can safely redo the
* clocking.
*/
/* If they're equal it doesn't matter which is used */
if (wm8994->aifclk[0] == wm8994->aifclk[1]) {
wm8958_micd_set_rate(codec);
return 0;
}
if (wm8994->aifclk[0] < wm8994->aifclk[1])
new = WM8994_SYSCLK_SRC;
else
new = 0;
change = snd_soc_update_bits(codec, WM8994_CLOCKING_1,
WM8994_SYSCLK_SRC, new);
if (change)
snd_soc_dapm_sync(&codec->dapm);
wm8958_micd_set_rate(codec);
return 0;
}
static int check_clk_sys(struct snd_soc_dapm_widget *source,
struct snd_soc_dapm_widget *sink)
{
int reg = snd_soc_read(source->codec, WM8994_CLOCKING_1);
const char *clk;
/* Check what we're currently using for CLK_SYS */
if (reg & WM8994_SYSCLK_SRC)
clk = "AIF2CLK";
else
clk = "AIF1CLK";
return strcmp(source->name, clk) == 0;
}
static const char *sidetone_hpf_text[] = {
"2.7kHz", "1.35kHz", "675Hz", "370Hz", "180Hz", "90Hz", "45Hz"
};
static const struct soc_enum sidetone_hpf =
SOC_ENUM_SINGLE(WM8994_SIDETONE, 7, 7, sidetone_hpf_text);
static const char *adc_hpf_text[] = {
"HiFi", "Voice 1", "Voice 2", "Voice 3"
};
static const struct soc_enum aif1adc1_hpf =
SOC_ENUM_SINGLE(WM8994_AIF1_ADC1_FILTERS, 13, 4, adc_hpf_text);
static const struct soc_enum aif1adc2_hpf =
SOC_ENUM_SINGLE(WM8994_AIF1_ADC2_FILTERS, 13, 4, adc_hpf_text);
static const struct soc_enum aif2adc_hpf =
SOC_ENUM_SINGLE(WM8994_AIF2_ADC_FILTERS, 13, 4, adc_hpf_text);
static const DECLARE_TLV_DB_SCALE(aif_tlv, 0, 600, 0);
static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
static const DECLARE_TLV_DB_SCALE(st_tlv, -3600, 300, 0);
static const DECLARE_TLV_DB_SCALE(wm8994_3d_tlv, -1600, 183, 0);
static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
static const DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0);
static const DECLARE_TLV_DB_SCALE(mixin_boost_tlv, 0, 900, 0);
#define WM8994_DRC_SWITCH(xname, reg, shift) \
SOC_SINGLE_EXT(xname, reg, shift, 1, 0, \
snd_soc_get_volsw, wm8994_put_drc_sw)
static int wm8994_put_drc_sw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
int mask, ret;
/* Can't enable both ADC and DAC paths simultaneously */
if (mc->shift == WM8994_AIF1DAC1_DRC_ENA_SHIFT)
mask = WM8994_AIF1ADC1L_DRC_ENA_MASK |
WM8994_AIF1ADC1R_DRC_ENA_MASK;
else
mask = WM8994_AIF1DAC1_DRC_ENA_MASK;
ret = snd_soc_read(codec, mc->reg);
if (ret < 0)
return ret;
if (ret & mask)
return -EINVAL;
return snd_soc_put_volsw(kcontrol, ucontrol);
}
static void wm8994_set_drc(struct snd_soc_codec *codec, int drc)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
struct wm8994_pdata *pdata = &control->pdata;
int base = wm8994_drc_base[drc];
int cfg = wm8994->drc_cfg[drc];
int save, i;
/* Save any enables; the configuration should clear them. */
save = snd_soc_read(codec, base);
save &= WM8994_AIF1DAC1_DRC_ENA | WM8994_AIF1ADC1L_DRC_ENA |
WM8994_AIF1ADC1R_DRC_ENA;
for (i = 0; i < WM8994_DRC_REGS; i++)
snd_soc_update_bits(codec, base + i, 0xffff,
pdata->drc_cfgs[cfg].regs[i]);
snd_soc_update_bits(codec, base, WM8994_AIF1DAC1_DRC_ENA |
WM8994_AIF1ADC1L_DRC_ENA |
WM8994_AIF1ADC1R_DRC_ENA, save);
}
/* Icky as hell but saves code duplication */
static int wm8994_get_drc(const char *name)
{
if (strcmp(name, "AIF1DRC1 Mode") == 0)
return 0;
if (strcmp(name, "AIF1DRC2 Mode") == 0)
return 1;
if (strcmp(name, "AIF2DRC Mode") == 0)
return 2;
return -EINVAL;
}
static int wm8994_put_drc_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
struct wm8994_pdata *pdata = &control->pdata;
int drc = wm8994_get_drc(kcontrol->id.name);
int value = ucontrol->value.integer.value[0];
if (drc < 0)
return drc;
if (value >= pdata->num_drc_cfgs)
return -EINVAL;
wm8994->drc_cfg[drc] = value;
wm8994_set_drc(codec, drc);
return 0;
}
static int wm8994_get_drc_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int drc = wm8994_get_drc(kcontrol->id.name);
if (drc < 0)
return drc;
ucontrol->value.enumerated.item[0] = wm8994->drc_cfg[drc];
return 0;
}
static void wm8994_set_retune_mobile(struct snd_soc_codec *codec, int block)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
struct wm8994_pdata *pdata = &control->pdata;
int base = wm8994_retune_mobile_base[block];
int iface, best, best_val, save, i, cfg;
if (!pdata || !wm8994->num_retune_mobile_texts)
return;
switch (block) {
case 0:
case 1:
iface = 0;
break;
case 2:
iface = 1;
break;
default:
return;
}
/* Find the version of the currently selected configuration
* with the nearest sample rate. */
cfg = wm8994->retune_mobile_cfg[block];
best = 0;
best_val = INT_MAX;
for (i = 0; i < pdata->num_retune_mobile_cfgs; i++) {
if (strcmp(pdata->retune_mobile_cfgs[i].name,
wm8994->retune_mobile_texts[cfg]) == 0 &&
abs(pdata->retune_mobile_cfgs[i].rate
- wm8994->dac_rates[iface]) < best_val) {
best = i;
best_val = abs(pdata->retune_mobile_cfgs[i].rate
- wm8994->dac_rates[iface]);
}
}
dev_dbg(codec->dev, "ReTune Mobile %d %s/%dHz for %dHz sample rate\n",
block,
pdata->retune_mobile_cfgs[best].name,
pdata->retune_mobile_cfgs[best].rate,
wm8994->dac_rates[iface]);
/* The EQ will be disabled while reconfiguring it, remember the
* current configuration.
*/
save = snd_soc_read(codec, base);
save &= WM8994_AIF1DAC1_EQ_ENA;
for (i = 0; i < WM8994_EQ_REGS; i++)
snd_soc_update_bits(codec, base + i, 0xffff,
pdata->retune_mobile_cfgs[best].regs[i]);
snd_soc_update_bits(codec, base, WM8994_AIF1DAC1_EQ_ENA, save);
}
/* Icky as hell but saves code duplication */
static int wm8994_get_retune_mobile_block(const char *name)
{
if (strcmp(name, "AIF1.1 EQ Mode") == 0)
return 0;
if (strcmp(name, "AIF1.2 EQ Mode") == 0)
return 1;
if (strcmp(name, "AIF2 EQ Mode") == 0)
return 2;
return -EINVAL;
}
static int wm8994_put_retune_mobile_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
struct wm8994_pdata *pdata = &control->pdata;
int block = wm8994_get_retune_mobile_block(kcontrol->id.name);
int value = ucontrol->value.integer.value[0];
if (block < 0)
return block;
if (value >= pdata->num_retune_mobile_cfgs)
return -EINVAL;
wm8994->retune_mobile_cfg[block] = value;
wm8994_set_retune_mobile(codec, block);
return 0;
}
static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int block = wm8994_get_retune_mobile_block(kcontrol->id.name);
if (block < 0)
return block;
ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block];
return 0;
}
static const char *aif_chan_src_text[] = {
"Left", "Right"
};
static const struct soc_enum aif1adcl_src =
SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_1, 15, 2, aif_chan_src_text);
static const struct soc_enum aif1adcr_src =
SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_1, 14, 2, aif_chan_src_text);
static const struct soc_enum aif2adcl_src =
SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_1, 15, 2, aif_chan_src_text);
static const struct soc_enum aif2adcr_src =
SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_1, 14, 2, aif_chan_src_text);
static const struct soc_enum aif1dacl_src =
SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 15, 2, aif_chan_src_text);
static const struct soc_enum aif1dacr_src =
SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 14, 2, aif_chan_src_text);
static const struct soc_enum aif2dacl_src =
SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 15, 2, aif_chan_src_text);
static const struct soc_enum aif2dacr_src =
SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 14, 2, aif_chan_src_text);
static const char *osr_text[] = {
"Low Power", "High Performance",
};
static const struct soc_enum dac_osr =
SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 0, 2, osr_text);
static const struct soc_enum adc_osr =
SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 1, 2, osr_text);
static const struct snd_kcontrol_new wm8994_snd_controls[] = {
SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME,
WM8994_AIF1_ADC1_RIGHT_VOLUME,
1, 119, 0, digital_tlv),
SOC_DOUBLE_R_TLV("AIF1ADC2 Volume", WM8994_AIF1_ADC2_LEFT_VOLUME,
WM8994_AIF1_ADC2_RIGHT_VOLUME,
1, 119, 0, digital_tlv),
SOC_DOUBLE_R_TLV("AIF2ADC Volume", WM8994_AIF2_ADC_LEFT_VOLUME,
WM8994_AIF2_ADC_RIGHT_VOLUME,
1, 119, 0, digital_tlv),
SOC_ENUM("AIF1ADCL Source", aif1adcl_src),
SOC_ENUM("AIF1ADCR Source", aif1adcr_src),
SOC_ENUM("AIF2ADCL Source", aif2adcl_src),
SOC_ENUM("AIF2ADCR Source", aif2adcr_src),
SOC_ENUM("AIF1DACL Source", aif1dacl_src),
SOC_ENUM("AIF1DACR Source", aif1dacr_src),
SOC_ENUM("AIF2DACL Source", aif2dacl_src),
SOC_ENUM("AIF2DACR Source", aif2dacr_src),
SOC_DOUBLE_R_TLV("AIF1DAC1 Volume", WM8994_AIF1_DAC1_LEFT_VOLUME,
WM8994_AIF1_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
SOC_DOUBLE_R_TLV("AIF1DAC2 Volume", WM8994_AIF1_DAC2_LEFT_VOLUME,
WM8994_AIF1_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
SOC_DOUBLE_R_TLV("AIF2DAC Volume", WM8994_AIF2_DAC_LEFT_VOLUME,
WM8994_AIF2_DAC_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
SOC_SINGLE_TLV("AIF1 Boost Volume", WM8994_AIF1_CONTROL_2, 10, 3, 0, aif_tlv),
SOC_SINGLE_TLV("AIF2 Boost Volume", WM8994_AIF2_CONTROL_2, 10, 3, 0, aif_tlv),
SOC_SINGLE("AIF1DAC1 EQ Switch", WM8994_AIF1_DAC1_EQ_GAINS_1, 0, 1, 0),
SOC_SINGLE("AIF1DAC2 EQ Switch", WM8994_AIF1_DAC2_EQ_GAINS_1, 0, 1, 0),
SOC_SINGLE("AIF2 EQ Switch", WM8994_AIF2_EQ_GAINS_1, 0, 1, 0),
WM8994_DRC_SWITCH("AIF1DAC1 DRC Switch", WM8994_AIF1_DRC1_1, 2),
WM8994_DRC_SWITCH("AIF1ADC1L DRC Switch", WM8994_AIF1_DRC1_1, 1),
WM8994_DRC_SWITCH("AIF1ADC1R DRC Switch", WM8994_AIF1_DRC1_1, 0),
WM8994_DRC_SWITCH("AIF1DAC2 DRC Switch", WM8994_AIF1_DRC2_1, 2),
WM8994_DRC_SWITCH("AIF1ADC2L DRC Switch", WM8994_AIF1_DRC2_1, 1),
WM8994_DRC_SWITCH("AIF1ADC2R DRC Switch", WM8994_AIF1_DRC2_1, 0),
WM8994_DRC_SWITCH("AIF2DAC DRC Switch", WM8994_AIF2_DRC_1, 2),
WM8994_DRC_SWITCH("AIF2ADCL DRC Switch", WM8994_AIF2_DRC_1, 1),
WM8994_DRC_SWITCH("AIF2ADCR DRC Switch", WM8994_AIF2_DRC_1, 0),
SOC_SINGLE_TLV("DAC1 Right Sidetone Volume", WM8994_DAC1_MIXER_VOLUMES,
5, 12, 0, st_tlv),
SOC_SINGLE_TLV("DAC1 Left Sidetone Volume", WM8994_DAC1_MIXER_VOLUMES,
0, 12, 0, st_tlv),
SOC_SINGLE_TLV("DAC2 Right Sidetone Volume", WM8994_DAC2_MIXER_VOLUMES,
5, 12, 0, st_tlv),
SOC_SINGLE_TLV("DAC2 Left Sidetone Volume", WM8994_DAC2_MIXER_VOLUMES,
0, 12, 0, st_tlv),
SOC_ENUM("Sidetone HPF Mux", sidetone_hpf),
SOC_SINGLE("Sidetone HPF Switch", WM8994_SIDETONE, 6, 1, 0),
SOC_ENUM("AIF1ADC1 HPF Mode", aif1adc1_hpf),
SOC_DOUBLE("AIF1ADC1 HPF Switch", WM8994_AIF1_ADC1_FILTERS, 12, 11, 1, 0),
SOC_ENUM("AIF1ADC2 HPF Mode", aif1adc2_hpf),
SOC_DOUBLE("AIF1ADC2 HPF Switch", WM8994_AIF1_ADC2_FILTERS, 12, 11, 1, 0),
SOC_ENUM("AIF2ADC HPF Mode", aif2adc_hpf),
SOC_DOUBLE("AIF2ADC HPF Switch", WM8994_AIF2_ADC_FILTERS, 12, 11, 1, 0),
SOC_ENUM("ADC OSR", adc_osr),
SOC_ENUM("DAC OSR", dac_osr),
SOC_DOUBLE_R_TLV("DAC1 Volume", WM8994_DAC1_LEFT_VOLUME,
WM8994_DAC1_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
SOC_DOUBLE_R("DAC1 Switch", WM8994_DAC1_LEFT_VOLUME,
WM8994_DAC1_RIGHT_VOLUME, 9, 1, 1),
SOC_DOUBLE_R_TLV("DAC2 Volume", WM8994_DAC2_LEFT_VOLUME,
WM8994_DAC2_RIGHT_VOLUME, 1, 96, 0, digital_tlv),
SOC_DOUBLE_R("DAC2 Switch", WM8994_DAC2_LEFT_VOLUME,
WM8994_DAC2_RIGHT_VOLUME, 9, 1, 1),
SOC_SINGLE_TLV("SPKL DAC2 Volume", WM8994_SPKMIXL_ATTENUATION,
6, 1, 1, wm_hubs_spkmix_tlv),
SOC_SINGLE_TLV("SPKL DAC1 Volume", WM8994_SPKMIXL_ATTENUATION,
2, 1, 1, wm_hubs_spkmix_tlv),
SOC_SINGLE_TLV("SPKR DAC2 Volume", WM8994_SPKMIXR_ATTENUATION,
6, 1, 1, wm_hubs_spkmix_tlv),
SOC_SINGLE_TLV("SPKR DAC1 Volume", WM8994_SPKMIXR_ATTENUATION,
2, 1, 1, wm_hubs_spkmix_tlv),
SOC_SINGLE_TLV("AIF1DAC1 3D Stereo Volume", WM8994_AIF1_DAC1_FILTERS_2,
10, 15, 0, wm8994_3d_tlv),
SOC_SINGLE("AIF1DAC1 3D Stereo Switch", WM8994_AIF1_DAC1_FILTERS_2,
8, 1, 0),
SOC_SINGLE_TLV("AIF1DAC2 3D Stereo Volume", WM8994_AIF1_DAC2_FILTERS_2,
10, 15, 0, wm8994_3d_tlv),
SOC_SINGLE("AIF1DAC2 3D Stereo Switch", WM8994_AIF1_DAC2_FILTERS_2,
8, 1, 0),
SOC_SINGLE_TLV("AIF2DAC 3D Stereo Volume", WM8994_AIF2_DAC_FILTERS_2,
10, 15, 0, wm8994_3d_tlv),
SOC_SINGLE("AIF2DAC 3D Stereo Switch", WM8994_AIF2_DAC_FILTERS_2,
8, 1, 0),
};
static const struct snd_kcontrol_new wm8994_eq_controls[] = {
SOC_SINGLE_TLV("AIF1DAC1 EQ1 Volume", WM8994_AIF1_DAC1_EQ_GAINS_1, 11, 31, 0,
eq_tlv),
SOC_SINGLE_TLV("AIF1DAC1 EQ2 Volume", WM8994_AIF1_DAC1_EQ_GAINS_1, 6, 31, 0,
eq_tlv),
SOC_SINGLE_TLV("AIF1DAC1 EQ3 Volume", WM8994_AIF1_DAC1_EQ_GAINS_1, 1, 31, 0,
eq_tlv),
SOC_SINGLE_TLV("AIF1DAC1 EQ4 Volume", WM8994_AIF1_DAC1_EQ_GAINS_2, 11, 31, 0,
eq_tlv),
SOC_SINGLE_TLV("AIF1DAC1 EQ5 Volume", WM8994_AIF1_DAC1_EQ_GAINS_2, 6, 31, 0,
eq_tlv),
SOC_SINGLE_TLV("AIF1DAC2 EQ1 Volume", WM8994_AIF1_DAC2_EQ_GAINS_1, 11, 31, 0,
eq_tlv),
SOC_SINGLE_TLV("AIF1DAC2 EQ2 Volume", WM8994_AIF1_DAC2_EQ_GAINS_1, 6, 31, 0,
eq_tlv),
SOC_SINGLE_TLV("AIF1DAC2 EQ3 Volume", WM8994_AIF1_DAC2_EQ_GAINS_1, 1, 31, 0,
eq_tlv),
SOC_SINGLE_TLV("AIF1DAC2 EQ4 Volume", WM8994_AIF1_DAC2_EQ_GAINS_2, 11, 31, 0,
eq_tlv),
SOC_SINGLE_TLV("AIF1DAC2 EQ5 Volume", WM8994_AIF1_DAC2_EQ_GAINS_2, 6, 31, 0,
eq_tlv),
SOC_SINGLE_TLV("AIF2 EQ1 Volume", WM8994_AIF2_EQ_GAINS_1, 11, 31, 0,
eq_tlv),
SOC_SINGLE_TLV("AIF2 EQ2 Volume", WM8994_AIF2_EQ_GAINS_1, 6, 31, 0,
eq_tlv),
SOC_SINGLE_TLV("AIF2 EQ3 Volume", WM8994_AIF2_EQ_GAINS_1, 1, 31, 0,
eq_tlv),
SOC_SINGLE_TLV("AIF2 EQ4 Volume", WM8994_AIF2_EQ_GAINS_2, 11, 31, 0,
eq_tlv),
SOC_SINGLE_TLV("AIF2 EQ5 Volume", WM8994_AIF2_EQ_GAINS_2, 6, 31, 0,
eq_tlv),
};
static const struct snd_kcontrol_new wm8994_drc_controls[] = {
SND_SOC_BYTES_MASK("AIF1.1 DRC", WM8994_AIF1_DRC1_1, 5,
WM8994_AIF1DAC1_DRC_ENA | WM8994_AIF1ADC1L_DRC_ENA |
WM8994_AIF1ADC1R_DRC_ENA),
SND_SOC_BYTES_MASK("AIF1.2 DRC", WM8994_AIF1_DRC2_1, 5,
WM8994_AIF1DAC2_DRC_ENA | WM8994_AIF1ADC2L_DRC_ENA |
WM8994_AIF1ADC2R_DRC_ENA),
SND_SOC_BYTES_MASK("AIF2 DRC", WM8994_AIF2_DRC_1, 5,
WM8994_AIF2DAC_DRC_ENA | WM8994_AIF2ADCL_DRC_ENA |
WM8994_AIF2ADCR_DRC_ENA),
};
static const char *wm8958_ng_text[] = {
"30ms", "125ms", "250ms", "500ms",
};
static const struct soc_enum wm8958_aif1dac1_ng_hold =
SOC_ENUM_SINGLE(WM8958_AIF1_DAC1_NOISE_GATE,
WM8958_AIF1DAC1_NG_THR_SHIFT, 4, wm8958_ng_text);
static const struct soc_enum wm8958_aif1dac2_ng_hold =
SOC_ENUM_SINGLE(WM8958_AIF1_DAC2_NOISE_GATE,
WM8958_AIF1DAC2_NG_THR_SHIFT, 4, wm8958_ng_text);
static const struct soc_enum wm8958_aif2dac_ng_hold =
SOC_ENUM_SINGLE(WM8958_AIF2_DAC_NOISE_GATE,
WM8958_AIF2DAC_NG_THR_SHIFT, 4, wm8958_ng_text);
static const struct snd_kcontrol_new wm8958_snd_controls[] = {
SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv),
SOC_SINGLE("AIF1DAC1 Noise Gate Switch", WM8958_AIF1_DAC1_NOISE_GATE,
WM8958_AIF1DAC1_NG_ENA_SHIFT, 1, 0),
SOC_ENUM("AIF1DAC1 Noise Gate Hold Time", wm8958_aif1dac1_ng_hold),
SOC_SINGLE_TLV("AIF1DAC1 Noise Gate Threshold Volume",
WM8958_AIF1_DAC1_NOISE_GATE, WM8958_AIF1DAC1_NG_THR_SHIFT,
7, 1, ng_tlv),
SOC_SINGLE("AIF1DAC2 Noise Gate Switch", WM8958_AIF1_DAC2_NOISE_GATE,
WM8958_AIF1DAC2_NG_ENA_SHIFT, 1, 0),
SOC_ENUM("AIF1DAC2 Noise Gate Hold Time", wm8958_aif1dac2_ng_hold),
SOC_SINGLE_TLV("AIF1DAC2 Noise Gate Threshold Volume",
WM8958_AIF1_DAC2_NOISE_GATE, WM8958_AIF1DAC2_NG_THR_SHIFT,
7, 1, ng_tlv),
SOC_SINGLE("AIF2DAC Noise Gate Switch", WM8958_AIF2_DAC_NOISE_GATE,
WM8958_AIF2DAC_NG_ENA_SHIFT, 1, 0),
SOC_ENUM("AIF2DAC Noise Gate Hold Time", wm8958_aif2dac_ng_hold),
SOC_SINGLE_TLV("AIF2DAC Noise Gate Threshold Volume",
WM8958_AIF2_DAC_NOISE_GATE, WM8958_AIF2DAC_NG_THR_SHIFT,
7, 1, ng_tlv),
};
static const struct snd_kcontrol_new wm1811_snd_controls[] = {
SOC_SINGLE_TLV("MIXINL IN1LP Boost Volume", WM8994_INPUT_MIXER_1, 7, 1, 0,
mixin_boost_tlv),
SOC_SINGLE_TLV("MIXINL IN1RP Boost Volume", WM8994_INPUT_MIXER_1, 8, 1, 0,
mixin_boost_tlv),
};
/* We run all mode setting through a function to enforce audio mode */
static void wm1811_jackdet_set_mode(struct snd_soc_codec *codec, u16 mode)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
if (!wm8994->jackdet || !wm8994->micdet[0].jack)
return;
if (wm8994->active_refcount)
mode = WM1811_JACKDET_MODE_AUDIO;
if (mode == wm8994->jackdet_mode)
return;
wm8994->jackdet_mode = mode;
/* Always use audio mode to detect while the system is active */
if (mode != WM1811_JACKDET_MODE_NONE)
mode = WM1811_JACKDET_MODE_AUDIO;
snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
WM1811_JACKDET_MODE_MASK, mode);
}
static void active_reference(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
mutex_lock(&wm8994->accdet_lock);
wm8994->active_refcount++;
dev_dbg(codec->dev, "Active refcount incremented, now %d\n",
wm8994->active_refcount);
/* If we're using jack detection go into audio mode */
wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_AUDIO);
mutex_unlock(&wm8994->accdet_lock);
}
static void active_dereference(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
u16 mode;
mutex_lock(&wm8994->accdet_lock);
wm8994->active_refcount--;
dev_dbg(codec->dev, "Active refcount decremented, now %d\n",
wm8994->active_refcount);
if (wm8994->active_refcount == 0) {
/* Go into appropriate detection only mode */
if (wm8994->jack_mic || wm8994->mic_detecting)
mode = WM1811_JACKDET_MODE_MIC;
else
mode = WM1811_JACKDET_MODE_JACK;
wm1811_jackdet_set_mode(codec, mode);
}
mutex_unlock(&wm8994->accdet_lock);
}
static int clk_sys_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
return configure_clock(codec);
case SND_SOC_DAPM_POST_PMU:
/*
* JACKDET won't run until we start the clock and it
* only reports deltas, make sure we notify the state
* up the stack on startup. Use a *very* generous
* timeout for paranoia, there's no urgency and we
* don't want false reports.
*/
if (wm8994->jackdet && !wm8994->clk_has_run) {
queue_delayed_work(system_power_efficient_wq,
&wm8994->jackdet_bootstrap,
msecs_to_jiffies(1000));
wm8994->clk_has_run = true;
}
break;
case SND_SOC_DAPM_POST_PMD:
configure_clock(codec);
break;
}
return 0;
}
static void vmid_reference(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
pm_runtime_get_sync(codec->dev);
wm8994->vmid_refcount++;
dev_dbg(codec->dev, "Referencing VMID, refcount is now %d\n",
wm8994->vmid_refcount);
if (wm8994->vmid_refcount == 1) {
snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
WM8994_LINEOUT1_DISCH |
WM8994_LINEOUT2_DISCH, 0);
wm_hubs_vmid_ena(codec);
switch (wm8994->vmid_mode) {
default:
WARN_ON(NULL == "Invalid VMID mode");
case WM8994_VMID_NORMAL:
/* Startup bias, VMID ramp & buffer */
snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
WM8994_BIAS_SRC |
WM8994_VMID_DISCH |
WM8994_STARTUP_BIAS_ENA |
WM8994_VMID_BUF_ENA |
WM8994_VMID_RAMP_MASK,
WM8994_BIAS_SRC |
WM8994_STARTUP_BIAS_ENA |
WM8994_VMID_BUF_ENA |
(0x2 << WM8994_VMID_RAMP_SHIFT));
/* Main bias enable, VMID=2x40k */
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
WM8994_BIAS_ENA |
WM8994_VMID_SEL_MASK,
WM8994_BIAS_ENA | 0x2);
msleep(300);
snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
WM8994_VMID_RAMP_MASK |
WM8994_BIAS_SRC,
0);
break;
case WM8994_VMID_FORCE:
/* Startup bias, slow VMID ramp & buffer */
snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
WM8994_BIAS_SRC |
WM8994_VMID_DISCH |
WM8994_STARTUP_BIAS_ENA |
WM8994_VMID_BUF_ENA |
WM8994_VMID_RAMP_MASK,
WM8994_BIAS_SRC |
WM8994_STARTUP_BIAS_ENA |
WM8994_VMID_BUF_ENA |
(0x2 << WM8994_VMID_RAMP_SHIFT));
/* Main bias enable, VMID=2x40k */
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
WM8994_BIAS_ENA |
WM8994_VMID_SEL_MASK,
WM8994_BIAS_ENA | 0x2);
msleep(400);
snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
WM8994_VMID_RAMP_MASK |
WM8994_BIAS_SRC,
0);
break;
}
}
}
static void vmid_dereference(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
wm8994->vmid_refcount--;
dev_dbg(codec->dev, "Dereferencing VMID, refcount is now %d\n",
wm8994->vmid_refcount);
if (wm8994->vmid_refcount == 0) {
if (wm8994->hubs.lineout1_se)
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_3,
WM8994_LINEOUT1N_ENA |
WM8994_LINEOUT1P_ENA,
WM8994_LINEOUT1N_ENA |
WM8994_LINEOUT1P_ENA);
if (wm8994->hubs.lineout2_se)
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_3,
WM8994_LINEOUT2N_ENA |
WM8994_LINEOUT2P_ENA,
WM8994_LINEOUT2N_ENA |
WM8994_LINEOUT2P_ENA);
/* Start discharging VMID */
snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
WM8994_BIAS_SRC |
WM8994_VMID_DISCH,
WM8994_BIAS_SRC |
WM8994_VMID_DISCH);
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
WM8994_VMID_SEL_MASK, 0);
msleep(400);
/* Active discharge */
snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
WM8994_LINEOUT1_DISCH |
WM8994_LINEOUT2_DISCH,
WM8994_LINEOUT1_DISCH |
WM8994_LINEOUT2_DISCH);
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_3,
WM8994_LINEOUT1N_ENA |
WM8994_LINEOUT1P_ENA |
WM8994_LINEOUT2N_ENA |
WM8994_LINEOUT2P_ENA, 0);
/* Switch off startup biases */
snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
WM8994_BIAS_SRC |
WM8994_STARTUP_BIAS_ENA |
WM8994_VMID_BUF_ENA |
WM8994_VMID_RAMP_MASK, 0);
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1,
WM8994_VMID_SEL_MASK, 0);
}
pm_runtime_put(codec->dev);
}
static int vmid_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
vmid_reference(codec);
break;
case SND_SOC_DAPM_POST_PMD:
vmid_dereference(codec);
break;
}
return 0;
}
static bool wm8994_check_class_w_digital(struct snd_soc_codec *codec)
{
int source = 0; /* GCC flow analysis can't track enable */
int reg, reg_r;
/* We also need the same AIF source for L/R and only one path */
reg = snd_soc_read(codec, WM8994_DAC1_LEFT_MIXER_ROUTING);
switch (reg) {
case WM8994_AIF2DACL_TO_DAC1L:
dev_vdbg(codec->dev, "Class W source AIF2DAC\n");
source = 2 << WM8994_CP_DYN_SRC_SEL_SHIFT;
break;
case WM8994_AIF1DAC2L_TO_DAC1L:
dev_vdbg(codec->dev, "Class W source AIF1DAC2\n");
source = 1 << WM8994_CP_DYN_SRC_SEL_SHIFT;
break;
case WM8994_AIF1DAC1L_TO_DAC1L:
dev_vdbg(codec->dev, "Class W source AIF1DAC1\n");
source = 0 << WM8994_CP_DYN_SRC_SEL_SHIFT;
break;
default:
dev_vdbg(codec->dev, "DAC mixer setting: %x\n", reg);
return false;
}
reg_r = snd_soc_read(codec, WM8994_DAC1_RIGHT_MIXER_ROUTING);
if (reg_r != reg) {
dev_vdbg(codec->dev, "Left and right DAC mixers different\n");
return false;
}
/* Set the source up */
snd_soc_update_bits(codec, WM8994_CLASS_W_1,
WM8994_CP_DYN_SRC_SEL_MASK, source);
return true;
}
static int aif1clk_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA;
int i;
int dac;
int adc;
int val;
switch (control->type) {
case WM8994:
case WM8958:
mask |= WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA;
break;
default:
break;
}
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
/* Don't enable timeslot 2 if not in use */
if (wm8994->channels[0] <= 2)
mask &= ~(WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA);
val = snd_soc_read(codec, WM8994_AIF1_CONTROL_1);
if ((val & WM8994_AIF1ADCL_SRC) &&
(val & WM8994_AIF1ADCR_SRC))
adc = WM8994_AIF1ADC1R_ENA | WM8994_AIF1ADC2R_ENA;
else if (!(val & WM8994_AIF1ADCL_SRC) &&
!(val & WM8994_AIF1ADCR_SRC))
adc = WM8994_AIF1ADC1L_ENA | WM8994_AIF1ADC2L_ENA;
else
adc = WM8994_AIF1ADC1R_ENA | WM8994_AIF1ADC2R_ENA |
WM8994_AIF1ADC1L_ENA | WM8994_AIF1ADC2L_ENA;
val = snd_soc_read(codec, WM8994_AIF1_CONTROL_2);
if ((val & WM8994_AIF1DACL_SRC) &&
(val & WM8994_AIF1DACR_SRC))
dac = WM8994_AIF1DAC1R_ENA | WM8994_AIF1DAC2R_ENA;
else if (!(val & WM8994_AIF1DACL_SRC) &&
!(val & WM8994_AIF1DACR_SRC))
dac = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC2L_ENA;
else
dac = WM8994_AIF1DAC1R_ENA | WM8994_AIF1DAC2R_ENA |
WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC2L_ENA;
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4,
mask, adc);
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
mask, dac);
snd_soc_update_bits(codec, WM8994_CLOCKING_1,
WM8994_AIF1DSPCLK_ENA |
WM8994_SYSDSPCLK_ENA,
WM8994_AIF1DSPCLK_ENA |
WM8994_SYSDSPCLK_ENA);
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, mask,
WM8994_AIF1ADC1R_ENA |
WM8994_AIF1ADC1L_ENA |
WM8994_AIF1ADC2R_ENA |
WM8994_AIF1ADC2L_ENA);
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, mask,
WM8994_AIF1DAC1R_ENA |
WM8994_AIF1DAC1L_ENA |
WM8994_AIF1DAC2R_ENA |
WM8994_AIF1DAC2L_ENA);
break;
case SND_SOC_DAPM_POST_PMU:
for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++)
snd_soc_write(codec, wm8994_vu_bits[i].reg,
snd_soc_read(codec,
wm8994_vu_bits[i].reg));
break;
case SND_SOC_DAPM_PRE_PMD:
case SND_SOC_DAPM_POST_PMD:
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
mask, 0);
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4,
mask, 0);
val = snd_soc_read(codec, WM8994_CLOCKING_1);
if (val & WM8994_AIF2DSPCLK_ENA)
val = WM8994_SYSDSPCLK_ENA;
else
val = 0;
snd_soc_update_bits(codec, WM8994_CLOCKING_1,
WM8994_SYSDSPCLK_ENA |
WM8994_AIF1DSPCLK_ENA, val);
break;
}
return 0;
}
static int aif2clk_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
int i;
int dac;
int adc;
int val;
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
val = snd_soc_read(codec, WM8994_AIF2_CONTROL_1);
if ((val & WM8994_AIF2ADCL_SRC) &&
(val & WM8994_AIF2ADCR_SRC))
adc = WM8994_AIF2ADCR_ENA;
else if (!(val & WM8994_AIF2ADCL_SRC) &&
!(val & WM8994_AIF2ADCR_SRC))
adc = WM8994_AIF2ADCL_ENA;
else
adc = WM8994_AIF2ADCL_ENA | WM8994_AIF2ADCR_ENA;
val = snd_soc_read(codec, WM8994_AIF2_CONTROL_2);
if ((val & WM8994_AIF2DACL_SRC) &&
(val & WM8994_AIF2DACR_SRC))
dac = WM8994_AIF2DACR_ENA;
else if (!(val & WM8994_AIF2DACL_SRC) &&
!(val & WM8994_AIF2DACR_SRC))
dac = WM8994_AIF2DACL_ENA;
else
dac = WM8994_AIF2DACL_ENA | WM8994_AIF2DACR_ENA;
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4,
WM8994_AIF2ADCL_ENA |
WM8994_AIF2ADCR_ENA, adc);
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
WM8994_AIF2DACL_ENA |
WM8994_AIF2DACR_ENA, dac);
snd_soc_update_bits(codec, WM8994_CLOCKING_1,
WM8994_AIF2DSPCLK_ENA |
WM8994_SYSDSPCLK_ENA,
WM8994_AIF2DSPCLK_ENA |
WM8994_SYSDSPCLK_ENA);
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4,
WM8994_AIF2ADCL_ENA |
WM8994_AIF2ADCR_ENA,
WM8994_AIF2ADCL_ENA |
WM8994_AIF2ADCR_ENA);
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
WM8994_AIF2DACL_ENA |
WM8994_AIF2DACR_ENA,
WM8994_AIF2DACL_ENA |
WM8994_AIF2DACR_ENA);
break;
case SND_SOC_DAPM_POST_PMU:
for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++)
snd_soc_write(codec, wm8994_vu_bits[i].reg,
snd_soc_read(codec,
wm8994_vu_bits[i].reg));
break;
case SND_SOC_DAPM_PRE_PMD:
case SND_SOC_DAPM_POST_PMD:
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
WM8994_AIF2DACL_ENA |
WM8994_AIF2DACR_ENA, 0);
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4,
WM8994_AIF2ADCL_ENA |
WM8994_AIF2ADCR_ENA, 0);
val = snd_soc_read(codec, WM8994_CLOCKING_1);
if (val & WM8994_AIF1DSPCLK_ENA)
val = WM8994_SYSDSPCLK_ENA;
else
val = 0;
snd_soc_update_bits(codec, WM8994_CLOCKING_1,
WM8994_SYSDSPCLK_ENA |
WM8994_AIF2DSPCLK_ENA, val);
break;
}
return 0;
}
static int aif1clk_late_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
wm8994->aif1clk_enable = 1;
break;
case SND_SOC_DAPM_POST_PMD:
wm8994->aif1clk_disable = 1;
break;
}
return 0;
}
static int aif2clk_late_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
wm8994->aif2clk_enable = 1;
break;
case SND_SOC_DAPM_POST_PMD:
wm8994->aif2clk_disable = 1;
break;
}
return 0;
}
static int late_enable_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
if (wm8994->aif1clk_enable) {
aif1clk_ev(w, kcontrol, SND_SOC_DAPM_PRE_PMU);
snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
WM8994_AIF1CLK_ENA_MASK,
WM8994_AIF1CLK_ENA);
aif1clk_ev(w, kcontrol, SND_SOC_DAPM_POST_PMU);
wm8994->aif1clk_enable = 0;
}
if (wm8994->aif2clk_enable) {
aif2clk_ev(w, kcontrol, SND_SOC_DAPM_PRE_PMU);
snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
WM8994_AIF2CLK_ENA_MASK,
WM8994_AIF2CLK_ENA);
aif2clk_ev(w, kcontrol, SND_SOC_DAPM_POST_PMU);
wm8994->aif2clk_enable = 0;
}
break;
}
/* We may also have postponed startup of DSP, handle that. */
wm8958_aif_ev(w, kcontrol, event);
return 0;
}
static int late_disable_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_POST_PMD:
if (wm8994->aif1clk_disable) {
aif1clk_ev(w, kcontrol, SND_SOC_DAPM_PRE_PMD);
snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
WM8994_AIF1CLK_ENA_MASK, 0);
aif1clk_ev(w, kcontrol, SND_SOC_DAPM_POST_PMD);
wm8994->aif1clk_disable = 0;
}
if (wm8994->aif2clk_disable) {
aif2clk_ev(w, kcontrol, SND_SOC_DAPM_PRE_PMD);
snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
WM8994_AIF2CLK_ENA_MASK, 0);
aif2clk_ev(w, kcontrol, SND_SOC_DAPM_POST_PMD);
wm8994->aif2clk_disable = 0;
}
break;
}
return 0;
}
static int adc_mux_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
late_enable_ev(w, kcontrol, event);
return 0;
}
static int micbias_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
late_enable_ev(w, kcontrol, event);
return 0;
}
static int dac_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
unsigned int mask = 1 << w->shift;
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
mask, mask);
return 0;
}
static const char *adc_mux_text[] = {
"ADC",
"DMIC",
};
static const struct soc_enum adc_enum =
SOC_ENUM_SINGLE(0, 0, 2, adc_mux_text);
static const struct snd_kcontrol_new adcl_mux =
SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum);
static const struct snd_kcontrol_new adcr_mux =
SOC_DAPM_ENUM_VIRT("ADCR Mux", adc_enum);
static const struct snd_kcontrol_new left_speaker_mixer[] = {
SOC_DAPM_SINGLE("DAC2 Switch", WM8994_SPEAKER_MIXER, 9, 1, 0),
SOC_DAPM_SINGLE("Input Switch", WM8994_SPEAKER_MIXER, 7, 1, 0),
SOC_DAPM_SINGLE("IN1LP Switch", WM8994_SPEAKER_MIXER, 5, 1, 0),
SOC_DAPM_SINGLE("Output Switch", WM8994_SPEAKER_MIXER, 3, 1, 0),
SOC_DAPM_SINGLE("DAC1 Switch", WM8994_SPEAKER_MIXER, 1, 1, 0),
};
static const struct snd_kcontrol_new right_speaker_mixer[] = {
SOC_DAPM_SINGLE("DAC2 Switch", WM8994_SPEAKER_MIXER, 8, 1, 0),
SOC_DAPM_SINGLE("Input Switch", WM8994_SPEAKER_MIXER, 6, 1, 0),
SOC_DAPM_SINGLE("IN1RP Switch", WM8994_SPEAKER_MIXER, 4, 1, 0),
SOC_DAPM_SINGLE("Output Switch", WM8994_SPEAKER_MIXER, 2, 1, 0),
SOC_DAPM_SINGLE("DAC1 Switch", WM8994_SPEAKER_MIXER, 0, 1, 0),
};
/* Debugging; dump chip status after DAPM transitions */
static int post_ev(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
dev_dbg(codec->dev, "SRC status: %x\n",
snd_soc_read(codec,
WM8994_RATE_STATUS));
return 0;
}
static const struct snd_kcontrol_new aif1adc1l_mix[] = {
SOC_DAPM_SINGLE("ADC/DMIC Switch", WM8994_AIF1_ADC1_LEFT_MIXER_ROUTING,
1, 1, 0),
SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC1_LEFT_MIXER_ROUTING,
0, 1, 0),
};
static const struct snd_kcontrol_new aif1adc1r_mix[] = {
SOC_DAPM_SINGLE("ADC/DMIC Switch", WM8994_AIF1_ADC1_RIGHT_MIXER_ROUTING,
1, 1, 0),
SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC1_RIGHT_MIXER_ROUTING,
0, 1, 0),
};
static const struct snd_kcontrol_new aif1adc2l_mix[] = {
SOC_DAPM_SINGLE("DMIC Switch", WM8994_AIF1_ADC2_LEFT_MIXER_ROUTING,
1, 1, 0),
SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC2_LEFT_MIXER_ROUTING,
0, 1, 0),
};
static const struct snd_kcontrol_new aif1adc2r_mix[] = {
SOC_DAPM_SINGLE("DMIC Switch", WM8994_AIF1_ADC2_RIGHT_MIXER_ROUTING,
1, 1, 0),
SOC_DAPM_SINGLE("AIF2 Switch", WM8994_AIF1_ADC2_RIGHT_MIXER_ROUTING,
0, 1, 0),
};
static const struct snd_kcontrol_new aif2dac2l_mix[] = {
SOC_DAPM_SINGLE("Right Sidetone Switch", WM8994_DAC2_LEFT_MIXER_ROUTING,
5, 1, 0),
SOC_DAPM_SINGLE("Left Sidetone Switch", WM8994_DAC2_LEFT_MIXER_ROUTING,
4, 1, 0),
SOC_DAPM_SINGLE("AIF2 Switch", WM8994_DAC2_LEFT_MIXER_ROUTING,
2, 1, 0),
SOC_DAPM_SINGLE("AIF1.2 Switch", WM8994_DAC2_LEFT_MIXER_ROUTING,
1, 1, 0),
SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_LEFT_MIXER_ROUTING,
0, 1, 0),
};
static const struct snd_kcontrol_new aif2dac2r_mix[] = {
SOC_DAPM_SINGLE("Right Sidetone Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING,
5, 1, 0),
SOC_DAPM_SINGLE("Left Sidetone Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING,
4, 1, 0),
SOC_DAPM_SINGLE("AIF2 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING,
2, 1, 0),
SOC_DAPM_SINGLE("AIF1.2 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING,
1, 1, 0),
SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING,
0, 1, 0),
};
#define WM8994_CLASS_W_SWITCH(xname, reg, shift, max, invert) \
SOC_SINGLE_EXT(xname, reg, shift, max, invert, \
snd_soc_dapm_get_volsw, wm8994_put_class_w)
static int wm8994_put_class_w(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol);
int ret;
ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
wm_hubs_update_class_w(codec);
return ret;
}
static const struct snd_kcontrol_new dac1l_mix[] = {
WM8994_CLASS_W_SWITCH("Right Sidetone Switch", WM8994_DAC1_LEFT_MIXER_ROUTING,
5, 1, 0),
WM8994_CLASS_W_SWITCH("Left Sidetone Switch", WM8994_DAC1_LEFT_MIXER_ROUTING,
4, 1, 0),
WM8994_CLASS_W_SWITCH("AIF2 Switch", WM8994_DAC1_LEFT_MIXER_ROUTING,
2, 1, 0),
WM8994_CLASS_W_SWITCH("AIF1.2 Switch", WM8994_DAC1_LEFT_MIXER_ROUTING,
1, 1, 0),
WM8994_CLASS_W_SWITCH("AIF1.1 Switch", WM8994_DAC1_LEFT_MIXER_ROUTING,
0, 1, 0),
};
static const struct snd_kcontrol_new dac1r_mix[] = {
WM8994_CLASS_W_SWITCH("Right Sidetone Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING,
5, 1, 0),
WM8994_CLASS_W_SWITCH("Left Sidetone Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING,
4, 1, 0),
WM8994_CLASS_W_SWITCH("AIF2 Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING,
2, 1, 0),
WM8994_CLASS_W_SWITCH("AIF1.2 Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING,
1, 1, 0),
WM8994_CLASS_W_SWITCH("AIF1.1 Switch", WM8994_DAC1_RIGHT_MIXER_ROUTING,
0, 1, 0),
};
static const char *sidetone_text[] = {
"ADC/DMIC1", "DMIC2",
};
static const struct soc_enum sidetone1_enum =
SOC_ENUM_SINGLE(WM8994_SIDETONE, 0, 2, sidetone_text);
static const struct snd_kcontrol_new sidetone1_mux =
SOC_DAPM_ENUM("Left Sidetone Mux", sidetone1_enum);
static const struct soc_enum sidetone2_enum =
SOC_ENUM_SINGLE(WM8994_SIDETONE, 1, 2, sidetone_text);
static const struct snd_kcontrol_new sidetone2_mux =
SOC_DAPM_ENUM("Right Sidetone Mux", sidetone2_enum);
static const char *aif1dac_text[] = {
"AIF1DACDAT", "AIF3DACDAT",
};
static const char *loopback_text[] = {
"None", "ADCDAT",
};
static const struct soc_enum aif1_loopback_enum =
SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, WM8994_AIF1_LOOPBACK_SHIFT, 2,
loopback_text);
static const struct snd_kcontrol_new aif1_loopback =
SOC_DAPM_ENUM("AIF1 Loopback", aif1_loopback_enum);
static const struct soc_enum aif2_loopback_enum =
SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, WM8994_AIF2_LOOPBACK_SHIFT, 2,
loopback_text);
static const struct snd_kcontrol_new aif2_loopback =
SOC_DAPM_ENUM("AIF2 Loopback", aif2_loopback_enum);
static const struct soc_enum aif1dac_enum =
SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text);
static const struct snd_kcontrol_new aif1dac_mux =
SOC_DAPM_ENUM("AIF1DAC Mux", aif1dac_enum);
static const char *aif2dac_text[] = {
"AIF2DACDAT", "AIF3DACDAT",
};
static const struct soc_enum aif2dac_enum =
SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 1, 2, aif2dac_text);
static const struct snd_kcontrol_new aif2dac_mux =
SOC_DAPM_ENUM("AIF2DAC Mux", aif2dac_enum);
static const char *aif2adc_text[] = {
"AIF2ADCDAT", "AIF3DACDAT",
};
static const struct soc_enum aif2adc_enum =
SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 2, 2, aif2adc_text);
static const struct snd_kcontrol_new aif2adc_mux =
SOC_DAPM_ENUM("AIF2ADC Mux", aif2adc_enum);
static const char *aif3adc_text[] = {
"AIF1ADCDAT", "AIF2ADCDAT", "AIF2DACDAT", "Mono PCM",
};
static const struct soc_enum wm8994_aif3adc_enum =
SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 3, aif3adc_text);
static const struct snd_kcontrol_new wm8994_aif3adc_mux =
SOC_DAPM_ENUM("AIF3ADC Mux", wm8994_aif3adc_enum);
static const struct soc_enum wm8958_aif3adc_enum =
SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 4, aif3adc_text);
static const struct snd_kcontrol_new wm8958_aif3adc_mux =
SOC_DAPM_ENUM("AIF3ADC Mux", wm8958_aif3adc_enum);
static const char *mono_pcm_out_text[] = {
"None", "AIF2ADCL", "AIF2ADCR",
};
static const struct soc_enum mono_pcm_out_enum =
SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 9, 3, mono_pcm_out_text);
static const struct snd_kcontrol_new mono_pcm_out_mux =
SOC_DAPM_ENUM("Mono PCM Out Mux", mono_pcm_out_enum);
static const char *aif2dac_src_text[] = {
"AIF2", "AIF3",
};
/* Note that these two control shouldn't be simultaneously switched to AIF3 */
static const struct soc_enum aif2dacl_src_enum =
SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 7, 2, aif2dac_src_text);
static const struct snd_kcontrol_new aif2dacl_src_mux =
SOC_DAPM_ENUM("AIF2DACL Mux", aif2dacl_src_enum);
static const struct soc_enum aif2dacr_src_enum =
SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 8, 2, aif2dac_src_text);
static const struct snd_kcontrol_new aif2dacr_src_mux =
SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum);
static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = {
SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_late_ev,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_late_ev,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
late_enable_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_PGA_E("Late DAC1R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
late_enable_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
late_enable_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
late_enable_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_PGA_E("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0,
late_enable_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_MIXER_E("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0,
left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer),
late_enable_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_MIXER_E("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0,
right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer),
late_enable_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux,
late_enable_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux,
late_enable_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev)
};
static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = {
SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, aif1clk_ev,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, aif2clk_ev,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0,
left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)),
SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0,
right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpl_mux),
SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &wm_hubs_hpr_mux),
};
static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = {
SND_SOC_DAPM_DAC_E("DAC2L", NULL, SND_SOC_NOPM, 3, 0,
dac_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_DAC_E("DAC2R", NULL, SND_SOC_NOPM, 2, 0,
dac_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_DAC_E("DAC1L", NULL, SND_SOC_NOPM, 1, 0,
dac_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_DAC_E("DAC1R", NULL, SND_SOC_NOPM, 0, 0,
dac_ev, SND_SOC_DAPM_PRE_PMU),
};
static const struct snd_soc_dapm_widget wm8994_dac_widgets[] = {
SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0),
SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0),
SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0),
SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0),
};
static const struct snd_soc_dapm_widget wm8994_adc_revd_widgets[] = {
SND_SOC_DAPM_VIRT_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux,
adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_VIRT_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux,
adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
};
static const struct snd_soc_dapm_widget wm8994_adc_widgets[] = {
SND_SOC_DAPM_VIRT_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
SND_SOC_DAPM_VIRT_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
};
static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("DMIC1DAT"),
SND_SOC_DAPM_INPUT("DMIC2DAT"),
SND_SOC_DAPM_INPUT("Clock"),
SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev,
SND_SOC_DAPM_PRE_PMU),
SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, vmid_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_SUPPLY("DSP1CLK", SND_SOC_NOPM, 3, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("DSP2CLK", SND_SOC_NOPM, 2, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("DSPINTCLK", SND_SOC_NOPM, 1, 0, NULL, 0),
SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL,
0, SND_SOC_NOPM, 9, 0),
SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL,
0, SND_SOC_NOPM, 8, 0),
SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0,
SND_SOC_NOPM, 9, 0, wm8958_aif_ev,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0,
SND_SOC_NOPM, 8, 0, wm8958_aif_ev,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL,
0, SND_SOC_NOPM, 11, 0),
SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL,
0, SND_SOC_NOPM, 10, 0),
SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0,
SND_SOC_NOPM, 11, 0, wm8958_aif_ev,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_AIF_IN_E("AIF1DAC2R", NULL, 0,
SND_SOC_NOPM, 10, 0, wm8958_aif_ev,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0,
aif1adc1l_mix, ARRAY_SIZE(aif1adc1l_mix)),
SND_SOC_DAPM_MIXER("AIF1ADC1R Mixer", SND_SOC_NOPM, 0, 0,
aif1adc1r_mix, ARRAY_SIZE(aif1adc1r_mix)),
SND_SOC_DAPM_MIXER("AIF1ADC2L Mixer", SND_SOC_NOPM, 0, 0,
aif1adc2l_mix, ARRAY_SIZE(aif1adc2l_mix)),
SND_SOC_DAPM_MIXER("AIF1ADC2R Mixer", SND_SOC_NOPM, 0, 0,
aif1adc2r_mix, ARRAY_SIZE(aif1adc2r_mix)),
SND_SOC_DAPM_MIXER("AIF2DAC2L Mixer", SND_SOC_NOPM, 0, 0,
aif2dac2l_mix, ARRAY_SIZE(aif2dac2l_mix)),
SND_SOC_DAPM_MIXER("AIF2DAC2R Mixer", SND_SOC_NOPM, 0, 0,
aif2dac2r_mix, ARRAY_SIZE(aif2dac2r_mix)),
SND_SOC_DAPM_MUX("Left Sidetone", SND_SOC_NOPM, 0, 0, &sidetone1_mux),
SND_SOC_DAPM_MUX("Right Sidetone", SND_SOC_NOPM, 0, 0, &sidetone2_mux),
SND_SOC_DAPM_MIXER("DAC1L Mixer", SND_SOC_NOPM, 0, 0,
dac1l_mix, ARRAY_SIZE(dac1l_mix)),
SND_SOC_DAPM_MIXER("DAC1R Mixer", SND_SOC_NOPM, 0, 0,
dac1r_mix, ARRAY_SIZE(dac1r_mix)),
SND_SOC_DAPM_AIF_OUT("AIF2ADCL", NULL, 0,
SND_SOC_NOPM, 13, 0),
SND_SOC_DAPM_AIF_OUT("AIF2ADCR", NULL, 0,
SND_SOC_NOPM, 12, 0),
SND_SOC_DAPM_AIF_IN_E("AIF2DACL", NULL, 0,
SND_SOC_NOPM, 13, 0, wm8958_aif_ev,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0,
SND_SOC_NOPM, 12, 0, wm8958_aif_ev,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_AIF_IN("AIF1DACDAT", NULL, 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("AIF2DACDAT", NULL, 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", NULL, 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", NULL, 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux),
SND_SOC_DAPM_MUX("AIF2DAC Mux", SND_SOC_NOPM, 0, 0, &aif2dac_mux),
SND_SOC_DAPM_MUX("AIF2ADC Mux", SND_SOC_NOPM, 0, 0, &aif2adc_mux),
SND_SOC_DAPM_AIF_IN("AIF3DACDAT", NULL, 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("AIF3ADCDAT", NULL, 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_SUPPLY("TOCLK", WM8994_CLOCKING_1, 4, 0, NULL, 0),
SND_SOC_DAPM_ADC("DMIC2L", NULL, WM8994_POWER_MANAGEMENT_4, 5, 0),
SND_SOC_DAPM_ADC("DMIC2R", NULL, WM8994_POWER_MANAGEMENT_4, 4, 0),
SND_SOC_DAPM_ADC("DMIC1L", NULL, WM8994_POWER_MANAGEMENT_4, 3, 0),
SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0),
/* Power is done with the muxes since the ADC power also controls the
* downsampling chain, the chip will automatically manage the analogue
* specific portions.
*/
SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0),
SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MUX("AIF1 Loopback", SND_SOC_NOPM, 0, 0, &aif1_loopback),
SND_SOC_DAPM_MUX("AIF2 Loopback", SND_SOC_NOPM, 0, 0, &aif2_loopback),
SND_SOC_DAPM_POST("Debug log", post_ev),
};
static const struct snd_soc_dapm_widget wm8994_specific_dapm_widgets[] = {
SND_SOC_DAPM_MUX("AIF3ADC Mux", SND_SOC_NOPM, 0, 0, &wm8994_aif3adc_mux),
};
static const struct snd_soc_dapm_widget wm8958_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("AIF3", WM8994_POWER_MANAGEMENT_6, 5, 1, NULL, 0),
SND_SOC_DAPM_MUX("Mono PCM Out Mux", SND_SOC_NOPM, 0, 0, &mono_pcm_out_mux),
SND_SOC_DAPM_MUX("AIF2DACL Mux", SND_SOC_NOPM, 0, 0, &aif2dacl_src_mux),
SND_SOC_DAPM_MUX("AIF2DACR Mux", SND_SOC_NOPM, 0, 0, &aif2dacr_src_mux),
SND_SOC_DAPM_MUX("AIF3ADC Mux", SND_SOC_NOPM, 0, 0, &wm8958_aif3adc_mux),
};
static const struct snd_soc_dapm_route intercon[] = {
{ "CLK_SYS", NULL, "AIF1CLK", check_clk_sys },
{ "CLK_SYS", NULL, "AIF2CLK", check_clk_sys },
{ "DSP1CLK", NULL, "CLK_SYS" },
{ "DSP2CLK", NULL, "CLK_SYS" },
{ "DSPINTCLK", NULL, "CLK_SYS" },
{ "AIF1ADC1L", NULL, "AIF1CLK" },
{ "AIF1ADC1L", NULL, "DSP1CLK" },
{ "AIF1ADC1R", NULL, "AIF1CLK" },
{ "AIF1ADC1R", NULL, "DSP1CLK" },
{ "AIF1ADC1R", NULL, "DSPINTCLK" },
{ "AIF1DAC1L", NULL, "AIF1CLK" },
{ "AIF1DAC1L", NULL, "DSP1CLK" },
{ "AIF1DAC1R", NULL, "AIF1CLK" },
{ "AIF1DAC1R", NULL, "DSP1CLK" },
{ "AIF1DAC1R", NULL, "DSPINTCLK" },
{ "AIF1ADC2L", NULL, "AIF1CLK" },
{ "AIF1ADC2L", NULL, "DSP1CLK" },
{ "AIF1ADC2R", NULL, "AIF1CLK" },
{ "AIF1ADC2R", NULL, "DSP1CLK" },
{ "AIF1ADC2R", NULL, "DSPINTCLK" },
{ "AIF1DAC2L", NULL, "AIF1CLK" },
{ "AIF1DAC2L", NULL, "DSP1CLK" },
{ "AIF1DAC2R", NULL, "AIF1CLK" },
{ "AIF1DAC2R", NULL, "DSP1CLK" },
{ "AIF1DAC2R", NULL, "DSPINTCLK" },
{ "AIF2ADCL", NULL, "AIF2CLK" },
{ "AIF2ADCL", NULL, "DSP2CLK" },
{ "AIF2ADCR", NULL, "AIF2CLK" },
{ "AIF2ADCR", NULL, "DSP2CLK" },
{ "AIF2ADCR", NULL, "DSPINTCLK" },
{ "AIF2DACL", NULL, "AIF2CLK" },
{ "AIF2DACL", NULL, "DSP2CLK" },
{ "AIF2DACR", NULL, "AIF2CLK" },
{ "AIF2DACR", NULL, "DSP2CLK" },
{ "AIF2DACR", NULL, "DSPINTCLK" },
{ "DMIC1L", NULL, "DMIC1DAT" },
{ "DMIC1L", NULL, "CLK_SYS" },
{ "DMIC1R", NULL, "DMIC1DAT" },
{ "DMIC1R", NULL, "CLK_SYS" },
{ "DMIC2L", NULL, "DMIC2DAT" },
{ "DMIC2L", NULL, "CLK_SYS" },
{ "DMIC2R", NULL, "DMIC2DAT" },
{ "DMIC2R", NULL, "CLK_SYS" },
{ "ADCL", NULL, "AIF1CLK" },
{ "ADCL", NULL, "DSP1CLK" },
{ "ADCL", NULL, "DSPINTCLK" },
{ "ADCR", NULL, "AIF1CLK" },
{ "ADCR", NULL, "DSP1CLK" },
{ "ADCR", NULL, "DSPINTCLK" },
{ "ADCL Mux", "ADC", "ADCL" },
{ "ADCL Mux", "DMIC", "DMIC1L" },
{ "ADCR Mux", "ADC", "ADCR" },
{ "ADCR Mux", "DMIC", "DMIC1R" },
{ "DAC1L", NULL, "AIF1CLK" },
{ "DAC1L", NULL, "DSP1CLK" },
{ "DAC1L", NULL, "DSPINTCLK" },
{ "DAC1R", NULL, "AIF1CLK" },
{ "DAC1R", NULL, "DSP1CLK" },
{ "DAC1R", NULL, "DSPINTCLK" },
{ "DAC2L", NULL, "AIF2CLK" },
{ "DAC2L", NULL, "DSP2CLK" },
{ "DAC2L", NULL, "DSPINTCLK" },
{ "DAC2R", NULL, "AIF2DACR" },
{ "DAC2R", NULL, "AIF2CLK" },
{ "DAC2R", NULL, "DSP2CLK" },
{ "DAC2R", NULL, "DSPINTCLK" },
{ "TOCLK", NULL, "CLK_SYS" },
{ "AIF1DACDAT", NULL, "AIF1 Playback" },
{ "AIF2DACDAT", NULL, "AIF2 Playback" },
{ "AIF3DACDAT", NULL, "AIF3 Playback" },
{ "AIF1 Capture", NULL, "AIF1ADCDAT" },
{ "AIF2 Capture", NULL, "AIF2ADCDAT" },
{ "AIF3 Capture", NULL, "AIF3ADCDAT" },
/* AIF1 outputs */
{ "AIF1ADC1L", NULL, "AIF1ADC1L Mixer" },
{ "AIF1ADC1L Mixer", "ADC/DMIC Switch", "ADCL Mux" },
{ "AIF1ADC1L Mixer", "AIF2 Switch", "AIF2DACL" },
{ "AIF1ADC1R", NULL, "AIF1ADC1R Mixer" },
{ "AIF1ADC1R Mixer", "ADC/DMIC Switch", "ADCR Mux" },
{ "AIF1ADC1R Mixer", "AIF2 Switch", "AIF2DACR" },
{ "AIF1ADC2L", NULL, "AIF1ADC2L Mixer" },
{ "AIF1ADC2L Mixer", "DMIC Switch", "DMIC2L" },
{ "AIF1ADC2L Mixer", "AIF2 Switch", "AIF2DACL" },
{ "AIF1ADC2R", NULL, "AIF1ADC2R Mixer" },
{ "AIF1ADC2R Mixer", "DMIC Switch", "DMIC2R" },
{ "AIF1ADC2R Mixer", "AIF2 Switch", "AIF2DACR" },
/* Pin level routing for AIF3 */
{ "AIF1DAC1L", NULL, "AIF1DAC Mux" },
{ "AIF1DAC1R", NULL, "AIF1DAC Mux" },
{ "AIF1DAC2L", NULL, "AIF1DAC Mux" },
{ "AIF1DAC2R", NULL, "AIF1DAC Mux" },
{ "AIF1DAC Mux", "AIF1DACDAT", "AIF1 Loopback" },
{ "AIF1DAC Mux", "AIF3DACDAT", "AIF3DACDAT" },
{ "AIF2DAC Mux", "AIF2DACDAT", "AIF2 Loopback" },
{ "AIF2DAC Mux", "AIF3DACDAT", "AIF3DACDAT" },
{ "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCL" },
{ "AIF2ADC Mux", "AIF2ADCDAT", "AIF2ADCR" },
{ "AIF2ADC Mux", "AIF3DACDAT", "AIF3ADCDAT" },
/* DAC1 inputs */
{ "DAC1L Mixer", "AIF2 Switch", "AIF2DACL" },
{ "DAC1L Mixer", "AIF1.2 Switch", "AIF1DAC2L" },
{ "DAC1L Mixer", "AIF1.1 Switch", "AIF1DAC1L" },
{ "DAC1L Mixer", "Left Sidetone Switch", "Left Sidetone" },
{ "DAC1L Mixer", "Right Sidetone Switch", "Right Sidetone" },
{ "DAC1R Mixer", "AIF2 Switch", "AIF2DACR" },
{ "DAC1R Mixer", "AIF1.2 Switch", "AIF1DAC2R" },
{ "DAC1R Mixer", "AIF1.1 Switch", "AIF1DAC1R" },
{ "DAC1R Mixer", "Left Sidetone Switch", "Left Sidetone" },
{ "DAC1R Mixer", "Right Sidetone Switch", "Right Sidetone" },
/* DAC2/AIF2 outputs */
{ "AIF2ADCL", NULL, "AIF2DAC2L Mixer" },
{ "AIF2DAC2L Mixer", "AIF2 Switch", "AIF2DACL" },
{ "AIF2DAC2L Mixer", "AIF1.2 Switch", "AIF1DAC2L" },
{ "AIF2DAC2L Mixer", "AIF1.1 Switch", "AIF1DAC1L" },
{ "AIF2DAC2L Mixer", "Left Sidetone Switch", "Left Sidetone" },
{ "AIF2DAC2L Mixer", "Right Sidetone Switch", "Right Sidetone" },
{ "AIF2ADCR", NULL, "AIF2DAC2R Mixer" },
{ "AIF2DAC2R Mixer", "AIF2 Switch", "AIF2DACR" },
{ "AIF2DAC2R Mixer", "AIF1.2 Switch", "AIF1DAC2R" },
{ "AIF2DAC2R Mixer", "AIF1.1 Switch", "AIF1DAC1R" },
{ "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" },
{ "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" },
{ "AIF1ADCDAT", NULL, "AIF1ADC1L" },
{ "AIF1ADCDAT", NULL, "AIF1ADC1R" },
{ "AIF1ADCDAT", NULL, "AIF1ADC2L" },
{ "AIF1ADCDAT", NULL, "AIF1ADC2R" },
{ "AIF2ADCDAT", NULL, "AIF2ADC Mux" },
/* AIF3 output */
{ "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC1L" },
{ "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC1R" },
{ "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC2L" },
{ "AIF3ADCDAT", "AIF1ADCDAT", "AIF1ADC2R" },
{ "AIF3ADCDAT", "AIF2ADCDAT", "AIF2ADCL" },
{ "AIF3ADCDAT", "AIF2ADCDAT", "AIF2ADCR" },
{ "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACL" },
{ "AIF3ADCDAT", "AIF2DACDAT", "AIF2DACR" },
/* Loopback */
{ "AIF1 Loopback", "ADCDAT", "AIF1ADCDAT" },
{ "AIF1 Loopback", "None", "AIF1DACDAT" },
{ "AIF2 Loopback", "ADCDAT", "AIF2ADCDAT" },
{ "AIF2 Loopback", "None", "AIF2DACDAT" },
/* Sidetone */
{ "Left Sidetone", "ADC/DMIC1", "ADCL Mux" },
{ "Left Sidetone", "DMIC2", "DMIC2L" },
{ "Right Sidetone", "ADC/DMIC1", "ADCR Mux" },
{ "Right Sidetone", "DMIC2", "DMIC2R" },
/* Output stages */
{ "Left Output Mixer", "DAC Switch", "DAC1L" },
{ "Right Output Mixer", "DAC Switch", "DAC1R" },
{ "SPKL", "DAC1 Switch", "DAC1L" },
{ "SPKL", "DAC2 Switch", "DAC2L" },
{ "SPKR", "DAC1 Switch", "DAC1R" },
{ "SPKR", "DAC2 Switch", "DAC2R" },
{ "Left Headphone Mux", "DAC", "DAC1L" },
{ "Right Headphone Mux", "DAC", "DAC1R" },
};
static const struct snd_soc_dapm_route wm8994_lateclk_revd_intercon[] = {
{ "DAC1L", NULL, "Late DAC1L Enable PGA" },
{ "Late DAC1L Enable PGA", NULL, "DAC1L Mixer" },
{ "DAC1R", NULL, "Late DAC1R Enable PGA" },
{ "Late DAC1R Enable PGA", NULL, "DAC1R Mixer" },
{ "DAC2L", NULL, "Late DAC2L Enable PGA" },
{ "Late DAC2L Enable PGA", NULL, "AIF2DAC2L Mixer" },
{ "DAC2R", NULL, "Late DAC2R Enable PGA" },
{ "Late DAC2R Enable PGA", NULL, "AIF2DAC2R Mixer" }
};
static const struct snd_soc_dapm_route wm8994_lateclk_intercon[] = {
{ "DAC1L", NULL, "DAC1L Mixer" },
{ "DAC1R", NULL, "DAC1R Mixer" },
{ "DAC2L", NULL, "AIF2DAC2L Mixer" },
{ "DAC2R", NULL, "AIF2DAC2R Mixer" },
};
static const struct snd_soc_dapm_route wm8994_revd_intercon[] = {
{ "AIF1DACDAT", NULL, "AIF2DACDAT" },
{ "AIF2DACDAT", NULL, "AIF1DACDAT" },
{ "AIF1ADCDAT", NULL, "AIF2ADCDAT" },
{ "AIF2ADCDAT", NULL, "AIF1ADCDAT" },
{ "MICBIAS1", NULL, "CLK_SYS" },
{ "MICBIAS1", NULL, "MICBIAS Supply" },
{ "MICBIAS2", NULL, "CLK_SYS" },
{ "MICBIAS2", NULL, "MICBIAS Supply" },
};
static const struct snd_soc_dapm_route wm8994_intercon[] = {
{ "AIF2DACL", NULL, "AIF2DAC Mux" },
{ "AIF2DACR", NULL, "AIF2DAC Mux" },
{ "MICBIAS1", NULL, "VMID" },
{ "MICBIAS2", NULL, "VMID" },
};
static const struct snd_soc_dapm_route wm8958_intercon[] = {
{ "AIF2DACL", NULL, "AIF2DACL Mux" },
{ "AIF2DACR", NULL, "AIF2DACR Mux" },
{ "AIF2DACL Mux", "AIF2", "AIF2DAC Mux" },
{ "AIF2DACL Mux", "AIF3", "AIF3DACDAT" },
{ "AIF2DACR Mux", "AIF2", "AIF2DAC Mux" },
{ "AIF2DACR Mux", "AIF3", "AIF3DACDAT" },
{ "AIF3DACDAT", NULL, "AIF3" },
{ "AIF3ADCDAT", NULL, "AIF3" },
{ "Mono PCM Out Mux", "AIF2ADCL", "AIF2ADCL" },
{ "Mono PCM Out Mux", "AIF2ADCR", "AIF2ADCR" },
{ "AIF3ADC Mux", "Mono PCM", "Mono PCM Out Mux" },
};
/* The size in bits of the FLL divide multiplied by 10
* to allow rounding later */
#define FIXED_FLL_SIZE ((1 << 16) * 10)
struct fll_div {
u16 outdiv;
u16 n;
u16 k;
u16 lambda;
u16 clk_ref_div;
u16 fll_fratio;
};
static int wm8994_get_fll_config(struct wm8994 *control, struct fll_div *fll,
int freq_in, int freq_out)
{
u64 Kpart;
unsigned int K, Ndiv, Nmod, gcd_fll;
pr_debug("FLL input=%dHz, output=%dHz\n", freq_in, freq_out);
/* Scale the input frequency down to <= 13.5MHz */
fll->clk_ref_div = 0;
while (freq_in > 13500000) {
fll->clk_ref_div++;
freq_in /= 2;
if (fll->clk_ref_div > 3)
return -EINVAL;
}
pr_debug("CLK_REF_DIV=%d, Fref=%dHz\n", fll->clk_ref_div, freq_in);
/* Scale the output to give 90MHz<=Fvco<=100MHz */
fll->outdiv = 3;
while (freq_out * (fll->outdiv + 1) < 90000000) {
fll->outdiv++;
if (fll->outdiv > 63)
return -EINVAL;
}
freq_out *= fll->outdiv + 1;
pr_debug("OUTDIV=%d, Fvco=%dHz\n", fll->outdiv, freq_out);
if (freq_in > 1000000) {
fll->fll_fratio = 0;
} else if (freq_in > 256000) {
fll->fll_fratio = 1;
freq_in *= 2;
} else if (freq_in > 128000) {
fll->fll_fratio = 2;
freq_in *= 4;
} else if (freq_in > 64000) {
fll->fll_fratio = 3;
freq_in *= 8;
} else {
fll->fll_fratio = 4;
freq_in *= 16;
}
pr_debug("FLL_FRATIO=%d, Fref=%dHz\n", fll->fll_fratio, freq_in);
/* Now, calculate N.K */
Ndiv = freq_out / freq_in;
fll->n = Ndiv;
Nmod = freq_out % freq_in;
pr_debug("Nmod=%d\n", Nmod);
switch (control->type) {
case WM8994:
/* Calculate fractional part - scale up so we can round. */
Kpart = FIXED_FLL_SIZE * (long long)Nmod;
do_div(Kpart, freq_in);
K = Kpart & 0xFFFFFFFF;
if ((K % 10) >= 5)
K += 5;
/* Move down to proper range now rounding is done */
fll->k = K / 10;
fll->lambda = 0;
pr_debug("N=%x K=%x\n", fll->n, fll->k);
break;
default:
gcd_fll = gcd(freq_out, freq_in);
fll->k = (freq_out - (freq_in * fll->n)) / gcd_fll;
fll->lambda = freq_in / gcd_fll;
}
return 0;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
unsigned int freq_in, unsigned int freq_out)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
int reg_offset, ret;
struct fll_div fll;
u16 reg, clk1, aif_reg, aif_src;
unsigned long timeout;
bool was_enabled;
switch (id) {
case WM8994_FLL1:
reg_offset = 0;
id = 0;
aif_src = 0x10;
break;
case WM8994_FLL2:
reg_offset = 0x20;
id = 1;
aif_src = 0x18;
break;
default:
return -EINVAL;
}
reg = snd_soc_read(codec, WM8994_FLL1_CONTROL_1 + reg_offset);
was_enabled = reg & WM8994_FLL1_ENA;
switch (src) {
case 0:
/* Allow no source specification when stopping */
if (freq_out)
return -EINVAL;
src = wm8994->fll[id].src;
break;
case WM8994_FLL_SRC_MCLK1:
case WM8994_FLL_SRC_MCLK2:
case WM8994_FLL_SRC_LRCLK:
case WM8994_FLL_SRC_BCLK:
break;
case WM8994_FLL_SRC_INTERNAL:
freq_in = 12000000;
freq_out = 12000000;
break;
default:
return -EINVAL;
}
/* Are we changing anything? */
if (wm8994->fll[id].src == src &&
wm8994->fll[id].in == freq_in && wm8994->fll[id].out == freq_out)
return 0;
/* If we're stopping the FLL redo the old config - no
* registers will actually be written but we avoid GCC flow
* analysis bugs spewing warnings.
*/
if (freq_out)
ret = wm8994_get_fll_config(control, &fll, freq_in, freq_out);
else
ret = wm8994_get_fll_config(control, &fll, wm8994->fll[id].in,
wm8994->fll[id].out);
if (ret < 0)
return ret;
/* Make sure that we're not providing SYSCLK right now */
clk1 = snd_soc_read(codec, WM8994_CLOCKING_1);
if (clk1 & WM8994_SYSCLK_SRC)
aif_reg = WM8994_AIF2_CLOCKING_1;
else
aif_reg = WM8994_AIF1_CLOCKING_1;
reg = snd_soc_read(codec, aif_reg);
if ((reg & WM8994_AIF1CLK_ENA) &&
(reg & WM8994_AIF1CLK_SRC_MASK) == aif_src) {
dev_err(codec->dev, "FLL%d is currently providing SYSCLK\n",
id + 1);
return -EBUSY;
}
/* We always need to disable the FLL while reconfiguring */
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset,
WM8994_FLL1_ENA, 0);
if (wm8994->fll_byp && src == WM8994_FLL_SRC_BCLK &&
freq_in == freq_out && freq_out) {
dev_dbg(codec->dev, "Bypassing FLL%d\n", id + 1);
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset,
WM8958_FLL1_BYP, WM8958_FLL1_BYP);
goto out;
}
reg = (fll.outdiv << WM8994_FLL1_OUTDIV_SHIFT) |
(fll.fll_fratio << WM8994_FLL1_FRATIO_SHIFT);
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_2 + reg_offset,
WM8994_FLL1_OUTDIV_MASK |
WM8994_FLL1_FRATIO_MASK, reg);
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_3 + reg_offset,
WM8994_FLL1_K_MASK, fll.k);
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_4 + reg_offset,
WM8994_FLL1_N_MASK,
fll.n << WM8994_FLL1_N_SHIFT);
if (fll.lambda) {
snd_soc_update_bits(codec, WM8958_FLL1_EFS_1 + reg_offset,
WM8958_FLL1_LAMBDA_MASK,
fll.lambda);
snd_soc_update_bits(codec, WM8958_FLL1_EFS_2 + reg_offset,
WM8958_FLL1_EFS_ENA, WM8958_FLL1_EFS_ENA);
} else {
snd_soc_update_bits(codec, WM8958_FLL1_EFS_2 + reg_offset,
WM8958_FLL1_EFS_ENA, 0);
}
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_5 + reg_offset,
WM8994_FLL1_FRC_NCO | WM8958_FLL1_BYP |
WM8994_FLL1_REFCLK_DIV_MASK |
WM8994_FLL1_REFCLK_SRC_MASK,
((src == WM8994_FLL_SRC_INTERNAL)
<< WM8994_FLL1_FRC_NCO_SHIFT) |
(fll.clk_ref_div << WM8994_FLL1_REFCLK_DIV_SHIFT) |
(src - 1));
/* Clear any pending completion from a previous failure */
try_wait_for_completion(&wm8994->fll_locked[id]);
/* Enable (with fractional mode if required) */
if (freq_out) {
/* Enable VMID if we need it */
if (!was_enabled) {
active_reference(codec);
switch (control->type) {
case WM8994:
vmid_reference(codec);
break;
case WM8958:
if (control->revision < 1)
vmid_reference(codec);
break;
default:
break;
}
}
reg = WM8994_FLL1_ENA;
if (fll.k)
reg |= WM8994_FLL1_FRAC;
if (src == WM8994_FLL_SRC_INTERNAL)
reg |= WM8994_FLL1_OSC_ENA;
snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset,
WM8994_FLL1_ENA | WM8994_FLL1_OSC_ENA |
WM8994_FLL1_FRAC, reg);
if (wm8994->fll_locked_irq) {
timeout = wait_for_completion_timeout(&wm8994->fll_locked[id],
msecs_to_jiffies(10));
if (timeout == 0)
dev_warn(codec->dev,
"Timed out waiting for FLL lock\n");
} else {
msleep(5);
}
} else {
if (was_enabled) {
switch (control->type) {
case WM8994:
vmid_dereference(codec);
break;
case WM8958:
if (control->revision < 1)
vmid_dereference(codec);
break;
default:
break;
}
active_dereference(codec);
}
}
out:
wm8994->fll[id].in = freq_in;
wm8994->fll[id].out = freq_out;
wm8994->fll[id].src = src;
configure_clock(codec);
/*
* If SYSCLK will be less than 50kHz adjust AIFnCLK dividers
* for detection.
*/
if (max(wm8994->aifclk[0], wm8994->aifclk[1]) < 50000) {
dev_dbg(codec->dev, "Configuring AIFs for 128fs\n");
wm8994->aifdiv[0] = snd_soc_read(codec, WM8994_AIF1_RATE)
& WM8994_AIF1CLK_RATE_MASK;
wm8994->aifdiv[1] = snd_soc_read(codec, WM8994_AIF2_RATE)
& WM8994_AIF1CLK_RATE_MASK;
snd_soc_update_bits(codec, WM8994_AIF1_RATE,
WM8994_AIF1CLK_RATE_MASK, 0x1);
snd_soc_update_bits(codec, WM8994_AIF2_RATE,
WM8994_AIF2CLK_RATE_MASK, 0x1);
} else if (wm8994->aifdiv[0]) {
snd_soc_update_bits(codec, WM8994_AIF1_RATE,
WM8994_AIF1CLK_RATE_MASK,
wm8994->aifdiv[0]);
snd_soc_update_bits(codec, WM8994_AIF2_RATE,
WM8994_AIF2CLK_RATE_MASK,
wm8994->aifdiv[1]);
wm8994->aifdiv[0] = 0;
wm8994->aifdiv[1] = 0;
}
return 0;
}
static irqreturn_t wm8994_fll_locked_irq(int irq, void *data)
{
struct completion *completion = data;
complete(completion);
return IRQ_HANDLED;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static int opclk_divs[] = { 10, 20, 30, 40, 55, 60, 80, 120, 160 };
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static int wm8994_set_fll(struct snd_soc_dai *dai, int id, int src,
unsigned int freq_in, unsigned int freq_out)
{
return _wm8994_set_fll(dai->codec, id, src, freq_in, freq_out);
}
static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = dai->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int i;
switch (dai->id) {
case 1:
case 2:
break;
default:
/* AIF3 shares clocking with AIF1/2 */
return -EINVAL;
}
switch (clk_id) {
case WM8994_SYSCLK_MCLK1:
wm8994->sysclk[dai->id - 1] = WM8994_SYSCLK_MCLK1;
wm8994->mclk[0] = freq;
dev_dbg(dai->dev, "AIF%d using MCLK1 at %uHz\n",
dai->id, freq);
break;
case WM8994_SYSCLK_MCLK2:
/* TODO: Set GPIO AF */
wm8994->sysclk[dai->id - 1] = WM8994_SYSCLK_MCLK2;
wm8994->mclk[1] = freq;
dev_dbg(dai->dev, "AIF%d using MCLK2 at %uHz\n",
dai->id, freq);
break;
case WM8994_SYSCLK_FLL1:
wm8994->sysclk[dai->id - 1] = WM8994_SYSCLK_FLL1;
dev_dbg(dai->dev, "AIF%d using FLL1\n", dai->id);
break;
case WM8994_SYSCLK_FLL2:
wm8994->sysclk[dai->id - 1] = WM8994_SYSCLK_FLL2;
dev_dbg(dai->dev, "AIF%d using FLL2\n", dai->id);
break;
case WM8994_SYSCLK_OPCLK:
/* Special case - a division (times 10) is given and
* no effect on main clocking.
*/
if (freq) {
for (i = 0; i < ARRAY_SIZE(opclk_divs); i++)
if (opclk_divs[i] == freq)
break;
if (i == ARRAY_SIZE(opclk_divs))
return -EINVAL;
snd_soc_update_bits(codec, WM8994_CLOCKING_2,
WM8994_OPCLK_DIV_MASK, i);
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_2,
WM8994_OPCLK_ENA, WM8994_OPCLK_ENA);
} else {
snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_2,
WM8994_OPCLK_ENA, 0);
}
default:
return -EINVAL;
}
configure_clock(codec);
/*
* If SYSCLK will be less than 50kHz adjust AIFnCLK dividers
* for detection.
*/
if (max(wm8994->aifclk[0], wm8994->aifclk[1]) < 50000) {
dev_dbg(codec->dev, "Configuring AIFs for 128fs\n");
wm8994->aifdiv[0] = snd_soc_read(codec, WM8994_AIF1_RATE)
& WM8994_AIF1CLK_RATE_MASK;
wm8994->aifdiv[1] = snd_soc_read(codec, WM8994_AIF2_RATE)
& WM8994_AIF1CLK_RATE_MASK;
snd_soc_update_bits(codec, WM8994_AIF1_RATE,
WM8994_AIF1CLK_RATE_MASK, 0x1);
snd_soc_update_bits(codec, WM8994_AIF2_RATE,
WM8994_AIF2CLK_RATE_MASK, 0x1);
} else if (wm8994->aifdiv[0]) {
snd_soc_update_bits(codec, WM8994_AIF1_RATE,
WM8994_AIF1CLK_RATE_MASK,
wm8994->aifdiv[0]);
snd_soc_update_bits(codec, WM8994_AIF2_RATE,
WM8994_AIF2CLK_RATE_MASK,
wm8994->aifdiv[1]);
wm8994->aifdiv[0] = 0;
wm8994->aifdiv[1] = 0;
}
return 0;
}
static int wm8994_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
wm_hubs_set_bias_level(codec, level);
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
/* MICBIAS into regulating mode */
switch (control->type) {
case WM8958:
case WM1811:
snd_soc_update_bits(codec, WM8958_MICBIAS1,
WM8958_MICB1_MODE, 0);
snd_soc_update_bits(codec, WM8958_MICBIAS2,
WM8958_MICB2_MODE, 0);
break;
default:
break;
}
if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
active_reference(codec);
break;
case SND_SOC_BIAS_STANDBY:
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
switch (control->type) {
case WM8958:
if (control->revision == 0) {
/* Optimise performance for rev A */
snd_soc_update_bits(codec,
WM8958_CHARGE_PUMP_2,
WM8958_CP_DISCH,
WM8958_CP_DISCH);
}
break;
default:
break;
}
/* Discharge LINEOUT1 & 2 */
snd_soc_update_bits(codec, WM8994_ANTIPOP_1,
WM8994_LINEOUT1_DISCH |
WM8994_LINEOUT2_DISCH,
WM8994_LINEOUT1_DISCH |
WM8994_LINEOUT2_DISCH);
}
if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE)
active_dereference(codec);
/* MICBIAS into bypass mode on newer devices */
switch (control->type) {
case WM8958:
case WM1811:
snd_soc_update_bits(codec, WM8958_MICBIAS1,
WM8958_MICB1_MODE,
WM8958_MICB1_MODE);
snd_soc_update_bits(codec, WM8958_MICBIAS2,
WM8958_MICB2_MODE,
WM8958_MICB2_MODE);
break;
default:
break;
}
break;
case SND_SOC_BIAS_OFF:
if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY)
wm8994->cur_fw = NULL;
break;
}
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
codec->dapm.bias_level = level;
return 0;
}
int wm8994_vmid_mode(struct snd_soc_codec *codec, enum wm8994_vmid_mode mode)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
switch (mode) {
case WM8994_VMID_NORMAL:
if (wm8994->hubs.lineout1_se) {
snd_soc_dapm_disable_pin(&codec->dapm,
"LINEOUT1N Driver");
snd_soc_dapm_disable_pin(&codec->dapm,
"LINEOUT1P Driver");
}
if (wm8994->hubs.lineout2_se) {
snd_soc_dapm_disable_pin(&codec->dapm,
"LINEOUT2N Driver");
snd_soc_dapm_disable_pin(&codec->dapm,
"LINEOUT2P Driver");
}
/* Do the sync with the old mode to allow it to clean up */
snd_soc_dapm_sync(&codec->dapm);
wm8994->vmid_mode = mode;
break;
case WM8994_VMID_FORCE:
if (wm8994->hubs.lineout1_se) {
snd_soc_dapm_force_enable_pin(&codec->dapm,
"LINEOUT1N Driver");
snd_soc_dapm_force_enable_pin(&codec->dapm,
"LINEOUT1P Driver");
}
if (wm8994->hubs.lineout2_se) {
snd_soc_dapm_force_enable_pin(&codec->dapm,
"LINEOUT2N Driver");
snd_soc_dapm_force_enable_pin(&codec->dapm,
"LINEOUT2P Driver");
}
wm8994->vmid_mode = mode;
snd_soc_dapm_sync(&codec->dapm);
break;
default:
return -EINVAL;
}
return 0;
}
static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct snd_soc_codec *codec = dai->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
int ms_reg;
int aif1_reg;
int dac_reg;
int adc_reg;
int ms = 0;
int aif1 = 0;
int lrclk = 0;
switch (dai->id) {
case 1:
ms_reg = WM8994_AIF1_MASTER_SLAVE;
aif1_reg = WM8994_AIF1_CONTROL_1;
dac_reg = WM8994_AIF1DAC_LRCLK;
adc_reg = WM8994_AIF1ADC_LRCLK;
break;
case 2:
ms_reg = WM8994_AIF2_MASTER_SLAVE;
aif1_reg = WM8994_AIF2_CONTROL_1;
dac_reg = WM8994_AIF1DAC_LRCLK;
adc_reg = WM8994_AIF1ADC_LRCLK;
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
break;
case SND_SOC_DAIFMT_CBM_CFM:
ms = WM8994_AIF1_MSTR;
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_B:
aif1 |= WM8994_AIF1_LRCLK_INV;
lrclk |= WM8958_AIF1_LRCLK_INV;
case SND_SOC_DAIFMT_DSP_A:
aif1 |= 0x18;
break;
case SND_SOC_DAIFMT_I2S:
aif1 |= 0x10;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
aif1 |= 0x8;
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_A:
case SND_SOC_DAIFMT_DSP_B:
/* frame inversion not valid for DSP modes */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_IB_NF:
aif1 |= WM8994_AIF1_BCLK_INV;
break;
default:
return -EINVAL;
}
break;
case SND_SOC_DAIFMT_I2S:
case SND_SOC_DAIFMT_RIGHT_J:
case SND_SOC_DAIFMT_LEFT_J:
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_IB_IF:
aif1 |= WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV;
lrclk |= WM8958_AIF1_LRCLK_INV;
break;
case SND_SOC_DAIFMT_IB_NF:
aif1 |= WM8994_AIF1_BCLK_INV;
break;
case SND_SOC_DAIFMT_NB_IF:
aif1 |= WM8994_AIF1_LRCLK_INV;
lrclk |= WM8958_AIF1_LRCLK_INV;
break;
default:
return -EINVAL;
}
break;
default:
return -EINVAL;
}
/* The AIF2 format configuration needs to be mirrored to AIF3
* on WM8958 if it's in use so just do it all the time. */
switch (control->type) {
case WM1811:
case WM8958:
if (dai->id == 2)
snd_soc_update_bits(codec, WM8958_AIF3_CONTROL_1,
WM8994_AIF1_LRCLK_INV |
WM8958_AIF3_FMT_MASK, aif1);
break;
default:
break;
}
snd_soc_update_bits(codec, aif1_reg,
WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV |
WM8994_AIF1_FMT_MASK,
aif1);
snd_soc_update_bits(codec, ms_reg, WM8994_AIF1_MSTR,
ms);
snd_soc_update_bits(codec, dac_reg,
WM8958_AIF1_LRCLK_INV, lrclk);
snd_soc_update_bits(codec, adc_reg,
WM8958_AIF1_LRCLK_INV, lrclk);
return 0;
}
static struct {
int val, rate;
} srs[] = {
{ 0, 8000 },
{ 1, 11025 },
{ 2, 12000 },
{ 3, 16000 },
{ 4, 22050 },
{ 5, 24000 },
{ 6, 32000 },
{ 7, 44100 },
{ 8, 48000 },
{ 9, 88200 },
{ 10, 96000 },
};
static int fs_ratios[] = {
64, 128, 192, 256, 348, 512, 768, 1024, 1408, 1536
};
static int bclk_divs[] = {
10, 15, 20, 30, 40, 50, 60, 80, 110, 120, 160, 220, 240, 320, 440, 480,
640, 880, 960, 1280, 1760, 1920
};
static int wm8994_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
struct wm8994_pdata *pdata = &control->pdata;
int aif1_reg;
int aif2_reg;
int bclk_reg;
int lrclk_reg;
int rate_reg;
int aif1 = 0;
int aif2 = 0;
int bclk = 0;
int lrclk = 0;
int rate_val = 0;
int id = dai->id - 1;
int i, cur_val, best_val, bclk_rate, best;
switch (dai->id) {
case 1:
aif1_reg = WM8994_AIF1_CONTROL_1;
aif2_reg = WM8994_AIF1_CONTROL_2;
bclk_reg = WM8994_AIF1_BCLK;
rate_reg = WM8994_AIF1_RATE;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
wm8994->lrclk_shared[0]) {
lrclk_reg = WM8994_AIF1DAC_LRCLK;
} else {
lrclk_reg = WM8994_AIF1ADC_LRCLK;
dev_dbg(codec->dev, "AIF1 using split LRCLK\n");
}
break;
case 2:
aif1_reg = WM8994_AIF2_CONTROL_1;
aif2_reg = WM8994_AIF2_CONTROL_2;
bclk_reg = WM8994_AIF2_BCLK;
rate_reg = WM8994_AIF2_RATE;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
wm8994->lrclk_shared[1]) {
lrclk_reg = WM8994_AIF2DAC_LRCLK;
} else {
lrclk_reg = WM8994_AIF2ADC_LRCLK;
dev_dbg(codec->dev, "AIF2 using split LRCLK\n");
}
break;
default:
return -EINVAL;
}
bclk_rate = params_rate(params);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
bclk_rate *= 16;
break;
case SNDRV_PCM_FORMAT_S20_3LE:
bclk_rate *= 20;
aif1 |= 0x20;
break;
case SNDRV_PCM_FORMAT_S24_LE:
bclk_rate *= 24;
aif1 |= 0x40;
break;
case SNDRV_PCM_FORMAT_S32_LE:
bclk_rate *= 32;
aif1 |= 0x60;
break;
default:
return -EINVAL;
}
wm8994->channels[id] = params_channels(params);
if (pdata->max_channels_clocked[id] &&
wm8994->channels[id] > pdata->max_channels_clocked[id]) {
dev_dbg(dai->dev, "Constraining channels to %d from %d\n",
pdata->max_channels_clocked[id], wm8994->channels[id]);
wm8994->channels[id] = pdata->max_channels_clocked[id];
}
switch (wm8994->channels[id]) {
case 1:
case 2:
bclk_rate *= 2;
break;
default:
bclk_rate *= 4;
break;
}
/* Try to find an appropriate sample rate; look for an exact match. */
for (i = 0; i < ARRAY_SIZE(srs); i++)
if (srs[i].rate == params_rate(params))
break;
if (i == ARRAY_SIZE(srs))
return -EINVAL;
rate_val |= srs[i].val << WM8994_AIF1_SR_SHIFT;
dev_dbg(dai->dev, "Sample rate is %dHz\n", srs[i].rate);
dev_dbg(dai->dev, "AIF%dCLK is %dHz, target BCLK %dHz\n",
dai->id, wm8994->aifclk[id], bclk_rate);
if (wm8994->channels[id] == 1 &&
(snd_soc_read(codec, aif1_reg) & 0x18) == 0x18)
aif2 |= WM8994_AIF1_MONO;
if (wm8994->aifclk[id] == 0) {
dev_err(dai->dev, "AIF%dCLK not configured\n", dai->id);
return -EINVAL;
}
/* AIFCLK/fs ratio; look for a close match in either direction */
best = 0;
best_val = abs((fs_ratios[0] * params_rate(params))
- wm8994->aifclk[id]);
for (i = 1; i < ARRAY_SIZE(fs_ratios); i++) {
cur_val = abs((fs_ratios[i] * params_rate(params))
- wm8994->aifclk[id]);
if (cur_val >= best_val)
continue;
best = i;
best_val = cur_val;
}
dev_dbg(dai->dev, "Selected AIF%dCLK/fs = %d\n",
dai->id, fs_ratios[best]);
rate_val |= best;
/* We may not get quite the right frequency if using
* approximate clocks so look for the closest match that is
* higher than the target (we need to ensure that there enough
* BCLKs to clock out the samples).
*/
best = 0;
for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) {
cur_val = (wm8994->aifclk[id] * 10 / bclk_divs[i]) - bclk_rate;
if (cur_val < 0) /* BCLK table is sorted */
break;
best = i;
}
bclk_rate = wm8994->aifclk[id] * 10 / bclk_divs[best];
dev_dbg(dai->dev, "Using BCLK_DIV %d for actual BCLK %dHz\n",
bclk_divs[best], bclk_rate);
bclk |= best << WM8994_AIF1_BCLK_DIV_SHIFT;
lrclk = bclk_rate / params_rate(params);
if (!lrclk) {
dev_err(dai->dev, "Unable to generate LRCLK from %dHz BCLK\n",
bclk_rate);
return -EINVAL;
}
dev_dbg(dai->dev, "Using LRCLK rate %d for actual LRCLK %dHz\n",
lrclk, bclk_rate / lrclk);
snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1);
snd_soc_update_bits(codec, aif2_reg, WM8994_AIF1_MONO, aif2);
snd_soc_update_bits(codec, bclk_reg, WM8994_AIF1_BCLK_DIV_MASK, bclk);
snd_soc_update_bits(codec, lrclk_reg, WM8994_AIF1DAC_RATE_MASK,
lrclk);
snd_soc_update_bits(codec, rate_reg, WM8994_AIF1_SR_MASK |
WM8994_AIF1CLK_RATE_MASK, rate_val);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
switch (dai->id) {
case 1:
wm8994->dac_rates[0] = params_rate(params);
wm8994_set_retune_mobile(codec, 0);
wm8994_set_retune_mobile(codec, 1);
break;
case 2:
wm8994->dac_rates[1] = params_rate(params);
wm8994_set_retune_mobile(codec, 2);
break;
}
}
return 0;
}
static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
int aif1_reg;
int aif1 = 0;
switch (dai->id) {
case 3:
switch (control->type) {
case WM1811:
case WM8958:
aif1_reg = WM8958_AIF3_CONTROL_1;
break;
default:
return 0;
}
break;
default:
return 0;
}
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
aif1 |= 0x20;
break;
case SNDRV_PCM_FORMAT_S24_LE:
aif1 |= 0x40;
break;
case SNDRV_PCM_FORMAT_S32_LE:
aif1 |= 0x60;
break;
default:
return -EINVAL;
}
return snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1);
}
static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute)
{
struct snd_soc_codec *codec = codec_dai->codec;
int mute_reg;
int reg;
switch (codec_dai->id) {
case 1:
mute_reg = WM8994_AIF1_DAC1_FILTERS_1;
break;
case 2:
mute_reg = WM8994_AIF2_DAC_FILTERS_1;
break;
default:
return -EINVAL;
}
if (mute)
reg = WM8994_AIF1DAC1_MUTE;
else
reg = 0;
snd_soc_update_bits(codec, mute_reg, WM8994_AIF1DAC1_MUTE, reg);
return 0;
}
static int wm8994_set_tristate(struct snd_soc_dai *codec_dai, int tristate)
{
struct snd_soc_codec *codec = codec_dai->codec;
int reg, val, mask;
switch (codec_dai->id) {
case 1:
reg = WM8994_AIF1_MASTER_SLAVE;
mask = WM8994_AIF1_TRI;
break;
case 2:
reg = WM8994_AIF2_MASTER_SLAVE;
mask = WM8994_AIF2_TRI;
break;
default:
return -EINVAL;
}
if (tristate)
val = mask;
else
val = 0;
return snd_soc_update_bits(codec, reg, mask, val);
}
static int wm8994_aif2_probe(struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
/* Disable the pulls on the AIF if we're using it to save power. */
snd_soc_update_bits(codec, WM8994_GPIO_3,
WM8994_GPN_PU | WM8994_GPN_PD, 0);
snd_soc_update_bits(codec, WM8994_GPIO_4,
WM8994_GPN_PU | WM8994_GPN_PD, 0);
snd_soc_update_bits(codec, WM8994_GPIO_5,
WM8994_GPN_PU | WM8994_GPN_PD, 0);
return 0;
}
#define WM8994_RATES SNDRV_PCM_RATE_8000_96000
#define WM8994_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
static const struct snd_soc_dai_ops wm8994_aif1_dai_ops = {
.set_sysclk = wm8994_set_dai_sysclk,
.set_fmt = wm8994_set_dai_fmt,
.hw_params = wm8994_hw_params,
.digital_mute = wm8994_aif_mute,
.set_pll = wm8994_set_fll,
.set_tristate = wm8994_set_tristate,
};
static const struct snd_soc_dai_ops wm8994_aif2_dai_ops = {
.set_sysclk = wm8994_set_dai_sysclk,
.set_fmt = wm8994_set_dai_fmt,
.hw_params = wm8994_hw_params,
.digital_mute = wm8994_aif_mute,
.set_pll = wm8994_set_fll,
.set_tristate = wm8994_set_tristate,
};
static const struct snd_soc_dai_ops wm8994_aif3_dai_ops = {
.hw_params = wm8994_aif3_hw_params,
};
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static struct snd_soc_dai_driver wm8994_dai[] = {
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
.name = "wm8994-aif1",
.id = 1,
.playback = {
.stream_name = "AIF1 Playback",
.channels_min = 1,
.channels_max = 2,
.rates = WM8994_RATES,
.formats = WM8994_FORMATS,
.sig_bits = 24,
},
.capture = {
.stream_name = "AIF1 Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8994_RATES,
.formats = WM8994_FORMATS,
.sig_bits = 24,
},
.ops = &wm8994_aif1_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>
2010-03-18 04:15:21 +08:00
.name = "wm8994-aif2",
.id = 2,
.playback = {
.stream_name = "AIF2 Playback",
.channels_min = 1,
.channels_max = 2,
.rates = WM8994_RATES,
.formats = WM8994_FORMATS,
.sig_bits = 24,
},
.capture = {
.stream_name = "AIF2 Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8994_RATES,
.formats = WM8994_FORMATS,
.sig_bits = 24,
},
.probe = wm8994_aif2_probe,
.ops = &wm8994_aif2_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>
2010-03-18 04:15:21 +08:00
.name = "wm8994-aif3",
.id = 3,
.playback = {
.stream_name = "AIF3 Playback",
.channels_min = 1,
.channels_max = 2,
.rates = WM8994_RATES,
.formats = WM8994_FORMATS,
.sig_bits = 24,
},
.capture = {
.stream_name = "AIF3 Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8994_RATES,
.formats = WM8994_FORMATS,
.sig_bits = 24,
},
.ops = &wm8994_aif3_dai_ops,
}
};
#ifdef CONFIG_PM
static int wm8994_codec_suspend(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int i, ret;
for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) {
memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i],
sizeof(struct wm8994_fll_config));
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
ret = _wm8994_set_fll(codec, i + 1, 0, 0, 0);
if (ret < 0)
dev_warn(codec->dev, "Failed to stop FLL%d: %d\n",
i + 1, ret);
}
wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int wm8994_codec_resume(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int i, ret;
for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) {
if (!wm8994->fll_suspend[i].out)
continue;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
ret = _wm8994_set_fll(codec, i + 1,
wm8994->fll_suspend[i].src,
wm8994->fll_suspend[i].in,
wm8994->fll_suspend[i].out);
if (ret < 0)
dev_warn(codec->dev, "Failed to restore FLL%d: %d\n",
i + 1, ret);
}
return 0;
}
#else
#define wm8994_codec_suspend NULL
#define wm8994_codec_resume NULL
#endif
static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994)
{
struct snd_soc_codec *codec = wm8994->hubs.codec;
struct wm8994 *control = wm8994->wm8994;
struct wm8994_pdata *pdata = &control->pdata;
struct snd_kcontrol_new controls[] = {
SOC_ENUM_EXT("AIF1.1 EQ Mode",
wm8994->retune_mobile_enum,
wm8994_get_retune_mobile_enum,
wm8994_put_retune_mobile_enum),
SOC_ENUM_EXT("AIF1.2 EQ Mode",
wm8994->retune_mobile_enum,
wm8994_get_retune_mobile_enum,
wm8994_put_retune_mobile_enum),
SOC_ENUM_EXT("AIF2 EQ Mode",
wm8994->retune_mobile_enum,
wm8994_get_retune_mobile_enum,
wm8994_put_retune_mobile_enum),
};
int ret, i, j;
const char **t;
/* We need an array of texts for the enum API but the number
* of texts is likely to be less than the number of
* configurations due to the sample rate dependency of the
* configurations. */
wm8994->num_retune_mobile_texts = 0;
wm8994->retune_mobile_texts = NULL;
for (i = 0; i < pdata->num_retune_mobile_cfgs; i++) {
for (j = 0; j < wm8994->num_retune_mobile_texts; j++) {
if (strcmp(pdata->retune_mobile_cfgs[i].name,
wm8994->retune_mobile_texts[j]) == 0)
break;
}
if (j != wm8994->num_retune_mobile_texts)
continue;
/* Expand the array... */
t = krealloc(wm8994->retune_mobile_texts,
sizeof(char *) *
(wm8994->num_retune_mobile_texts + 1),
GFP_KERNEL);
if (t == NULL)
continue;
/* ...store the new entry... */
t[wm8994->num_retune_mobile_texts] =
pdata->retune_mobile_cfgs[i].name;
/* ...and remember the new version. */
wm8994->num_retune_mobile_texts++;
wm8994->retune_mobile_texts = t;
}
dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n",
wm8994->num_retune_mobile_texts);
wm8994->retune_mobile_enum.max = wm8994->num_retune_mobile_texts;
wm8994->retune_mobile_enum.texts = wm8994->retune_mobile_texts;
ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls,
ARRAY_SIZE(controls));
if (ret != 0)
dev_err(wm8994->hubs.codec->dev,
"Failed to add ReTune Mobile controls: %d\n", ret);
}
static void wm8994_handle_pdata(struct wm8994_priv *wm8994)
{
struct snd_soc_codec *codec = wm8994->hubs.codec;
struct wm8994 *control = wm8994->wm8994;
struct wm8994_pdata *pdata = &control->pdata;
int ret, i;
if (!pdata)
return;
wm_hubs_handle_analogue_pdata(codec, pdata->lineout1_diff,
pdata->lineout2_diff,
pdata->lineout1fb,
pdata->lineout2fb,
pdata->jd_scthr,
pdata->jd_thr,
pdata->micb1_delay,
pdata->micb2_delay,
pdata->micbias1_lvl,
pdata->micbias2_lvl);
dev_dbg(codec->dev, "%d DRC configurations\n", pdata->num_drc_cfgs);
if (pdata->num_drc_cfgs) {
struct snd_kcontrol_new controls[] = {
SOC_ENUM_EXT("AIF1DRC1 Mode", wm8994->drc_enum,
wm8994_get_drc_enum, wm8994_put_drc_enum),
SOC_ENUM_EXT("AIF1DRC2 Mode", wm8994->drc_enum,
wm8994_get_drc_enum, wm8994_put_drc_enum),
SOC_ENUM_EXT("AIF2DRC Mode", wm8994->drc_enum,
wm8994_get_drc_enum, wm8994_put_drc_enum),
};
/* We need an array of texts for the enum API */
wm8994->drc_texts = devm_kzalloc(wm8994->hubs.codec->dev,
sizeof(char *) * pdata->num_drc_cfgs, GFP_KERNEL);
if (!wm8994->drc_texts) {
dev_err(wm8994->hubs.codec->dev,
"Failed to allocate %d DRC config texts\n",
pdata->num_drc_cfgs);
return;
}
for (i = 0; i < pdata->num_drc_cfgs; i++)
wm8994->drc_texts[i] = pdata->drc_cfgs[i].name;
wm8994->drc_enum.max = pdata->num_drc_cfgs;
wm8994->drc_enum.texts = wm8994->drc_texts;
ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls,
ARRAY_SIZE(controls));
for (i = 0; i < WM8994_NUM_DRC; i++)
wm8994_set_drc(codec, i);
} else {
ret = snd_soc_add_codec_controls(wm8994->hubs.codec,
wm8994_drc_controls,
ARRAY_SIZE(wm8994_drc_controls));
}
if (ret != 0)
dev_err(wm8994->hubs.codec->dev,
"Failed to add DRC mode controls: %d\n", ret);
dev_dbg(codec->dev, "%d ReTune Mobile configurations\n",
pdata->num_retune_mobile_cfgs);
if (pdata->num_retune_mobile_cfgs)
wm8994_handle_retune_mobile_pdata(wm8994);
else
snd_soc_add_codec_controls(wm8994->hubs.codec, wm8994_eq_controls,
ARRAY_SIZE(wm8994_eq_controls));
for (i = 0; i < ARRAY_SIZE(pdata->micbias); i++) {
if (pdata->micbias[i]) {
snd_soc_write(codec, WM8958_MICBIAS1 + i,
pdata->micbias[i] & 0xffff);
}
}
}
/**
* wm8994_mic_detect - Enable microphone detection via the WM8994 IRQ
*
* @codec: WM8994 codec
* @jack: jack to report detection events on
* @micbias: microphone bias to detect on
*
* Enable microphone detection via IRQ on the WM8994. If GPIOs are
* being used to bring out signals to the processor then only platform
* data configuration is needed for WM8994 and processor GPIOs should
* be configured using snd_soc_jack_add_gpios() instead.
*
* Configuration of detection levels is available via the micbias1_lvl
* and micbias2_lvl platform data members.
*/
int wm8994_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
int micbias)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994_micdet *micdet;
struct wm8994 *control = wm8994->wm8994;
int reg, ret;
if (control->type != WM8994) {
dev_warn(codec->dev, "Not a WM8994\n");
return -EINVAL;
}
switch (micbias) {
case 1:
micdet = &wm8994->micdet[0];
if (jack)
ret = snd_soc_dapm_force_enable_pin(&codec->dapm,
"MICBIAS1");
else
ret = snd_soc_dapm_disable_pin(&codec->dapm,
"MICBIAS1");
break;
case 2:
micdet = &wm8994->micdet[1];
if (jack)
ret = snd_soc_dapm_force_enable_pin(&codec->dapm,
"MICBIAS1");
else
ret = snd_soc_dapm_disable_pin(&codec->dapm,
"MICBIAS1");
break;
default:
dev_warn(codec->dev, "Invalid MICBIAS %d\n", micbias);
return -EINVAL;
}
if (ret != 0)
dev_warn(codec->dev, "Failed to configure MICBIAS%d: %d\n",
micbias, ret);
dev_dbg(codec->dev, "Configuring microphone detection on %d %p\n",
micbias, jack);
/* Store the configuration */
micdet->jack = jack;
micdet->detecting = true;
/* If either of the jacks is set up then enable detection */
if (wm8994->micdet[0].jack || wm8994->micdet[1].jack)
reg = WM8994_MICD_ENA;
else
reg = 0;
snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, reg);
/* enable MICDET and MICSHRT deboune */
snd_soc_update_bits(codec, WM8994_IRQ_DEBOUNCE,
WM8994_MIC1_DET_DB_MASK | WM8994_MIC1_SHRT_DB_MASK |
WM8994_MIC2_DET_DB_MASK | WM8994_MIC2_SHRT_DB_MASK,
WM8994_MIC1_DET_DB | WM8994_MIC1_SHRT_DB);
snd_soc_dapm_sync(&codec->dapm);
return 0;
}
EXPORT_SYMBOL_GPL(wm8994_mic_detect);
static void wm8994_mic_work(struct work_struct *work)
{
struct wm8994_priv *priv = container_of(work,
struct wm8994_priv,
mic_work.work);
struct regmap *regmap = priv->wm8994->regmap;
struct device *dev = priv->wm8994->dev;
unsigned int reg;
int ret;
int report;
pm_runtime_get_sync(dev);
ret = regmap_read(regmap, WM8994_INTERRUPT_RAW_STATUS_2, &reg);
if (ret < 0) {
dev_err(dev, "Failed to read microphone status: %d\n",
ret);
pm_runtime_put(dev);
return;
}
dev_dbg(dev, "Microphone status: %x\n", reg);
report = 0;
if (reg & WM8994_MIC1_DET_STS) {
if (priv->micdet[0].detecting)
report = SND_JACK_HEADSET;
}
if (reg & WM8994_MIC1_SHRT_STS) {
if (priv->micdet[0].detecting)
report = SND_JACK_HEADPHONE;
else
report |= SND_JACK_BTN_0;
}
if (report)
priv->micdet[0].detecting = false;
else
priv->micdet[0].detecting = true;
snd_soc_jack_report(priv->micdet[0].jack, report,
SND_JACK_HEADSET | SND_JACK_BTN_0);
report = 0;
if (reg & WM8994_MIC2_DET_STS) {
if (priv->micdet[1].detecting)
report = SND_JACK_HEADSET;
}
if (reg & WM8994_MIC2_SHRT_STS) {
if (priv->micdet[1].detecting)
report = SND_JACK_HEADPHONE;
else
report |= SND_JACK_BTN_0;
}
if (report)
priv->micdet[1].detecting = false;
else
priv->micdet[1].detecting = true;
snd_soc_jack_report(priv->micdet[1].jack, report,
SND_JACK_HEADSET | SND_JACK_BTN_0);
pm_runtime_put(dev);
}
static irqreturn_t wm8994_mic_irq(int irq, void *data)
{
struct wm8994_priv *priv = data;
struct snd_soc_codec *codec = priv->hubs.codec;
#ifndef CONFIG_SND_SOC_WM8994_MODULE
trace_snd_soc_jack_irq(dev_name(codec->dev));
#endif
pm_wakeup_event(codec->dev, 300);
queue_delayed_work(system_power_efficient_wq,
&priv->mic_work, msecs_to_jiffies(250));
return IRQ_HANDLED;
}
static void wm1811_micd_stop(struct snd_soc_codec *codec)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
if (!wm8994->jackdet)
return;
mutex_lock(&wm8994->accdet_lock);
snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0);
wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_JACK);
mutex_unlock(&wm8994->accdet_lock);
if (wm8994->wm8994->pdata.jd_ext_cap)
snd_soc_dapm_disable_pin(&codec->dapm,
"MICBIAS2");
}
static void wm8958_button_det(struct snd_soc_codec *codec, u16 status)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
int report;
report = 0;
if (status & 0x4)
report |= SND_JACK_BTN_0;
if (status & 0x8)
report |= SND_JACK_BTN_1;
if (status & 0x10)
report |= SND_JACK_BTN_2;
if (status & 0x20)
report |= SND_JACK_BTN_3;
if (status & 0x40)
report |= SND_JACK_BTN_4;
if (status & 0x80)
report |= SND_JACK_BTN_5;
snd_soc_jack_report(wm8994->micdet[0].jack, report,
wm8994->btn_mask);
}
static void wm8958_open_circuit_work(struct work_struct *work)
{
struct wm8994_priv *wm8994 = container_of(work,
struct wm8994_priv,
open_circuit_work.work);
struct device *dev = wm8994->wm8994->dev;
wm1811_micd_stop(wm8994->hubs.codec);
mutex_lock(&wm8994->accdet_lock);
dev_dbg(dev, "Reporting open circuit\n");
wm8994->jack_mic = false;
wm8994->mic_detecting = true;
wm8958_micd_set_rate(wm8994->hubs.codec);
snd_soc_jack_report(wm8994->micdet[0].jack, 0,
wm8994->btn_mask |
SND_JACK_HEADSET);
mutex_unlock(&wm8994->accdet_lock);
}
static void wm8958_mic_id(void *data, u16 status)
{
struct snd_soc_codec *codec = data;
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
/* Either nothing present or just starting detection */
if (!(status & WM8958_MICD_STS)) {
/* If nothing present then clear our statuses */
dev_dbg(codec->dev, "Detected open circuit\n");
queue_delayed_work(system_power_efficient_wq,
&wm8994->open_circuit_work,
msecs_to_jiffies(2500));
return;
}
/* If the measurement is showing a high impedence we've got a
* microphone.
*/
if (status & 0x600) {
dev_dbg(codec->dev, "Detected microphone\n");
wm8994->mic_detecting = false;
wm8994->jack_mic = true;
wm8958_micd_set_rate(codec);
snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADSET,
SND_JACK_HEADSET);
}
if (status & 0xfc) {
dev_dbg(codec->dev, "Detected headphone\n");
wm8994->mic_detecting = false;
wm8958_micd_set_rate(codec);
/* If we have jackdet that will detect removal */
wm1811_micd_stop(codec);
snd_soc_jack_report(wm8994->micdet[0].jack, SND_JACK_HEADPHONE,
SND_JACK_HEADSET);
}
}
/* Deferred mic detection to allow for extra settling time */
static void wm1811_mic_work(struct work_struct *work)
{
struct wm8994_priv *wm8994 = container_of(work, struct wm8994_priv,
mic_work.work);
struct wm8994 *control = wm8994->wm8994;
struct snd_soc_codec *codec = wm8994->hubs.codec;
pm_runtime_get_sync(codec->dev);
/* If required for an external cap force MICBIAS on */
if (control->pdata.jd_ext_cap) {
snd_soc_dapm_force_enable_pin(&codec->dapm,
"MICBIAS2");
snd_soc_dapm_sync(&codec->dapm);
}
mutex_lock(&wm8994->accdet_lock);
dev_dbg(codec->dev, "Starting mic detection\n");
/* Use a user-supplied callback if we have one */
if (wm8994->micd_cb) {
wm8994->micd_cb(wm8994->micd_cb_data);
} else {
/*
* Start off measument of microphone impedence to find out
* what's actually there.
*/
wm8994->mic_detecting = true;
wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_MIC);
snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
WM8958_MICD_ENA, WM8958_MICD_ENA);
}
mutex_unlock(&wm8994->accdet_lock);
pm_runtime_put(codec->dev);
}
static irqreturn_t wm1811_jackdet_irq(int irq, void *data)
{
struct wm8994_priv *wm8994 = data;
struct wm8994 *control = wm8994->wm8994;
struct snd_soc_codec *codec = wm8994->hubs.codec;
int reg, delay;
bool present;
pm_runtime_get_sync(codec->dev);
cancel_delayed_work_sync(&wm8994->mic_complete_work);
mutex_lock(&wm8994->accdet_lock);
reg = snd_soc_read(codec, WM1811_JACKDET_CTRL);
if (reg < 0) {
dev_err(codec->dev, "Failed to read jack status: %d\n", reg);
mutex_unlock(&wm8994->accdet_lock);
pm_runtime_put(codec->dev);
return IRQ_NONE;
}
dev_dbg(codec->dev, "JACKDET %x\n", reg);
present = reg & WM1811_JACKDET_LVL;
if (present) {
dev_dbg(codec->dev, "Jack detected\n");
wm8958_micd_set_rate(codec);
snd_soc_update_bits(codec, WM8958_MICBIAS2,
WM8958_MICB2_DISCH, 0);
/* Disable debounce while inserted */
snd_soc_update_bits(codec, WM1811_JACKDET_CTRL,
WM1811_JACKDET_DB, 0);
delay = control->pdata.micdet_delay;
queue_delayed_work(system_power_efficient_wq,
&wm8994->mic_work,
msecs_to_jiffies(delay));
} else {
dev_dbg(codec->dev, "Jack not detected\n");
cancel_delayed_work_sync(&wm8994->mic_work);
snd_soc_update_bits(codec, WM8958_MICBIAS2,
WM8958_MICB2_DISCH, WM8958_MICB2_DISCH);
/* Enable debounce while removed */
snd_soc_update_bits(codec, WM1811_JACKDET_CTRL,
WM1811_JACKDET_DB, WM1811_JACKDET_DB);
wm8994->mic_detecting = false;
wm8994->jack_mic = false;
snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
WM8958_MICD_ENA, 0);
wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_JACK);
}
mutex_unlock(&wm8994->accdet_lock);
/* Turn off MICBIAS if it was on for an external cap */
if (control->pdata.jd_ext_cap && !present)
snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS2");
if (present)
snd_soc_jack_report(wm8994->micdet[0].jack,
SND_JACK_MECHANICAL, SND_JACK_MECHANICAL);
else
snd_soc_jack_report(wm8994->micdet[0].jack, 0,
SND_JACK_MECHANICAL | SND_JACK_HEADSET |
wm8994->btn_mask);
/* Since we only report deltas force an update, ensures we
* avoid bootstrapping issues with the core. */
snd_soc_jack_report(wm8994->micdet[0].jack, 0, 0);
pm_runtime_put(codec->dev);
return IRQ_HANDLED;
}
static void wm1811_jackdet_bootstrap(struct work_struct *work)
{
struct wm8994_priv *wm8994 = container_of(work,
struct wm8994_priv,
jackdet_bootstrap.work);
wm1811_jackdet_irq(0, wm8994);
}
/**
* wm8958_mic_detect - Enable microphone detection via the WM8958 IRQ
*
* @codec: WM8958 codec
* @jack: jack to report detection events on
*
* Enable microphone detection functionality for the WM8958. By
* default simple detection which supports the detection of up to 6
* buttons plus video and microphone functionality is supported.
*
* The WM8958 has an advanced jack detection facility which is able to
* support complex accessory detection, especially when used in
* conjunction with external circuitry. In order to provide maximum
* flexiblity a callback is provided which allows a completely custom
* detection algorithm.
*/
int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
wm1811_micdet_cb det_cb, void *det_cb_data,
wm1811_mic_id_cb id_cb, void *id_cb_data)
{
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
u16 micd_lvl_sel;
switch (control->type) {
case WM1811:
case WM8958:
break;
default:
return -EINVAL;
}
if (jack) {
snd_soc_dapm_force_enable_pin(&codec->dapm, "CLK_SYS");
snd_soc_dapm_sync(&codec->dapm);
wm8994->micdet[0].jack = jack;
if (det_cb) {
wm8994->micd_cb = det_cb;
wm8994->micd_cb_data = det_cb_data;
} else {
wm8994->mic_detecting = true;
wm8994->jack_mic = false;
}
if (id_cb) {
wm8994->mic_id_cb = id_cb;
wm8994->mic_id_cb_data = id_cb_data;
} else {
wm8994->mic_id_cb = wm8958_mic_id;
wm8994->mic_id_cb_data = codec;
}
wm8958_micd_set_rate(codec);
/* Detect microphones and short circuits by default */
if (control->pdata.micd_lvl_sel)
micd_lvl_sel = control->pdata.micd_lvl_sel;
else
micd_lvl_sel = 0x41;
wm8994->btn_mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 |
SND_JACK_BTN_2 | SND_JACK_BTN_3 |
SND_JACK_BTN_4 | SND_JACK_BTN_5;
snd_soc_update_bits(codec, WM8958_MIC_DETECT_2,
WM8958_MICD_LVL_SEL_MASK, micd_lvl_sel);
WARN_ON(codec->dapm.bias_level > SND_SOC_BIAS_STANDBY);
/*
* If we can use jack detection start off with that,
* otherwise jump straight to microphone detection.
*/
if (wm8994->jackdet) {
/* Disable debounce for the initial detect */
snd_soc_update_bits(codec, WM1811_JACKDET_CTRL,
WM1811_JACKDET_DB, 0);
snd_soc_update_bits(codec, WM8958_MICBIAS2,
WM8958_MICB2_DISCH,
WM8958_MICB2_DISCH);
snd_soc_update_bits(codec, WM8994_LDO_1,
WM8994_LDO1_DISCH, 0);
wm1811_jackdet_set_mode(codec,
WM1811_JACKDET_MODE_JACK);
} else {
snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
WM8958_MICD_ENA, WM8958_MICD_ENA);
}
} else {
snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
WM8958_MICD_ENA, 0);
wm1811_jackdet_set_mode(codec, WM1811_JACKDET_MODE_NONE);
snd_soc_dapm_disable_pin(&codec->dapm, "CLK_SYS");
snd_soc_dapm_sync(&codec->dapm);
}
return 0;
}
EXPORT_SYMBOL_GPL(wm8958_mic_detect);
static void wm8958_mic_work(struct work_struct *work)
{
struct wm8994_priv *wm8994 = container_of(work,
struct wm8994_priv,
mic_complete_work.work);
struct snd_soc_codec *codec = wm8994->hubs.codec;
pm_runtime_get_sync(codec->dev);
mutex_lock(&wm8994->accdet_lock);
wm8994->mic_id_cb(wm8994->mic_id_cb_data, wm8994->mic_status);
mutex_unlock(&wm8994->accdet_lock);
pm_runtime_put(codec->dev);
}
static irqreturn_t wm8958_mic_irq(int irq, void *data)
{
struct wm8994_priv *wm8994 = data;
struct snd_soc_codec *codec = wm8994->hubs.codec;
int reg, count, ret, id_delay;
/*
* Jack detection may have detected a removal simulataneously
* with an update of the MICDET status; if so it will have
* stopped detection and we can ignore this interrupt.
*/
if (!(snd_soc_read(codec, WM8958_MIC_DETECT_1) & WM8958_MICD_ENA))
return IRQ_HANDLED;
cancel_delayed_work_sync(&wm8994->mic_complete_work);
cancel_delayed_work_sync(&wm8994->open_circuit_work);
pm_runtime_get_sync(codec->dev);
/* We may occasionally read a detection without an impedence
* range being provided - if that happens loop again.
*/
count = 10;
do {
reg = snd_soc_read(codec, WM8958_MIC_DETECT_3);
if (reg < 0) {
dev_err(codec->dev,
"Failed to read mic detect status: %d\n",
reg);
pm_runtime_put(codec->dev);
return IRQ_NONE;
}
if (!(reg & WM8958_MICD_VALID)) {
dev_dbg(codec->dev, "Mic detect data not valid\n");
goto out;
}
if (!(reg & WM8958_MICD_STS) || (reg & WM8958_MICD_LVL_MASK))
break;
msleep(1);
} while (count--);
if (count == 0)
dev_warn(codec->dev, "No impedance range reported for jack\n");
#ifndef CONFIG_SND_SOC_WM8994_MODULE
trace_snd_soc_jack_irq(dev_name(codec->dev));
#endif
/* Avoid a transient report when the accessory is being removed */
if (wm8994->jackdet) {
ret = snd_soc_read(codec, WM1811_JACKDET_CTRL);
if (ret < 0) {
dev_err(codec->dev, "Failed to read jack status: %d\n",
ret);
} else if (!(ret & WM1811_JACKDET_LVL)) {
dev_dbg(codec->dev, "Ignoring removed jack\n");
goto out;
}
} else if (!(reg & WM8958_MICD_STS)) {
snd_soc_jack_report(wm8994->micdet[0].jack, 0,
SND_JACK_MECHANICAL | SND_JACK_HEADSET |
wm8994->btn_mask);
wm8994->mic_detecting = true;
goto out;
}
wm8994->mic_status = reg;
id_delay = wm8994->wm8994->pdata.mic_id_delay;
if (wm8994->mic_detecting)
queue_delayed_work(system_power_efficient_wq,
&wm8994->mic_complete_work,
msecs_to_jiffies(id_delay));
else
wm8958_button_det(codec, reg);
out:
pm_runtime_put(codec->dev);
return IRQ_HANDLED;
}
static irqreturn_t wm8994_fifo_error(int irq, void *data)
{
struct snd_soc_codec *codec = data;
dev_err(codec->dev, "FIFO error\n");
return IRQ_HANDLED;
}
static irqreturn_t wm8994_temp_warn(int irq, void *data)
{
struct snd_soc_codec *codec = data;
dev_err(codec->dev, "Thermal warning\n");
return IRQ_HANDLED;
}
static irqreturn_t wm8994_temp_shut(int irq, void *data)
{
struct snd_soc_codec *codec = data;
dev_crit(codec->dev, "Thermal shutdown\n");
return IRQ_HANDLED;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static int wm8994_codec_probe(struct snd_soc_codec *codec)
{
struct wm8994 *control = dev_get_drvdata(codec->dev->parent);
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
struct snd_soc_dapm_context *dapm = &codec->dapm;
unsigned int reg;
int ret, i;
wm8994->hubs.codec = codec;
codec->control_data = control->regmap;
snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP);
mutex_init(&wm8994->accdet_lock);
INIT_DELAYED_WORK(&wm8994->jackdet_bootstrap,
wm1811_jackdet_bootstrap);
INIT_DELAYED_WORK(&wm8994->open_circuit_work,
wm8958_open_circuit_work);
switch (control->type) {
case WM8994:
INIT_DELAYED_WORK(&wm8994->mic_work, wm8994_mic_work);
break;
case WM1811:
INIT_DELAYED_WORK(&wm8994->mic_work, wm1811_mic_work);
break;
default:
break;
}
INIT_DELAYED_WORK(&wm8994->mic_complete_work, wm8958_mic_work);
for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++)
init_completion(&wm8994->fll_locked[i]);
wm8994->micdet_irq = control->pdata.micdet_irq;
/* By default use idle_bias_off, will override for WM8994 */
codec->dapm.idle_bias_off = 1;
/* Set revision-specific configuration */
switch (control->type) {
case WM8994:
/* Single ended line outputs should have VMID on. */
if (!control->pdata.lineout1_diff ||
!control->pdata.lineout2_diff)
codec->dapm.idle_bias_off = 0;
switch (control->revision) {
case 2:
case 3:
wm8994->hubs.dcs_codes_l = -5;
wm8994->hubs.dcs_codes_r = -5;
wm8994->hubs.hp_startup_mode = 1;
wm8994->hubs.dcs_readback_mode = 1;
wm8994->hubs.series_startup = 1;
break;
default:
wm8994->hubs.dcs_readback_mode = 2;
break;
}
break;
case WM8958:
wm8994->hubs.dcs_readback_mode = 1;
wm8994->hubs.hp_startup_mode = 1;
switch (control->revision) {
case 0:
break;
default:
wm8994->fll_byp = true;
break;
}
break;
case WM1811:
wm8994->hubs.dcs_readback_mode = 2;
wm8994->hubs.no_series_update = 1;
wm8994->hubs.hp_startup_mode = 1;
wm8994->hubs.no_cache_dac_hp_direct = true;
wm8994->fll_byp = true;
wm8994->hubs.dcs_codes_l = -9;
wm8994->hubs.dcs_codes_r = -7;
snd_soc_update_bits(codec, WM8994_ANALOGUE_HP_1,
WM1811_HPOUT1_ATTN, WM1811_HPOUT1_ATTN);
break;
default:
break;
}
wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_FIFOS_ERR,
wm8994_fifo_error, "FIFO error", codec);
wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN,
wm8994_temp_warn, "Thermal warning", codec);
wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT,
wm8994_temp_shut, "Thermal shutdown", codec);
switch (control->type) {
case WM8994:
if (wm8994->micdet_irq) {
ret = request_threaded_irq(wm8994->micdet_irq, NULL,
wm8994_mic_irq,
IRQF_TRIGGER_RISING,
"Mic1 detect",
wm8994);
if (ret != 0)
dev_warn(codec->dev,
"Failed to request Mic1 detect IRQ: %d\n",
ret);
}
ret = wm8994_request_irq(wm8994->wm8994,
WM8994_IRQ_MIC1_SHRT,
wm8994_mic_irq, "Mic 1 short",
wm8994);
if (ret != 0)
dev_warn(codec->dev,
"Failed to request Mic1 short IRQ: %d\n",
ret);
ret = wm8994_request_irq(wm8994->wm8994,
WM8994_IRQ_MIC2_DET,
wm8994_mic_irq, "Mic 2 detect",
wm8994);
if (ret != 0)
dev_warn(codec->dev,
"Failed to request Mic2 detect IRQ: %d\n",
ret);
ret = wm8994_request_irq(wm8994->wm8994,
WM8994_IRQ_MIC2_SHRT,
wm8994_mic_irq, "Mic 2 short",
wm8994);
if (ret != 0)
dev_warn(codec->dev,
"Failed to request Mic2 short IRQ: %d\n",
ret);
break;
case WM8958:
case WM1811:
if (wm8994->micdet_irq) {
ret = request_threaded_irq(wm8994->micdet_irq, NULL,
wm8958_mic_irq,
IRQF_TRIGGER_RISING,
"Mic detect",
wm8994);
if (ret != 0)
dev_warn(codec->dev,
"Failed to request Mic detect IRQ: %d\n",
ret);
} else {
wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_MIC1_DET,
wm8958_mic_irq, "Mic detect",
wm8994);
}
}
switch (control->type) {
case WM1811:
if (control->cust_id > 1 || control->revision > 1) {
ret = wm8994_request_irq(wm8994->wm8994,
WM8994_IRQ_GPIO(6),
wm1811_jackdet_irq, "JACKDET",
wm8994);
if (ret == 0)
wm8994->jackdet = true;
}
break;
default:
break;
}
wm8994->fll_locked_irq = true;
for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) {
ret = wm8994_request_irq(wm8994->wm8994,
WM8994_IRQ_FLL1_LOCK + i,
wm8994_fll_locked_irq, "FLL lock",
&wm8994->fll_locked[i]);
if (ret != 0)
wm8994->fll_locked_irq = false;
}
/* Make sure we can read from the GPIOs if they're inputs */
pm_runtime_get_sync(codec->dev);
/* Remember if AIFnLRCLK is configured as a GPIO. This should be
* configured on init - if a system wants to do this dynamically
* at runtime we can deal with that then.
*/
ret = regmap_read(control->regmap, WM8994_GPIO_1, &reg);
if (ret < 0) {
dev_err(codec->dev, "Failed to read GPIO1 state: %d\n", ret);
goto err_irq;
}
if ((reg & WM8994_GPN_FN_MASK) != WM8994_GP_FN_PIN_SPECIFIC) {
wm8994->lrclk_shared[0] = 1;
wm8994_dai[0].symmetric_rates = 1;
} else {
wm8994->lrclk_shared[0] = 0;
}
ret = regmap_read(control->regmap, WM8994_GPIO_6, &reg);
if (ret < 0) {
dev_err(codec->dev, "Failed to read GPIO6 state: %d\n", ret);
goto err_irq;
}
if ((reg & WM8994_GPN_FN_MASK) != WM8994_GP_FN_PIN_SPECIFIC) {
wm8994->lrclk_shared[1] = 1;
wm8994_dai[1].symmetric_rates = 1;
} else {
wm8994->lrclk_shared[1] = 0;
}
pm_runtime_put(codec->dev);
/* Latch volume update bits */
for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++)
snd_soc_update_bits(codec, wm8994_vu_bits[i].reg,
wm8994_vu_bits[i].mask,
wm8994_vu_bits[i].mask);
/* Set the low bit of the 3D stereo depth so TLV matches */
snd_soc_update_bits(codec, WM8994_AIF1_DAC1_FILTERS_2,
1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT,
1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT);
snd_soc_update_bits(codec, WM8994_AIF1_DAC2_FILTERS_2,
1 << WM8994_AIF1DAC2_3D_GAIN_SHIFT,
1 << WM8994_AIF1DAC2_3D_GAIN_SHIFT);
snd_soc_update_bits(codec, WM8994_AIF2_DAC_FILTERS_2,
1 << WM8994_AIF2DAC_3D_GAIN_SHIFT,
1 << WM8994_AIF2DAC_3D_GAIN_SHIFT);
/* Unconditionally enable AIF1 ADC TDM mode on chips which can
* use this; it only affects behaviour on idle TDM clock
* cycles. */
switch (control->type) {
case WM8994:
case WM8958:
snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1,
WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM);
break;
default:
break;
}
/* Put MICBIAS into bypass mode by default on newer devices */
switch (control->type) {
case WM8958:
case WM1811:
snd_soc_update_bits(codec, WM8958_MICBIAS1,
WM8958_MICB1_MODE, WM8958_MICB1_MODE);
snd_soc_update_bits(codec, WM8958_MICBIAS2,
WM8958_MICB2_MODE, WM8958_MICB2_MODE);
break;
default:
break;
}
wm8994->hubs.check_class_w_digital = wm8994_check_class_w_digital;
wm_hubs_update_class_w(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>
2010-03-18 04:15:21 +08:00
wm8994_handle_pdata(wm8994);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
wm_hubs_add_analogue_controls(codec);
snd_soc_add_codec_controls(codec, wm8994_snd_controls,
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
ARRAY_SIZE(wm8994_snd_controls));
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
snd_soc_dapm_new_controls(dapm, wm8994_dapm_widgets,
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
ARRAY_SIZE(wm8994_dapm_widgets));
switch (control->type) {
case WM8994:
snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets,
ARRAY_SIZE(wm8994_specific_dapm_widgets));
if (control->revision < 4) {
snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets,
ARRAY_SIZE(wm8994_lateclk_revd_widgets));
snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets,
ARRAY_SIZE(wm8994_adc_revd_widgets));
snd_soc_dapm_new_controls(dapm, wm8994_dac_revd_widgets,
ARRAY_SIZE(wm8994_dac_revd_widgets));
} else {
snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets,
ARRAY_SIZE(wm8994_lateclk_widgets));
snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets,
ARRAY_SIZE(wm8994_adc_widgets));
snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets,
ARRAY_SIZE(wm8994_dac_widgets));
}
break;
case WM8958:
snd_soc_add_codec_controls(codec, wm8958_snd_controls,
ARRAY_SIZE(wm8958_snd_controls));
snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets,
ARRAY_SIZE(wm8958_dapm_widgets));
if (control->revision < 1) {
snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets,
ARRAY_SIZE(wm8994_lateclk_revd_widgets));
snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets,
ARRAY_SIZE(wm8994_adc_revd_widgets));
snd_soc_dapm_new_controls(dapm, wm8994_dac_revd_widgets,
ARRAY_SIZE(wm8994_dac_revd_widgets));
} else {
snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets,
ARRAY_SIZE(wm8994_lateclk_widgets));
snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets,
ARRAY_SIZE(wm8994_adc_widgets));
snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets,
ARRAY_SIZE(wm8994_dac_widgets));
}
break;
case WM1811:
snd_soc_add_codec_controls(codec, wm8958_snd_controls,
ARRAY_SIZE(wm8958_snd_controls));
snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets,
ARRAY_SIZE(wm8958_dapm_widgets));
snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets,
ARRAY_SIZE(wm8994_lateclk_widgets));
snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets,
ARRAY_SIZE(wm8994_adc_widgets));
snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets,
ARRAY_SIZE(wm8994_dac_widgets));
break;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
wm_hubs_add_analogue_routes(codec, 0, 0);
ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_DCS_DONE,
wm_hubs_dcs_done, "DC servo done",
&wm8994->hubs);
if (ret == 0)
wm8994->hubs.dcs_done_irq = true;
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
switch (control->type) {
case WM8994:
snd_soc_dapm_add_routes(dapm, wm8994_intercon,
ARRAY_SIZE(wm8994_intercon));
if (control->revision < 4) {
snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
ARRAY_SIZE(wm8994_revd_intercon));
snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon,
ARRAY_SIZE(wm8994_lateclk_revd_intercon));
} else {
snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon,
ARRAY_SIZE(wm8994_lateclk_intercon));
}
break;
case WM8958:
if (control->revision < 1) {
snd_soc_dapm_add_routes(dapm, wm8994_intercon,
ARRAY_SIZE(wm8994_intercon));
snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
ARRAY_SIZE(wm8994_revd_intercon));
snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon,
ARRAY_SIZE(wm8994_lateclk_revd_intercon));
} else {
snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon,
ARRAY_SIZE(wm8994_lateclk_intercon));
snd_soc_dapm_add_routes(dapm, wm8958_intercon,
ARRAY_SIZE(wm8958_intercon));
}
wm8958_dsp2_init(codec);
break;
case WM1811:
snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon,
ARRAY_SIZE(wm8994_lateclk_intercon));
snd_soc_dapm_add_routes(dapm, wm8958_intercon,
ARRAY_SIZE(wm8958_intercon));
break;
}
return 0;
err_irq:
if (wm8994->jackdet)
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_GPIO(6), wm8994);
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC2_SHRT, wm8994);
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC2_DET, wm8994);
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC1_SHRT, wm8994);
if (wm8994->micdet_irq)
free_irq(wm8994->micdet_irq, wm8994);
for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++)
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FLL1_LOCK + i,
&wm8994->fll_locked[i]);
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_DCS_DONE,
&wm8994->hubs);
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FIFOS_ERR, codec);
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, codec);
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, codec);
return ret;
}
static int wm8994_codec_remove(struct snd_soc_codec *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>
2010-03-18 04:15:21 +08:00
struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
struct wm8994 *control = wm8994->wm8994;
int i;
wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++)
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FLL1_LOCK + i,
&wm8994->fll_locked[i]);
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_DCS_DONE,
&wm8994->hubs);
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_FIFOS_ERR, codec);
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_SHUT, codec);
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_TEMP_WARN, codec);
if (wm8994->jackdet)
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_GPIO(6), wm8994);
switch (control->type) {
case WM8994:
if (wm8994->micdet_irq)
free_irq(wm8994->micdet_irq, wm8994);
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC2_DET,
wm8994);
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC1_SHRT,
wm8994);
wm8994_free_irq(wm8994->wm8994, WM8994_IRQ_MIC1_DET,
wm8994);
break;
case WM1811:
case WM8958:
if (wm8994->micdet_irq)
free_irq(wm8994->micdet_irq, wm8994);
break;
}
release_firmware(wm8994->mbc);
release_firmware(wm8994->mbc_vss);
release_firmware(wm8994->enh_eq);
kfree(wm8994->retune_mobile_texts);
return 0;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static struct snd_soc_codec_driver soc_codec_dev_wm8994 = {
.probe = wm8994_codec_probe,
.remove = wm8994_codec_remove,
.suspend = wm8994_codec_suspend,
.resume = wm8994_codec_resume,
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
.set_bias_level = wm8994_set_bias_level,
};
static int wm8994_probe(struct platform_device *pdev)
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
{
struct wm8994_priv *wm8994;
wm8994 = devm_kzalloc(&pdev->dev, sizeof(struct wm8994_priv),
GFP_KERNEL);
if (wm8994 == NULL)
return -ENOMEM;
platform_set_drvdata(pdev, wm8994);
wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent);
pm_runtime_enable(&pdev->dev);
pm_runtime_idle(&pdev->dev);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm8994,
wm8994_dai, ARRAY_SIZE(wm8994_dai));
}
static int wm8994_remove(struct platform_device *pdev)
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
{
snd_soc_unregister_codec(&pdev->dev);
pm_runtime_disable(&pdev->dev);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
return 0;
}
#ifdef CONFIG_PM_SLEEP
static int wm8994_suspend(struct device *dev)
{
struct wm8994_priv *wm8994 = dev_get_drvdata(dev);
/* Drop down to power saving mode when system is suspended */
if (wm8994->jackdet && !wm8994->active_refcount)
regmap_update_bits(wm8994->wm8994->regmap, WM8994_ANTIPOP_2,
WM1811_JACKDET_MODE_MASK,
wm8994->jackdet_mode);
return 0;
}
static int wm8994_resume(struct device *dev)
{
struct wm8994_priv *wm8994 = dev_get_drvdata(dev);
if (wm8994->jackdet && wm8994->jackdet_mode)
regmap_update_bits(wm8994->wm8994->regmap, WM8994_ANTIPOP_2,
WM1811_JACKDET_MODE_MASK,
WM1811_JACKDET_MODE_AUDIO);
return 0;
}
#endif
static const struct dev_pm_ops wm8994_pm_ops = {
SET_SYSTEM_SLEEP_PM_OPS(wm8994_suspend, wm8994_resume)
};
static struct platform_driver wm8994_codec_driver = {
.driver = {
.name = "wm8994-codec",
.owner = THIS_MODULE,
.pm = &wm8994_pm_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>
2010-03-18 04:15:21 +08:00
.probe = wm8994_probe,
.remove = wm8994_remove,
};
module_platform_driver(wm8994_codec_driver);
MODULE_DESCRIPTION("ASoC WM8994 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:wm8994-codec");