linux_dsm_epyc7002/sound/soc/codecs/tlv320aic3x.c

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/*
* ALSA SoC TLV320AIC3X codec driver
*
* Author: Vladimir Barinov, <vbarinov@embeddedalley.com>
* Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
*
* Based on sound/soc/codecs/wm8753.c by Liam Girdwood
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*
* Notes:
* The AIC3X is a driver for a low power stereo audio
* codecs aic31, aic32, aic33, aic3007.
*
* It supports full aic33 codec functionality.
* The compatibility with aic32, aic31 and aic3007 is as follows:
* aic32/aic3007 | aic31
* ---------------------------------------
* MONO_LOUT -> N/A | MONO_LOUT -> N/A
* | IN1L -> LINE1L
* | IN1R -> LINE1R
* | IN2L -> LINE2L
* | IN2R -> LINE2R
* | MIC3L/R -> N/A
* truncated internal functionality in
* accordance with documentation
* ---------------------------------------
*
* Hence the machine layer should disable unsupported inputs/outputs by
* snd_soc_dapm_disable_pin(codec, "MONO_LOUT"), etc.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/gpio.h>
#include <linux/regulator/consumer.h>
#include <linux/of_gpio.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 15:04:11 +07:00
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include <sound/tlv320aic3x.h>
#include "tlv320aic3x.h"
#define AIC3X_NUM_SUPPLIES 4
static const char *aic3x_supply_names[AIC3X_NUM_SUPPLIES] = {
"IOVDD", /* I/O Voltage */
"DVDD", /* Digital Core Voltage */
"AVDD", /* Analog DAC Voltage */
"DRVDD", /* ADC Analog and Output Driver Voltage */
};
static LIST_HEAD(reset_list);
struct aic3x_priv;
struct aic3x_disable_nb {
struct notifier_block nb;
struct aic3x_priv *aic3x;
};
/* codec private data */
struct aic3x_priv {
struct snd_soc_codec *codec;
struct regulator_bulk_data supplies[AIC3X_NUM_SUPPLIES];
struct aic3x_disable_nb disable_nb[AIC3X_NUM_SUPPLIES];
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
enum snd_soc_control_type control_type;
struct aic3x_setup_data *setup;
unsigned int sysclk;
struct list_head list;
int master;
int gpio_reset;
int power;
#define AIC3X_MODEL_3X 0
#define AIC3X_MODEL_33 1
#define AIC3X_MODEL_3007 2
u16 model;
/* Selects the micbias voltage */
enum aic3x_micbias_voltage micbias_vg;
};
/*
* AIC3X register cache
* We can't read the AIC3X register space when we are
* using 2 wire for device control, so we cache them instead.
* There is no point in caching the reset register
*/
static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = {
0x00, 0x00, 0x00, 0x10, /* 0 */
0x04, 0x00, 0x00, 0x00, /* 4 */
0x00, 0x00, 0x00, 0x01, /* 8 */
0x00, 0x00, 0x00, 0x80, /* 12 */
0x80, 0xff, 0xff, 0x78, /* 16 */
0x78, 0x78, 0x78, 0x78, /* 20 */
0x78, 0x00, 0x00, 0xfe, /* 24 */
0x00, 0x00, 0xfe, 0x00, /* 28 */
0x18, 0x18, 0x00, 0x00, /* 32 */
0x00, 0x00, 0x00, 0x00, /* 36 */
0x00, 0x00, 0x00, 0x80, /* 40 */
0x80, 0x00, 0x00, 0x00, /* 44 */
0x00, 0x00, 0x00, 0x04, /* 48 */
0x00, 0x00, 0x00, 0x00, /* 52 */
0x00, 0x00, 0x04, 0x00, /* 56 */
0x00, 0x00, 0x00, 0x00, /* 60 */
0x00, 0x04, 0x00, 0x00, /* 64 */
0x00, 0x00, 0x00, 0x00, /* 68 */
0x04, 0x00, 0x00, 0x00, /* 72 */
0x00, 0x00, 0x00, 0x00, /* 76 */
0x00, 0x00, 0x00, 0x00, /* 80 */
0x00, 0x00, 0x00, 0x00, /* 84 */
0x00, 0x00, 0x00, 0x00, /* 88 */
0x00, 0x00, 0x00, 0x00, /* 92 */
0x00, 0x00, 0x00, 0x00, /* 96 */
0x00, 0x00, 0x02, 0x00, /* 100 */
0x00, 0x00, 0x00, 0x00, /* 104 */
0x00, 0x00, /* 108 */
};
#define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \
SOC_SINGLE_EXT(xname, reg, shift, mask, invert, \
snd_soc_dapm_get_volsw, snd_soc_dapm_put_volsw_aic3x)
/*
* All input lines are connected when !0xf and disconnected with 0xf bit field,
* so we have to use specific dapm_put call for input mixer
*/
static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
struct snd_soc_dapm_widget *widget = wlist->widgets[0];
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
unsigned int reg = mc->reg;
unsigned int shift = mc->shift;
int max = mc->max;
unsigned int mask = (1 << fls(max)) - 1;
unsigned int invert = mc->invert;
unsigned short val, val_mask;
int ret;
struct snd_soc_dapm_path *path;
int found = 0;
val = (ucontrol->value.integer.value[0] & mask);
mask = 0xf;
if (val)
val = mask;
if (invert)
val = mask - val;
val_mask = mask << shift;
val = val << shift;
mutex_lock(&widget->codec->mutex);
if (snd_soc_test_bits(widget->codec, reg, val_mask, val)) {
/* find dapm widget path assoc with kcontrol */
list_for_each_entry(path, &widget->dapm->card->paths, list) {
if (path->kcontrol != kcontrol)
continue;
/* found, now check type */
found = 1;
if (val)
/* new connection */
path->connect = invert ? 0 : 1;
else
/* old connection must be powered down */
path->connect = invert ? 1 : 0;
dapm_mark_dirty(path->source, "tlv320aic3x source");
dapm_mark_dirty(path->sink, "tlv320aic3x sink");
break;
}
}
mutex_unlock(&widget->codec->mutex);
if (found)
snd_soc_dapm_sync(widget->dapm);
ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val);
return ret;
}
/*
* mic bias power on/off share the same register bits with
* output voltage of mic bias. when power on mic bias, we
* need reclaim it to voltage value.
* 0x0 = Powered off
* 0x1 = MICBIAS output is powered to 2.0V,
* 0x2 = MICBIAS output is powered to 2.5V
* 0x3 = MICBIAS output is connected to AVDD
*/
static int mic_bias_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
/* change mic bias voltage to user defined */
snd_soc_update_bits(codec, MICBIAS_CTRL,
MICBIAS_LEVEL_MASK,
aic3x->micbias_vg << MICBIAS_LEVEL_SHIFT);
break;
case SND_SOC_DAPM_PRE_PMD:
snd_soc_update_bits(codec, MICBIAS_CTRL,
MICBIAS_LEVEL_MASK, 0);
break;
}
return 0;
}
static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" };
static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" };
static const char *aic3x_left_hpcom_mux[] =
{ "differential of HPLOUT", "constant VCM", "single-ended" };
static const char *aic3x_right_hpcom_mux[] =
{ "differential of HPROUT", "constant VCM", "single-ended",
"differential of HPLCOM", "external feedback" };
static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" };
static const char *aic3x_adc_hpf[] =
{ "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" };
#define LDAC_ENUM 0
#define RDAC_ENUM 1
#define LHPCOM_ENUM 2
#define RHPCOM_ENUM 3
#define LINE1L_2_L_ENUM 4
#define LINE1L_2_R_ENUM 5
#define LINE1R_2_L_ENUM 6
#define LINE1R_2_R_ENUM 7
#define LINE2L_ENUM 8
#define LINE2R_ENUM 9
#define ADC_HPF_ENUM 10
static const struct soc_enum aic3x_enum[] = {
SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux),
SOC_ENUM_SINGLE(DAC_LINE_MUX, 4, 3, aic3x_right_dac_mux),
SOC_ENUM_SINGLE(HPLCOM_CFG, 4, 3, aic3x_left_hpcom_mux),
SOC_ENUM_SINGLE(HPRCOM_CFG, 3, 5, aic3x_right_hpcom_mux),
SOC_ENUM_SINGLE(LINE1L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
SOC_ENUM_SINGLE(LINE1L_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
SOC_ENUM_SINGLE(LINE1R_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
SOC_ENUM_SINGLE(LINE1R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux),
SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux),
SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf),
};
static const char *aic3x_agc_level[] =
{ "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" };
static const struct soc_enum aic3x_agc_level_enum[] = {
SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level),
SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level),
};
static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" };
static const struct soc_enum aic3x_agc_attack_enum[] = {
SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack),
SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack),
};
static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" };
static const struct soc_enum aic3x_agc_decay_enum[] = {
SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay),
SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay),
};
/*
* DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps
*/
static DECLARE_TLV_DB_SCALE(dac_tlv, -6350, 50, 0);
/* ADC PGA gain volumes. From 0 to 59.5 dB in 0.5 dB steps */
static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 50, 0);
/*
* Output stage volumes. From -78.3 to 0 dB. Muted below -78.3 dB.
* Step size is approximately 0.5 dB over most of the scale but increasing
* near the very low levels.
* Define dB scale so that it is mostly correct for range about -55 to 0 dB
* but having increasing dB difference below that (and where it doesn't count
* so much). This setting shows -50 dB (actual is -50.3 dB) for register
* value 100 and -58.5 dB (actual is -78.3 dB) for register value 117.
*/
static DECLARE_TLV_DB_SCALE(output_stage_tlv, -5900, 50, 1);
static const struct snd_kcontrol_new aic3x_snd_controls[] = {
/* Output */
SOC_DOUBLE_R_TLV("PCM Playback Volume",
LDAC_VOL, RDAC_VOL, 0, 0x7f, 1, dac_tlv),
/*
* Output controls that map to output mixer switches. Note these are
* only for swapped L-to-R and R-to-L routes. See below stereo controls
* for direct L-to-L and R-to-R routes.
*/
SOC_SINGLE_TLV("Left Line Mixer Line2R Bypass Volume",
LINE2R_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Left Line Mixer PGAR Bypass Volume",
PGAR_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Left Line Mixer DACR1 Playback Volume",
DACR1_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Right Line Mixer Line2L Bypass Volume",
LINE2L_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Right Line Mixer PGAL Bypass Volume",
PGAL_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Right Line Mixer DACL1 Playback Volume",
DACL1_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Left HP Mixer Line2R Bypass Volume",
LINE2R_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Left HP Mixer PGAR Bypass Volume",
PGAR_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Left HP Mixer DACR1 Playback Volume",
DACR1_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Right HP Mixer Line2L Bypass Volume",
LINE2L_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Right HP Mixer PGAL Bypass Volume",
PGAL_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Right HP Mixer DACL1 Playback Volume",
DACL1_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Left HPCOM Mixer Line2R Bypass Volume",
LINE2R_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Left HPCOM Mixer PGAR Bypass Volume",
PGAR_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Left HPCOM Mixer DACR1 Playback Volume",
DACR1_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Right HPCOM Mixer Line2L Bypass Volume",
LINE2L_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Right HPCOM Mixer PGAL Bypass Volume",
PGAL_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv),
SOC_SINGLE_TLV("Right HPCOM Mixer DACL1 Playback Volume",
DACL1_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv),
/* Stereo output controls for direct L-to-L and R-to-R routes */
SOC_DOUBLE_R_TLV("Line Line2 Bypass Volume",
LINE2L_2_LLOPM_VOL, LINE2R_2_RLOPM_VOL,
0, 118, 1, output_stage_tlv),
SOC_DOUBLE_R_TLV("Line PGA Bypass Volume",
PGAL_2_LLOPM_VOL, PGAR_2_RLOPM_VOL,
0, 118, 1, output_stage_tlv),
SOC_DOUBLE_R_TLV("Line DAC Playback Volume",
DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL,
0, 118, 1, output_stage_tlv),
SOC_DOUBLE_R_TLV("Mono Line2 Bypass Volume",
LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL,
0, 118, 1, output_stage_tlv),
SOC_DOUBLE_R_TLV("Mono PGA Bypass Volume",
PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL,
0, 118, 1, output_stage_tlv),
SOC_DOUBLE_R_TLV("Mono DAC Playback Volume",
DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL,
0, 118, 1, output_stage_tlv),
SOC_DOUBLE_R_TLV("HP Line2 Bypass Volume",
LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL,
0, 118, 1, output_stage_tlv),
SOC_DOUBLE_R_TLV("HP PGA Bypass Volume",
PGAL_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL,
0, 118, 1, output_stage_tlv),
SOC_DOUBLE_R_TLV("HP DAC Playback Volume",
DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL,
0, 118, 1, output_stage_tlv),
SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Volume",
LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL,
0, 118, 1, output_stage_tlv),
SOC_DOUBLE_R_TLV("HPCOM PGA Bypass Volume",
PGAL_2_HPLCOM_VOL, PGAR_2_HPRCOM_VOL,
0, 118, 1, output_stage_tlv),
SOC_DOUBLE_R_TLV("HPCOM DAC Playback Volume",
DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL,
0, 118, 1, output_stage_tlv),
/* Output pin mute controls */
SOC_DOUBLE_R("Line Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3,
0x01, 0),
SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0),
SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3,
0x01, 0),
SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3,
0x01, 0),
/*
* Note: enable Automatic input Gain Controller with care. It can
* adjust PGA to max value when ADC is on and will never go back.
*/
SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0),
SOC_ENUM("Left AGC Target level", aic3x_agc_level_enum[0]),
SOC_ENUM("Right AGC Target level", aic3x_agc_level_enum[1]),
SOC_ENUM("Left AGC Attack time", aic3x_agc_attack_enum[0]),
SOC_ENUM("Right AGC Attack time", aic3x_agc_attack_enum[1]),
SOC_ENUM("Left AGC Decay time", aic3x_agc_decay_enum[0]),
SOC_ENUM("Right AGC Decay time", aic3x_agc_decay_enum[1]),
/* De-emphasis */
SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0),
/* Input */
SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL,
0, 119, 0, adc_tlv),
SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1),
SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
};
/*
* Class-D amplifier gain. From 0 to 18 dB in 6 dB steps
*/
static DECLARE_TLV_DB_SCALE(classd_amp_tlv, 0, 600, 0);
static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl =
SOC_DOUBLE_TLV("Class-D Playback Volume", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv);
/* Left DAC Mux */
static const struct snd_kcontrol_new aic3x_left_dac_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]);
/* Right DAC Mux */
static const struct snd_kcontrol_new aic3x_right_dac_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_enum[RDAC_ENUM]);
/* Left HPCOM Mux */
static const struct snd_kcontrol_new aic3x_left_hpcom_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_enum[LHPCOM_ENUM]);
/* Right HPCOM Mux */
static const struct snd_kcontrol_new aic3x_right_hpcom_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_enum[RHPCOM_ENUM]);
/* Left Line Mixer */
static const struct snd_kcontrol_new aic3x_left_line_mixer_controls[] = {
SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_LLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_LLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_LLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_LLOPM_VOL, 7, 1, 0),
};
/* Right Line Mixer */
static const struct snd_kcontrol_new aic3x_right_line_mixer_controls[] = {
SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_RLOPM_VOL, 7, 1, 0),
};
/* Mono Mixer */
static const struct snd_kcontrol_new aic3x_mono_mixer_controls[] = {
SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_MONOLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_MONOLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_MONOLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_MONOLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_MONOLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_MONOLOPM_VOL, 7, 1, 0),
};
/* Left HP Mixer */
static const struct snd_kcontrol_new aic3x_left_hp_mixer_controls[] = {
SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPLOUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPLOUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLOUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPLOUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPLOUT_VOL, 7, 1, 0),
};
/* Right HP Mixer */
static const struct snd_kcontrol_new aic3x_right_hp_mixer_controls[] = {
SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPROUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPROUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPROUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPROUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPROUT_VOL, 7, 1, 0),
};
/* Left HPCOM Mixer */
static const struct snd_kcontrol_new aic3x_left_hpcom_mixer_controls[] = {
SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPLCOM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPLCOM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLCOM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPLCOM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPLCOM_VOL, 7, 1, 0),
};
/* Right HPCOM Mixer */
static const struct snd_kcontrol_new aic3x_right_hpcom_mixer_controls[] = {
SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPRCOM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("PGAL Bypass Switch", PGAL_2_HPRCOM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACL1 Switch", DACL1_2_HPRCOM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPRCOM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("PGAR Bypass Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0),
};
/* Left PGA Mixer */
static const struct snd_kcontrol_new aic3x_left_pga_mixer_controls[] = {
SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_LADC_CTRL, 3, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_LADC_CTRL, 3, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Line2L Switch", LINE2L_2_LADC_CTRL, 3, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_LADC_CTRL, 0, 1, 1),
};
/* Right PGA Mixer */
static const struct snd_kcontrol_new aic3x_right_pga_mixer_controls[] = {
SOC_DAPM_SINGLE_AIC3X("Line1R Switch", LINE1R_2_RADC_CTRL, 3, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Line1L Switch", LINE1L_2_RADC_CTRL, 3, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Line2R Switch", LINE2R_2_RADC_CTRL, 3, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Mic3L Switch", MIC3LR_2_RADC_CTRL, 4, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Mic3R Switch", MIC3LR_2_RADC_CTRL, 0, 1, 1),
};
/* Left Line1 Mux */
static const struct snd_kcontrol_new aic3x_left_line1l_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_L_ENUM]);
static const struct snd_kcontrol_new aic3x_right_line1l_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_enum[LINE1L_2_R_ENUM]);
/* Right Line1 Mux */
static const struct snd_kcontrol_new aic3x_right_line1r_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_R_ENUM]);
static const struct snd_kcontrol_new aic3x_left_line1r_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_enum[LINE1R_2_L_ENUM]);
/* Left Line2 Mux */
static const struct snd_kcontrol_new aic3x_left_line2_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_enum[LINE2L_ENUM]);
/* Right Line2 Mux */
static const struct snd_kcontrol_new aic3x_right_line2_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_enum[LINE2R_ENUM]);
static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
/* Left DAC to Left Outputs */
SND_SOC_DAPM_DAC("Left DAC", "Left Playback", DAC_PWR, 7, 0),
SND_SOC_DAPM_MUX("Left DAC Mux", SND_SOC_NOPM, 0, 0,
&aic3x_left_dac_mux_controls),
SND_SOC_DAPM_MUX("Left HPCOM Mux", SND_SOC_NOPM, 0, 0,
&aic3x_left_hpcom_mux_controls),
SND_SOC_DAPM_PGA("Left Line Out", LLOPM_CTRL, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Left HP Out", HPLOUT_CTRL, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Left HP Com", HPLCOM_CTRL, 0, 0, NULL, 0),
/* Right DAC to Right Outputs */
SND_SOC_DAPM_DAC("Right DAC", "Right Playback", DAC_PWR, 6, 0),
SND_SOC_DAPM_MUX("Right DAC Mux", SND_SOC_NOPM, 0, 0,
&aic3x_right_dac_mux_controls),
SND_SOC_DAPM_MUX("Right HPCOM Mux", SND_SOC_NOPM, 0, 0,
&aic3x_right_hpcom_mux_controls),
SND_SOC_DAPM_PGA("Right Line Out", RLOPM_CTRL, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right HP Out", HPROUT_CTRL, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right HP Com", HPRCOM_CTRL, 0, 0, NULL, 0),
/* Mono Output */
SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0),
/* Inputs to Left ADC */
SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0),
SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0,
&aic3x_left_pga_mixer_controls[0],
ARRAY_SIZE(aic3x_left_pga_mixer_controls)),
SND_SOC_DAPM_MUX("Left Line1L Mux", SND_SOC_NOPM, 0, 0,
&aic3x_left_line1l_mux_controls),
SND_SOC_DAPM_MUX("Left Line1R Mux", SND_SOC_NOPM, 0, 0,
&aic3x_left_line1r_mux_controls),
SND_SOC_DAPM_MUX("Left Line2L Mux", SND_SOC_NOPM, 0, 0,
&aic3x_left_line2_mux_controls),
/* Inputs to Right ADC */
SND_SOC_DAPM_ADC("Right ADC", "Right Capture",
LINE1R_2_RADC_CTRL, 2, 0),
SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0,
&aic3x_right_pga_mixer_controls[0],
ARRAY_SIZE(aic3x_right_pga_mixer_controls)),
SND_SOC_DAPM_MUX("Right Line1L Mux", SND_SOC_NOPM, 0, 0,
&aic3x_right_line1l_mux_controls),
SND_SOC_DAPM_MUX("Right Line1R Mux", SND_SOC_NOPM, 0, 0,
&aic3x_right_line1r_mux_controls),
SND_SOC_DAPM_MUX("Right Line2R Mux", SND_SOC_NOPM, 0, 0,
&aic3x_right_line2_mux_controls),
/*
* Not a real mic bias widget but similar function. This is for dynamic
* control of GPIO1 digital mic modulator clock output function when
* using digital mic.
*/
SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "GPIO1 dmic modclk",
AIC3X_GPIO1_REG, 4, 0xf,
AIC3X_GPIO1_FUNC_DIGITAL_MIC_MODCLK,
AIC3X_GPIO1_FUNC_DISABLED),
/*
* Also similar function like mic bias. Selects digital mic with
* configurable oversampling rate instead of ADC converter.
*/
SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 128",
AIC3X_ASD_INTF_CTRLA, 0, 3, 1, 0),
SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 64",
AIC3X_ASD_INTF_CTRLA, 0, 3, 2, 0),
SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "DMic Rate 32",
AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0),
/* Mic Bias */
SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0,
mic_bias_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
/* Output mixers */
SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0,
&aic3x_left_line_mixer_controls[0],
ARRAY_SIZE(aic3x_left_line_mixer_controls)),
SND_SOC_DAPM_MIXER("Right Line Mixer", SND_SOC_NOPM, 0, 0,
&aic3x_right_line_mixer_controls[0],
ARRAY_SIZE(aic3x_right_line_mixer_controls)),
SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0,
&aic3x_mono_mixer_controls[0],
ARRAY_SIZE(aic3x_mono_mixer_controls)),
SND_SOC_DAPM_MIXER("Left HP Mixer", SND_SOC_NOPM, 0, 0,
&aic3x_left_hp_mixer_controls[0],
ARRAY_SIZE(aic3x_left_hp_mixer_controls)),
SND_SOC_DAPM_MIXER("Right HP Mixer", SND_SOC_NOPM, 0, 0,
&aic3x_right_hp_mixer_controls[0],
ARRAY_SIZE(aic3x_right_hp_mixer_controls)),
SND_SOC_DAPM_MIXER("Left HPCOM Mixer", SND_SOC_NOPM, 0, 0,
&aic3x_left_hpcom_mixer_controls[0],
ARRAY_SIZE(aic3x_left_hpcom_mixer_controls)),
SND_SOC_DAPM_MIXER("Right HPCOM Mixer", SND_SOC_NOPM, 0, 0,
&aic3x_right_hpcom_mixer_controls[0],
ARRAY_SIZE(aic3x_right_hpcom_mixer_controls)),
SND_SOC_DAPM_OUTPUT("LLOUT"),
SND_SOC_DAPM_OUTPUT("RLOUT"),
SND_SOC_DAPM_OUTPUT("MONO_LOUT"),
SND_SOC_DAPM_OUTPUT("HPLOUT"),
SND_SOC_DAPM_OUTPUT("HPROUT"),
SND_SOC_DAPM_OUTPUT("HPLCOM"),
SND_SOC_DAPM_OUTPUT("HPRCOM"),
SND_SOC_DAPM_INPUT("MIC3L"),
SND_SOC_DAPM_INPUT("MIC3R"),
SND_SOC_DAPM_INPUT("LINE1L"),
SND_SOC_DAPM_INPUT("LINE1R"),
SND_SOC_DAPM_INPUT("LINE2L"),
SND_SOC_DAPM_INPUT("LINE2R"),
/*
* Virtual output pin to detection block inside codec. This can be
* used to keep codec bias on if gpio or detection features are needed.
* Force pin on or construct a path with an input jack and mic bias
* widgets.
*/
SND_SOC_DAPM_OUTPUT("Detection"),
};
static const struct snd_soc_dapm_widget aic3007_dapm_widgets[] = {
/* Class-D outputs */
SND_SOC_DAPM_PGA("Left Class-D Out", CLASSD_CTRL, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Class-D Out", CLASSD_CTRL, 2, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("SPOP"),
SND_SOC_DAPM_OUTPUT("SPOM"),
};
static const struct snd_soc_dapm_route intercon[] = {
/* Left Input */
{"Left Line1L Mux", "single-ended", "LINE1L"},
{"Left Line1L Mux", "differential", "LINE1L"},
{"Left Line2L Mux", "single-ended", "LINE2L"},
{"Left Line2L Mux", "differential", "LINE2L"},
{"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"},
{"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"},
{"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"},
{"Left PGA Mixer", "Mic3L Switch", "MIC3L"},
{"Left PGA Mixer", "Mic3R Switch", "MIC3R"},
{"Left ADC", NULL, "Left PGA Mixer"},
{"Left ADC", NULL, "GPIO1 dmic modclk"},
/* Right Input */
{"Right Line1R Mux", "single-ended", "LINE1R"},
{"Right Line1R Mux", "differential", "LINE1R"},
{"Right Line2R Mux", "single-ended", "LINE2R"},
{"Right Line2R Mux", "differential", "LINE2R"},
{"Right PGA Mixer", "Line1L Switch", "Right Line1L Mux"},
{"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"},
{"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"},
{"Right PGA Mixer", "Mic3L Switch", "MIC3L"},
{"Right PGA Mixer", "Mic3R Switch", "MIC3R"},
{"Right ADC", NULL, "Right PGA Mixer"},
{"Right ADC", NULL, "GPIO1 dmic modclk"},
/*
* Logical path between digital mic enable and GPIO1 modulator clock
* output function
*/
{"GPIO1 dmic modclk", NULL, "DMic Rate 128"},
{"GPIO1 dmic modclk", NULL, "DMic Rate 64"},
{"GPIO1 dmic modclk", NULL, "DMic Rate 32"},
/* Left DAC Output */
{"Left DAC Mux", "DAC_L1", "Left DAC"},
{"Left DAC Mux", "DAC_L2", "Left DAC"},
{"Left DAC Mux", "DAC_L3", "Left DAC"},
/* Right DAC Output */
{"Right DAC Mux", "DAC_R1", "Right DAC"},
{"Right DAC Mux", "DAC_R2", "Right DAC"},
{"Right DAC Mux", "DAC_R3", "Right DAC"},
/* Left Line Output */
{"Left Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
{"Left Line Mixer", "PGAL Bypass Switch", "Left PGA Mixer"},
{"Left Line Mixer", "DACL1 Switch", "Left DAC Mux"},
{"Left Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
{"Left Line Mixer", "PGAR Bypass Switch", "Right PGA Mixer"},
{"Left Line Mixer", "DACR1 Switch", "Right DAC Mux"},
{"Left Line Out", NULL, "Left Line Mixer"},
{"Left Line Out", NULL, "Left DAC Mux"},
{"LLOUT", NULL, "Left Line Out"},
/* Right Line Output */
{"Right Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
{"Right Line Mixer", "PGAL Bypass Switch", "Left PGA Mixer"},
{"Right Line Mixer", "DACL1 Switch", "Left DAC Mux"},
{"Right Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
{"Right Line Mixer", "PGAR Bypass Switch", "Right PGA Mixer"},
{"Right Line Mixer", "DACR1 Switch", "Right DAC Mux"},
{"Right Line Out", NULL, "Right Line Mixer"},
{"Right Line Out", NULL, "Right DAC Mux"},
{"RLOUT", NULL, "Right Line Out"},
/* Mono Output */
{"Mono Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
{"Mono Mixer", "PGAL Bypass Switch", "Left PGA Mixer"},
{"Mono Mixer", "DACL1 Switch", "Left DAC Mux"},
{"Mono Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
{"Mono Mixer", "PGAR Bypass Switch", "Right PGA Mixer"},
{"Mono Mixer", "DACR1 Switch", "Right DAC Mux"},
{"Mono Out", NULL, "Mono Mixer"},
{"MONO_LOUT", NULL, "Mono Out"},
/* Left HP Output */
{"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
{"Left HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"},
{"Left HP Mixer", "DACL1 Switch", "Left DAC Mux"},
{"Left HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
{"Left HP Mixer", "PGAR Bypass Switch", "Right PGA Mixer"},
{"Left HP Mixer", "DACR1 Switch", "Right DAC Mux"},
{"Left HP Out", NULL, "Left HP Mixer"},
{"Left HP Out", NULL, "Left DAC Mux"},
{"HPLOUT", NULL, "Left HP Out"},
/* Right HP Output */
{"Right HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
{"Right HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"},
{"Right HP Mixer", "DACL1 Switch", "Left DAC Mux"},
{"Right HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
{"Right HP Mixer", "PGAR Bypass Switch", "Right PGA Mixer"},
{"Right HP Mixer", "DACR1 Switch", "Right DAC Mux"},
{"Right HP Out", NULL, "Right HP Mixer"},
{"Right HP Out", NULL, "Right DAC Mux"},
{"HPROUT", NULL, "Right HP Out"},
/* Left HPCOM Output */
{"Left HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
{"Left HPCOM Mixer", "PGAL Bypass Switch", "Left PGA Mixer"},
{"Left HPCOM Mixer", "DACL1 Switch", "Left DAC Mux"},
{"Left HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
{"Left HPCOM Mixer", "PGAR Bypass Switch", "Right PGA Mixer"},
{"Left HPCOM Mixer", "DACR1 Switch", "Right DAC Mux"},
{"Left HPCOM Mux", "differential of HPLOUT", "Left HP Mixer"},
{"Left HPCOM Mux", "constant VCM", "Left HPCOM Mixer"},
{"Left HPCOM Mux", "single-ended", "Left HPCOM Mixer"},
{"Left HP Com", NULL, "Left HPCOM Mux"},
{"HPLCOM", NULL, "Left HP Com"},
/* Right HPCOM Output */
{"Right HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
{"Right HPCOM Mixer", "PGAL Bypass Switch", "Left PGA Mixer"},
{"Right HPCOM Mixer", "DACL1 Switch", "Left DAC Mux"},
{"Right HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
{"Right HPCOM Mixer", "PGAR Bypass Switch", "Right PGA Mixer"},
{"Right HPCOM Mixer", "DACR1 Switch", "Right DAC Mux"},
{"Right HPCOM Mux", "differential of HPROUT", "Right HP Mixer"},
{"Right HPCOM Mux", "constant VCM", "Right HPCOM Mixer"},
{"Right HPCOM Mux", "single-ended", "Right HPCOM Mixer"},
{"Right HPCOM Mux", "differential of HPLCOM", "Left HPCOM Mixer"},
{"Right HPCOM Mux", "external feedback", "Right HPCOM Mixer"},
{"Right HP Com", NULL, "Right HPCOM Mux"},
{"HPRCOM", NULL, "Right HP Com"},
};
static const struct snd_soc_dapm_route intercon_3007[] = {
/* Class-D outputs */
{"Left Class-D Out", NULL, "Left Line Out"},
{"Right Class-D Out", NULL, "Left Line Out"},
{"SPOP", NULL, "Left Class-D Out"},
{"SPOM", NULL, "Right Class-D Out"},
};
static int aic3x_add_widgets(struct snd_soc_codec *codec)
{
struct aic3x_priv *aic3x = 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 20:53:46 +07:00
struct snd_soc_dapm_context *dapm = &codec->dapm;
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 20:53:46 +07:00
snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets,
ARRAY_SIZE(aic3x_dapm_widgets));
/* set up audio path interconnects */
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 20:53:46 +07:00
snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
if (aic3x->model == AIC3X_MODEL_3007) {
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 20:53:46 +07:00
snd_soc_dapm_new_controls(dapm, aic3007_dapm_widgets,
ARRAY_SIZE(aic3007_dapm_widgets));
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 20:53:46 +07:00
snd_soc_dapm_add_routes(dapm, intercon_3007,
ARRAY_SIZE(intercon_3007));
}
return 0;
}
static int aic3x_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 aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
u16 d, pll_d = 1;
int clk;
/* select data word length */
data = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
data |= (0x01 << 4);
break;
case SNDRV_PCM_FORMAT_S24_LE:
data |= (0x02 << 4);
break;
case SNDRV_PCM_FORMAT_S32_LE:
data |= (0x03 << 4);
break;
}
snd_soc_write(codec, AIC3X_ASD_INTF_CTRLB, data);
/* Fsref can be 44100 or 48000 */
fsref = (params_rate(params) % 11025 == 0) ? 44100 : 48000;
/* Try to find a value for Q which allows us to bypass the PLL and
* generate CODEC_CLK directly. */
for (pll_q = 2; pll_q < 18; pll_q++)
if (aic3x->sysclk / (128 * pll_q) == fsref) {
bypass_pll = 1;
break;
}
if (bypass_pll) {
pll_q &= 0xf;
snd_soc_write(codec, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT);
snd_soc_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV);
/* disable PLL if it is bypassed */
snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLL_ENABLE, 0);
} else {
snd_soc_write(codec, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV);
/* enable PLL when it is used */
snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG,
PLL_ENABLE, PLL_ENABLE);
}
/* Route Left DAC to left channel input and
* right DAC to right channel input */
data = (LDAC2LCH | RDAC2RCH);
data |= (fsref == 44100) ? FSREF_44100 : FSREF_48000;
if (params_rate(params) >= 64000)
data |= DUAL_RATE_MODE;
snd_soc_write(codec, AIC3X_CODEC_DATAPATH_REG, data);
/* codec sample rate select */
data = (fsref * 20) / params_rate(params);
if (params_rate(params) < 64000)
data /= 2;
data /= 5;
data -= 2;
data |= (data << 4);
snd_soc_write(codec, AIC3X_SAMPLE_RATE_SEL_REG, data);
if (bypass_pll)
return 0;
/* Use PLL, compute appropriate setup for j, d, r and p, the closest
* one wins the game. Try with d==0 first, next with d!=0.
* Constraints for j are according to the datasheet.
* The sysclk is divided by 1000 to prevent integer overflows.
*/
codec_clk = (2048 * fsref) / (aic3x->sysclk / 1000);
for (r = 1; r <= 16; r++)
for (p = 1; p <= 8; p++) {
for (j = 4; j <= 55; j++) {
/* This is actually 1000*((j+(d/10000))*r)/p
* The term had to be converted to get
* rid of the division by 10000; d = 0 here
*/
int tmp_clk = (1000 * j * r) / p;
/* Check whether this values get closer than
* the best ones we had before
*/
if (abs(codec_clk - tmp_clk) <
abs(codec_clk - last_clk)) {
pll_j = j; pll_d = 0;
pll_r = r; pll_p = p;
last_clk = tmp_clk;
}
/* Early exit for exact matches */
if (tmp_clk == codec_clk)
goto found;
}
}
/* try with d != 0 */
for (p = 1; p <= 8; p++) {
j = codec_clk * p / 1000;
if (j < 4 || j > 11)
continue;
/* do not use codec_clk here since we'd loose precision */
d = ((2048 * p * fsref) - j * aic3x->sysclk)
* 100 / (aic3x->sysclk/100);
clk = (10000 * j + d) / (10 * p);
/* check whether this values get closer than the best
* ones we had before */
if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) {
pll_j = j; pll_d = d; pll_r = 1; pll_p = p;
last_clk = clk;
}
/* Early exit for exact matches */
if (clk == codec_clk)
goto found;
}
if (last_clk == 0) {
printk(KERN_ERR "%s(): unable to setup PLL\n", __func__);
return -EINVAL;
}
found:
snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p);
snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG,
pll_r << PLLR_SHIFT);
snd_soc_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT);
snd_soc_write(codec, AIC3X_PLL_PROGC_REG,
(pll_d >> 6) << PLLD_MSB_SHIFT);
snd_soc_write(codec, AIC3X_PLL_PROGD_REG,
(pll_d & 0x3F) << PLLD_LSB_SHIFT);
return 0;
}
static int aic3x_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u8 ldac_reg = snd_soc_read(codec, LDAC_VOL) & ~MUTE_ON;
u8 rdac_reg = snd_soc_read(codec, RDAC_VOL) & ~MUTE_ON;
if (mute) {
snd_soc_write(codec, LDAC_VOL, ldac_reg | MUTE_ON);
snd_soc_write(codec, RDAC_VOL, rdac_reg | MUTE_ON);
} else {
snd_soc_write(codec, LDAC_VOL, ldac_reg);
snd_soc_write(codec, RDAC_VOL, rdac_reg);
}
return 0;
}
static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
/* set clock on MCLK or GPIO2 or BCLK */
snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, PLLCLK_IN_MASK,
clk_id << PLLCLK_IN_SHIFT);
snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, CLKDIV_IN_MASK,
clk_id << CLKDIV_IN_SHIFT);
aic3x->sysclk = freq;
return 0;
}
static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
u8 iface_areg, iface_breg;
int delay = 0;
iface_areg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLA) & 0x3f;
iface_breg = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & 0x3f;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
aic3x->master = 1;
iface_areg |= BIT_CLK_MASTER | WORD_CLK_MASTER;
break;
case SND_SOC_DAIFMT_CBS_CFS:
aic3x->master = 0;
iface_areg &= ~(BIT_CLK_MASTER | WORD_CLK_MASTER);
break;
default:
return -EINVAL;
}
/*
* match both interface format and signal polarities since they
* are fixed
*/
switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK |
SND_SOC_DAIFMT_INV_MASK)) {
case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
break;
case (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF):
delay = 1;
case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
iface_breg |= (0x01 << 6);
break;
case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF):
iface_breg |= (0x02 << 6);
break;
case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF):
iface_breg |= (0x03 << 6);
break;
default:
return -EINVAL;
}
/* set iface */
snd_soc_write(codec, AIC3X_ASD_INTF_CTRLA, iface_areg);
snd_soc_write(codec, AIC3X_ASD_INTF_CTRLB, iface_breg);
snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay);
return 0;
}
static int aic3x_init_3007(struct snd_soc_codec *codec)
{
u8 tmp1, tmp2, *cache = codec->reg_cache;
/*
* There is no need to cache writes to undocumented page 0xD but
* respective page 0 register cache entries must be preserved
*/
tmp1 = cache[0xD];
tmp2 = cache[0x8];
/* Class-D speaker driver init; datasheet p. 46 */
snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x0D);
snd_soc_write(codec, 0xD, 0x0D);
snd_soc_write(codec, 0x8, 0x5C);
snd_soc_write(codec, 0x8, 0x5D);
snd_soc_write(codec, 0x8, 0x5C);
snd_soc_write(codec, AIC3X_PAGE_SELECT, 0x00);
cache[0xD] = tmp1;
cache[0x8] = tmp2;
return 0;
}
static int aic3x_regulator_event(struct notifier_block *nb,
unsigned long event, void *data)
{
struct aic3x_disable_nb *disable_nb =
container_of(nb, struct aic3x_disable_nb, nb);
struct aic3x_priv *aic3x = disable_nb->aic3x;
if (event & REGULATOR_EVENT_DISABLE) {
/*
* Put codec to reset and require cache sync as at least one
* of the supplies was disabled
*/
if (gpio_is_valid(aic3x->gpio_reset))
gpio_set_value(aic3x->gpio_reset, 0);
aic3x->codec->cache_sync = 1;
}
return 0;
}
static int aic3x_set_power(struct snd_soc_codec *codec, int power)
{
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
int i, ret;
u8 *cache = codec->reg_cache;
if (power) {
ret = regulator_bulk_enable(ARRAY_SIZE(aic3x->supplies),
aic3x->supplies);
if (ret)
goto out;
aic3x->power = 1;
/*
* Reset release and cache sync is necessary only if some
* supply was off or if there were cached writes
*/
if (!codec->cache_sync)
goto out;
if (gpio_is_valid(aic3x->gpio_reset)) {
udelay(1);
gpio_set_value(aic3x->gpio_reset, 1);
}
/* Sync reg_cache with the hardware */
codec->cache_only = 0;
for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++)
snd_soc_write(codec, i, cache[i]);
if (aic3x->model == AIC3X_MODEL_3007)
aic3x_init_3007(codec);
codec->cache_sync = 0;
} else {
/*
* Do soft reset to this codec instance in order to clear
* possible VDD leakage currents in case the supply regulators
* remain on
*/
snd_soc_write(codec, AIC3X_RESET, SOFT_RESET);
codec->cache_sync = 1;
aic3x->power = 0;
/* HW writes are needless when bias is off */
codec->cache_only = 1;
ret = regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies),
aic3x->supplies);
}
out:
return ret;
}
static int aic3x_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
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 20:53:46 +07:00
if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY &&
aic3x->master) {
/* enable pll */
snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG,
PLL_ENABLE, PLL_ENABLE);
}
break;
case SND_SOC_BIAS_STANDBY:
if (!aic3x->power)
aic3x_set_power(codec, 1);
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 20:53:46 +07:00
if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE &&
aic3x->master) {
/* disable pll */
snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG,
PLL_ENABLE, 0);
}
break;
case SND_SOC_BIAS_OFF:
if (aic3x->power)
aic3x_set_power(codec, 0);
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 20:53:46 +07:00
codec->dapm.bias_level = level;
return 0;
}
#define AIC3X_RATES SNDRV_PCM_RATE_8000_96000
#define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
static const struct snd_soc_dai_ops aic3x_dai_ops = {
.hw_params = aic3x_hw_params,
.digital_mute = aic3x_mute,
.set_sysclk = aic3x_set_dai_sysclk,
.set_fmt = aic3x_set_dai_fmt,
};
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 03:15:21 +07:00
static struct snd_soc_dai_driver aic3x_dai = {
.name = "tlv320aic3x-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
.rates = AIC3X_RATES,
.formats = AIC3X_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = AIC3X_RATES,
.formats = AIC3X_FORMATS,},
.ops = &aic3x_dai_ops,
.symmetric_rates = 1,
};
static int aic3x_suspend(struct snd_soc_codec *codec)
{
aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
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 03:15:21 +07:00
static int aic3x_resume(struct snd_soc_codec *codec)
{
aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
}
/*
* initialise the AIC3X driver
* register the mixer and dsp interfaces with the kernel
*/
static int aic3x_init(struct snd_soc_codec *codec)
{
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
snd_soc_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT);
snd_soc_write(codec, AIC3X_RESET, SOFT_RESET);
/* DAC default volume and mute */
snd_soc_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON);
snd_soc_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON);
/* DAC to HP default volume and route to Output mixer */
snd_soc_write(codec, DACL1_2_HPLOUT_VOL, DEFAULT_VOL | ROUTE_ON);
snd_soc_write(codec, DACR1_2_HPROUT_VOL, DEFAULT_VOL | ROUTE_ON);
snd_soc_write(codec, DACL1_2_HPLCOM_VOL, DEFAULT_VOL | ROUTE_ON);
snd_soc_write(codec, DACR1_2_HPRCOM_VOL, DEFAULT_VOL | ROUTE_ON);
/* DAC to Line Out default volume and route to Output mixer */
snd_soc_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
snd_soc_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
/* DAC to Mono Line Out default volume and route to Output mixer */
snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
/* unmute all outputs */
snd_soc_update_bits(codec, LLOPM_CTRL, UNMUTE, UNMUTE);
snd_soc_update_bits(codec, RLOPM_CTRL, UNMUTE, UNMUTE);
snd_soc_update_bits(codec, MONOLOPM_CTRL, UNMUTE, UNMUTE);
snd_soc_update_bits(codec, HPLOUT_CTRL, UNMUTE, UNMUTE);
snd_soc_update_bits(codec, HPROUT_CTRL, UNMUTE, UNMUTE);
snd_soc_update_bits(codec, HPLCOM_CTRL, UNMUTE, UNMUTE);
snd_soc_update_bits(codec, HPRCOM_CTRL, UNMUTE, UNMUTE);
/* ADC default volume and unmute */
snd_soc_write(codec, LADC_VOL, DEFAULT_GAIN);
snd_soc_write(codec, RADC_VOL, DEFAULT_GAIN);
/* By default route Line1 to ADC PGA mixer */
snd_soc_write(codec, LINE1L_2_LADC_CTRL, 0x0);
snd_soc_write(codec, LINE1R_2_RADC_CTRL, 0x0);
/* PGA to HP Bypass default volume, disconnect from Output Mixer */
snd_soc_write(codec, PGAL_2_HPLOUT_VOL, DEFAULT_VOL);
snd_soc_write(codec, PGAR_2_HPROUT_VOL, DEFAULT_VOL);
snd_soc_write(codec, PGAL_2_HPLCOM_VOL, DEFAULT_VOL);
snd_soc_write(codec, PGAR_2_HPRCOM_VOL, DEFAULT_VOL);
/* PGA to Line Out default volume, disconnect from Output Mixer */
snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL);
snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL);
/* PGA to Mono Line Out default volume, disconnect from Output Mixer */
snd_soc_write(codec, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL);
snd_soc_write(codec, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL);
/* Line2 to HP Bypass default volume, disconnect from Output Mixer */
snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL);
snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL);
snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL);
snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL);
/* Line2 Line Out default volume, disconnect from Output Mixer */
snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL);
snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL);
/* Line2 to Mono Out default volume, disconnect from Output Mixer */
snd_soc_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL);
snd_soc_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL);
if (aic3x->model == AIC3X_MODEL_3007) {
aic3x_init_3007(codec);
snd_soc_write(codec, CLASSD_CTRL, 0);
}
return 0;
}
static bool aic3x_is_shared_reset(struct aic3x_priv *aic3x)
{
struct aic3x_priv *a;
list_for_each_entry(a, &reset_list, list) {
if (gpio_is_valid(aic3x->gpio_reset) &&
aic3x->gpio_reset == a->gpio_reset)
return true;
}
return false;
}
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 03:15:21 +07:00
static int aic3x_probe(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 03:15:21 +07:00
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
int ret, i;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
INIT_LIST_HEAD(&aic3x->list);
aic3x->codec = codec;
ret = snd_soc_codec_set_cache_io(codec, 8, 8, aic3x->control_type);
if (ret != 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
}
if (gpio_is_valid(aic3x->gpio_reset) &&
!aic3x_is_shared_reset(aic3x)) {
ret = gpio_request(aic3x->gpio_reset, "tlv320aic3x reset");
if (ret != 0)
goto err_gpio;
gpio_direction_output(aic3x->gpio_reset, 0);
}
for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++)
aic3x->supplies[i].supply = aic3x_supply_names[i];
ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(aic3x->supplies),
aic3x->supplies);
if (ret != 0) {
dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
goto err_get;
}
for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) {
aic3x->disable_nb[i].nb.notifier_call = aic3x_regulator_event;
aic3x->disable_nb[i].aic3x = aic3x;
ret = regulator_register_notifier(aic3x->supplies[i].consumer,
&aic3x->disable_nb[i].nb);
if (ret) {
dev_err(codec->dev,
"Failed to request regulator notifier: %d\n",
ret);
goto err_notif;
}
}
codec->cache_only = 1;
aic3x_init(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 03:15:21 +07:00
if (aic3x->setup) {
/* setup GPIO functions */
snd_soc_write(codec, AIC3X_GPIO1_REG,
(aic3x->setup->gpio_func[0] & 0xf) << 4);
snd_soc_write(codec, AIC3X_GPIO2_REG,
(aic3x->setup->gpio_func[1] & 0xf) << 4);
}
snd_soc_add_codec_controls(codec, aic3x_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 03:15:21 +07:00
ARRAY_SIZE(aic3x_snd_controls));
if (aic3x->model == AIC3X_MODEL_3007)
snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1);
/* set mic bias voltage */
switch (aic3x->micbias_vg) {
case AIC3X_MICBIAS_2_0V:
case AIC3X_MICBIAS_2_5V:
case AIC3X_MICBIAS_AVDDV:
snd_soc_update_bits(codec, MICBIAS_CTRL,
MICBIAS_LEVEL_MASK,
(aic3x->micbias_vg) << MICBIAS_LEVEL_SHIFT);
break;
case AIC3X_MICBIAS_OFF:
/*
* noting to do. target won't enter here. This is just to avoid
* compile time warning "warning: enumeration value
* 'AIC3X_MICBIAS_OFF' not handled in switch"
*/
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 03:15:21 +07:00
aic3x_add_widgets(codec);
list_add(&aic3x->list, &reset_list);
return 0;
err_notif:
while (i--)
regulator_unregister_notifier(aic3x->supplies[i].consumer,
&aic3x->disable_nb[i].nb);
regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies);
err_get:
if (gpio_is_valid(aic3x->gpio_reset) &&
!aic3x_is_shared_reset(aic3x))
gpio_free(aic3x->gpio_reset);
err_gpio:
return ret;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
static int aic3x_remove(struct snd_soc_codec *codec)
{
struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
int i;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
list_del(&aic3x->list);
if (gpio_is_valid(aic3x->gpio_reset) &&
!aic3x_is_shared_reset(aic3x)) {
gpio_set_value(aic3x->gpio_reset, 0);
gpio_free(aic3x->gpio_reset);
}
for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++)
regulator_unregister_notifier(aic3x->supplies[i].consumer,
&aic3x->disable_nb[i].nb);
regulator_bulk_free(ARRAY_SIZE(aic3x->supplies), aic3x->supplies);
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 03:15:21 +07:00
static struct snd_soc_codec_driver soc_codec_dev_aic3x = {
.set_bias_level = aic3x_set_bias_level,
.idle_bias_off = true,
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 03:15:21 +07:00
.reg_cache_size = ARRAY_SIZE(aic3x_reg),
.reg_word_size = sizeof(u8),
.reg_cache_default = aic3x_reg,
.probe = aic3x_probe,
.remove = aic3x_remove,
.suspend = aic3x_suspend,
.resume = aic3x_resume,
};
/*
* AIC3X 2 wire address can be up to 4 devices with device addresses
* 0x18, 0x19, 0x1A, 0x1B
*/
static const struct i2c_device_id aic3x_i2c_id[] = {
{ "tlv320aic3x", AIC3X_MODEL_3X },
{ "tlv320aic33", AIC3X_MODEL_33 },
{ "tlv320aic3007", AIC3X_MODEL_3007 },
{ }
};
MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
/*
* If the i2c layer weren't so broken, we could pass this kind of data
* around
*/
static int aic3x_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct aic3x_pdata *pdata = i2c->dev.platform_data;
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 03:15:21 +07:00
struct aic3x_priv *aic3x;
struct aic3x_setup_data *ai3x_setup;
struct device_node *np = i2c->dev.of_node;
int ret;
u32 value;
aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL);
if (aic3x == NULL) {
dev_err(&i2c->dev, "failed to create private data\n");
return -ENOMEM;
}
aic3x->control_type = SND_SOC_I2C;
i2c_set_clientdata(i2c, aic3x);
if (pdata) {
aic3x->gpio_reset = pdata->gpio_reset;
aic3x->setup = pdata->setup;
aic3x->micbias_vg = pdata->micbias_vg;
} else if (np) {
ai3x_setup = devm_kzalloc(&i2c->dev, sizeof(*ai3x_setup),
GFP_KERNEL);
if (ai3x_setup == NULL) {
dev_err(&i2c->dev, "failed to create private data\n");
return -ENOMEM;
}
ret = of_get_named_gpio(np, "gpio-reset", 0);
if (ret >= 0)
aic3x->gpio_reset = ret;
else
aic3x->gpio_reset = -1;
if (of_property_read_u32_array(np, "ai3x-gpio-func",
ai3x_setup->gpio_func, 2) >= 0) {
aic3x->setup = ai3x_setup;
}
if (!of_property_read_u32(np, "ai3x-micbias-vg", &value)) {
switch (value) {
case 1 :
aic3x->micbias_vg = AIC3X_MICBIAS_2_0V;
break;
case 2 :
aic3x->micbias_vg = AIC3X_MICBIAS_2_5V;
break;
case 3 :
aic3x->micbias_vg = AIC3X_MICBIAS_AVDDV;
break;
default :
aic3x->micbias_vg = AIC3X_MICBIAS_OFF;
dev_err(&i2c->dev, "Unsuitable MicBias voltage "
"found in DT\n");
}
} else {
aic3x->micbias_vg = AIC3X_MICBIAS_OFF;
}
} else {
aic3x->gpio_reset = -1;
}
aic3x->model = id->driver_data;
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 03:15:21 +07:00
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_aic3x, &aic3x_dai, 1);
return ret;
}
static int aic3x_i2c_remove(struct i2c_client *client)
{
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 03:15:21 +07:00
snd_soc_unregister_codec(&client->dev);
return 0;
}
#if defined(CONFIG_OF)
static const struct of_device_id tlv320aic3x_of_match[] = {
{ .compatible = "ti,tlv320aic3x", },
{},
};
MODULE_DEVICE_TABLE(of, tlv320aic3x_of_match);
#endif
/* machine i2c codec control layer */
static struct i2c_driver aic3x_i2c_driver = {
.driver = {
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 03:15:21 +07:00
.name = "tlv320aic3x-codec",
.owner = THIS_MODULE,
.of_match_table = of_match_ptr(tlv320aic3x_of_match),
},
.probe = aic3x_i2c_probe,
.remove = aic3x_i2c_remove,
.id_table = aic3x_i2c_id,
};
module_i2c_driver(aic3x_i2c_driver);
MODULE_DESCRIPTION("ASoC TLV320AIC3X codec driver");
MODULE_AUTHOR("Vladimir Barinov");
MODULE_LICENSE("GPL");