alistair23-linux/sound/soc/codecs/tlv320aic3x.c

1939 lines
64 KiB
C
Raw Normal View History

// SPDX-License-Identifier: GPL-2.0-only
/*
* 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
*
* 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.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 02:04:11 -06:00
#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_component *component;
struct regmap *regmap;
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>
2010-03-17 14:15:21 -06:00
struct aic3x_setup_data *setup;
unsigned int sysclk;
unsigned int dai_fmt;
unsigned int tdm_delay;
unsigned int slot_width;
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
#define AIC3X_MODEL_3104 3
u16 model;
/* Selects the micbias voltage */
enum aic3x_micbias_voltage micbias_vg;
/* Output Common-Mode Voltage */
u8 ocmv;
};
static const struct reg_default aic3x_reg[] = {
{ 0, 0x00 }, { 1, 0x00 }, { 2, 0x00 }, { 3, 0x10 },
{ 4, 0x04 }, { 5, 0x00 }, { 6, 0x00 }, { 7, 0x00 },
{ 8, 0x00 }, { 9, 0x00 }, { 10, 0x00 }, { 11, 0x01 },
{ 12, 0x00 }, { 13, 0x00 }, { 14, 0x00 }, { 15, 0x80 },
{ 16, 0x80 }, { 17, 0xff }, { 18, 0xff }, { 19, 0x78 },
{ 20, 0x78 }, { 21, 0x78 }, { 22, 0x78 }, { 23, 0x78 },
{ 24, 0x78 }, { 25, 0x00 }, { 26, 0x00 }, { 27, 0xfe },
{ 28, 0x00 }, { 29, 0x00 }, { 30, 0xfe }, { 31, 0x00 },
{ 32, 0x18 }, { 33, 0x18 }, { 34, 0x00 }, { 35, 0x00 },
{ 36, 0x00 }, { 37, 0x00 }, { 38, 0x00 }, { 39, 0x00 },
{ 40, 0x00 }, { 41, 0x00 }, { 42, 0x00 }, { 43, 0x80 },
{ 44, 0x80 }, { 45, 0x00 }, { 46, 0x00 }, { 47, 0x00 },
{ 48, 0x00 }, { 49, 0x00 }, { 50, 0x00 }, { 51, 0x04 },
{ 52, 0x00 }, { 53, 0x00 }, { 54, 0x00 }, { 55, 0x00 },
{ 56, 0x00 }, { 57, 0x00 }, { 58, 0x04 }, { 59, 0x00 },
{ 60, 0x00 }, { 61, 0x00 }, { 62, 0x00 }, { 63, 0x00 },
{ 64, 0x00 }, { 65, 0x04 }, { 66, 0x00 }, { 67, 0x00 },
{ 68, 0x00 }, { 69, 0x00 }, { 70, 0x00 }, { 71, 0x00 },
{ 72, 0x04 }, { 73, 0x00 }, { 74, 0x00 }, { 75, 0x00 },
{ 76, 0x00 }, { 77, 0x00 }, { 78, 0x00 }, { 79, 0x00 },
{ 80, 0x00 }, { 81, 0x00 }, { 82, 0x00 }, { 83, 0x00 },
{ 84, 0x00 }, { 85, 0x00 }, { 86, 0x00 }, { 87, 0x00 },
{ 88, 0x00 }, { 89, 0x00 }, { 90, 0x00 }, { 91, 0x00 },
{ 92, 0x00 }, { 93, 0x00 }, { 94, 0x00 }, { 95, 0x00 },
{ 96, 0x00 }, { 97, 0x00 }, { 98, 0x00 }, { 99, 0x00 },
{ 100, 0x00 }, { 101, 0x00 }, { 102, 0x02 }, { 103, 0x00 },
{ 104, 0x00 }, { 105, 0x00 }, { 106, 0x00 }, { 107, 0x00 },
{ 108, 0x00 }, { 109, 0x00 },
};
static bool aic3x_volatile_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
case AIC3X_RESET:
return true;
default:
return false;
}
}
static const struct regmap_config aic3x_regmap = {
.reg_bits = 8,
.val_bits = 8,
.max_register = DAC_ICC_ADJ,
.reg_defaults = aic3x_reg,
.num_reg_defaults = ARRAY_SIZE(aic3x_reg),
.volatile_reg = aic3x_volatile_reg,
.cache_type = REGCACHE_RBTREE,
};
#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_component *component = snd_soc_dapm_kcontrol_component(kcontrol);
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
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;
struct snd_soc_dapm_update update = {};
int connect, change;
val = (ucontrol->value.integer.value[0] & mask);
mask = 0xf;
if (val)
val = mask;
connect = !!val;
if (invert)
val = mask - val;
mask <<= shift;
val <<= shift;
change = snd_soc_component_test_bits(component, reg, mask, val);
if (change) {
update.kcontrol = kcontrol;
update.reg = reg;
update.mask = mask;
update.val = val;
snd_soc_dapm_mixer_update_power(dapm, kcontrol, connect,
&update);
}
return change;
}
/*
* 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_component *component = snd_soc_dapm_to_component(w->dapm);
struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
/* change mic bias voltage to user defined */
snd_soc_component_update_bits(component, MICBIAS_CTRL,
MICBIAS_LEVEL_MASK,
aic3x->micbias_vg << MICBIAS_LEVEL_SHIFT);
break;
case SND_SOC_DAPM_PRE_PMD:
snd_soc_component_update_bits(component, MICBIAS_CTRL,
MICBIAS_LEVEL_MASK, 0);
break;
}
return 0;
}
static const char * const aic3x_left_dac_mux[] = {
"DAC_L1", "DAC_L3", "DAC_L2" };
static SOC_ENUM_SINGLE_DECL(aic3x_left_dac_enum, DAC_LINE_MUX, 6,
aic3x_left_dac_mux);
static const char * const aic3x_right_dac_mux[] = {
"DAC_R1", "DAC_R3", "DAC_R2" };
static SOC_ENUM_SINGLE_DECL(aic3x_right_dac_enum, DAC_LINE_MUX, 4,
aic3x_right_dac_mux);
static const char * const aic3x_left_hpcom_mux[] = {
"differential of HPLOUT", "constant VCM", "single-ended" };
static SOC_ENUM_SINGLE_DECL(aic3x_left_hpcom_enum, HPLCOM_CFG, 4,
aic3x_left_hpcom_mux);
static const char * const aic3x_right_hpcom_mux[] = {
"differential of HPROUT", "constant VCM", "single-ended",
"differential of HPLCOM", "external feedback" };
static SOC_ENUM_SINGLE_DECL(aic3x_right_hpcom_enum, HPRCOM_CFG, 3,
aic3x_right_hpcom_mux);
static const char * const aic3x_linein_mode_mux[] = {
"single-ended", "differential" };
static SOC_ENUM_SINGLE_DECL(aic3x_line1l_2_l_enum, LINE1L_2_LADC_CTRL, 7,
aic3x_linein_mode_mux);
static SOC_ENUM_SINGLE_DECL(aic3x_line1l_2_r_enum, LINE1L_2_RADC_CTRL, 7,
aic3x_linein_mode_mux);
static SOC_ENUM_SINGLE_DECL(aic3x_line1r_2_l_enum, LINE1R_2_LADC_CTRL, 7,
aic3x_linein_mode_mux);
static SOC_ENUM_SINGLE_DECL(aic3x_line1r_2_r_enum, LINE1R_2_RADC_CTRL, 7,
aic3x_linein_mode_mux);
static SOC_ENUM_SINGLE_DECL(aic3x_line2l_2_ldac_enum, LINE2L_2_LADC_CTRL, 7,
aic3x_linein_mode_mux);
static SOC_ENUM_SINGLE_DECL(aic3x_line2r_2_rdac_enum, LINE2R_2_RADC_CTRL, 7,
aic3x_linein_mode_mux);
static const char * const aic3x_adc_hpf[] = {
"Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" };
static SOC_ENUM_DOUBLE_DECL(aic3x_adc_hpf_enum, AIC3X_CODEC_DFILT_CTRL, 6, 4,
aic3x_adc_hpf);
static const char * const aic3x_agc_level[] = {
"-5.5dB", "-8dB", "-10dB", "-12dB",
"-14dB", "-17dB", "-20dB", "-24dB" };
static SOC_ENUM_SINGLE_DECL(aic3x_lagc_level_enum, LAGC_CTRL_A, 4,
aic3x_agc_level);
static SOC_ENUM_SINGLE_DECL(aic3x_ragc_level_enum, RAGC_CTRL_A, 4,
aic3x_agc_level);
static const char * const aic3x_agc_attack[] = {
"8ms", "11ms", "16ms", "20ms" };
static SOC_ENUM_SINGLE_DECL(aic3x_lagc_attack_enum, LAGC_CTRL_A, 2,
aic3x_agc_attack);
static SOC_ENUM_SINGLE_DECL(aic3x_ragc_attack_enum, RAGC_CTRL_A, 2,
aic3x_agc_attack);
static const char * const aic3x_agc_decay[] = {
"100ms", "200ms", "400ms", "500ms" };
static SOC_ENUM_SINGLE_DECL(aic3x_lagc_decay_enum, LAGC_CTRL_A, 0,
aic3x_agc_decay);
static SOC_ENUM_SINGLE_DECL(aic3x_ragc_decay_enum, RAGC_CTRL_A, 0,
aic3x_agc_decay);
static const char * const aic3x_poweron_time[] = {
"0us", "10us", "100us", "1ms", "10ms", "50ms",
"100ms", "200ms", "400ms", "800ms", "2s", "4s" };
static SOC_ENUM_SINGLE_DECL(aic3x_poweron_time_enum, HPOUT_POP_REDUCTION, 4,
aic3x_poweron_time);
static const char * const aic3x_rampup_step[] = { "0ms", "1ms", "2ms", "4ms" };
static SOC_ENUM_SINGLE_DECL(aic3x_rampup_step_enum, HPOUT_POP_REDUCTION, 2,
aic3x_rampup_step);
/*
* 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);
/* Output volumes. From 0 to 9 dB in 1 dB steps */
static const DECLARE_TLV_DB_SCALE(out_tlv, 0, 100, 0);
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 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 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 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 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 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 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 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("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 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 controls */
SOC_DOUBLE_R_TLV("Line Playback Volume", LLOPM_CTRL, RLOPM_CTRL, 4,
9, 0, out_tlv),
SOC_DOUBLE_R("Line Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3,
0x01, 0),
SOC_DOUBLE_R_TLV("HP Playback Volume", HPLOUT_CTRL, HPROUT_CTRL, 4,
9, 0, out_tlv),
SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3,
0x01, 0),
SOC_DOUBLE_R_TLV("HPCOM Playback Volume", HPLCOM_CTRL, HPRCOM_CTRL,
4, 9, 0, out_tlv),
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_lagc_level_enum),
SOC_ENUM("Right AGC Target level", aic3x_ragc_level_enum),
SOC_ENUM("Left AGC Attack time", aic3x_lagc_attack_enum),
SOC_ENUM("Right AGC Attack time", aic3x_ragc_attack_enum),
SOC_ENUM("Left AGC Decay time", aic3x_lagc_decay_enum),
SOC_ENUM("Right AGC Decay time", aic3x_ragc_decay_enum),
/* 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_adc_hpf_enum),
/* Pop reduction */
SOC_ENUM("Output Driver Power-On time", aic3x_poweron_time_enum),
SOC_ENUM("Output Driver Ramp-up step", aic3x_rampup_step_enum),
};
/* For other than tlv320aic3104 */
static const struct snd_kcontrol_new aic3x_extra_snd_controls[] = {
/*
* 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("Right Line Mixer Line2L Bypass Volume",
LINE2L_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("Right HP Mixer Line2L Bypass Volume",
LINE2L_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("Right HPCOM Mixer Line2L Bypass Volume",
LINE2L_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("HP Line2 Bypass Volume",
LINE2L_2_HPLOUT_VOL, LINE2R_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),
};
static const struct snd_kcontrol_new aic3x_mono_controls[] = {
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_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0),
SOC_SINGLE_TLV("Mono Playback Volume", MONOLOPM_CTRL, 4, 9, 0,
out_tlv),
};
/*
* 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_left_dac_enum);
/* Right DAC Mux */
static const struct snd_kcontrol_new aic3x_right_dac_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_right_dac_enum);
/* Left HPCOM Mux */
static const struct snd_kcontrol_new aic3x_left_hpcom_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_left_hpcom_enum);
/* Right HPCOM Mux */
static const struct snd_kcontrol_new aic3x_right_hpcom_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_right_hpcom_enum);
/* Left Line Mixer */
static const struct snd_kcontrol_new aic3x_left_line_mixer_controls[] = {
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("PGAR Bypass Switch", PGAR_2_LLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_LLOPM_VOL, 7, 1, 0),
/* Not on tlv320aic3104 */
SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_LLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_LLOPM_VOL, 7, 1, 0),
};
/* Right Line Mixer */
static const struct snd_kcontrol_new aic3x_right_line_mixer_controls[] = {
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("PGAR Bypass Switch", PGAR_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_RLOPM_VOL, 7, 1, 0),
/* Not on tlv320aic3104 */
SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_RLOPM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_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("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("PGAR Bypass Switch", PGAR_2_HPLOUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPLOUT_VOL, 7, 1, 0),
/* Not on tlv320aic3104 */
SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLOUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLOUT_VOL, 7, 1, 0),
};
/* Right HP Mixer */
static const struct snd_kcontrol_new aic3x_right_hp_mixer_controls[] = {
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("PGAR Bypass Switch", PGAR_2_HPROUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPROUT_VOL, 7, 1, 0),
/* Not on tlv320aic3104 */
SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPROUT_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPROUT_VOL, 7, 1, 0),
};
/* Left HPCOM Mixer */
static const struct snd_kcontrol_new aic3x_left_hpcom_mixer_controls[] = {
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("PGAR Bypass Switch", PGAR_2_HPLCOM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPLCOM_VOL, 7, 1, 0),
/* Not on tlv320aic3104 */
SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPLCOM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_2_HPLCOM_VOL, 7, 1, 0),
};
/* Right HPCOM Mixer */
static const struct snd_kcontrol_new aic3x_right_hpcom_mixer_controls[] = {
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("PGAR Bypass Switch", PGAR_2_HPRCOM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("DACR1 Switch", DACR1_2_HPRCOM_VOL, 7, 1, 0),
/* Not on tlv320aic3104 */
SOC_DAPM_SINGLE("Line2L Bypass Switch", LINE2L_2_HPRCOM_VOL, 7, 1, 0),
SOC_DAPM_SINGLE("Line2R Bypass Switch", LINE2R_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 PGA Mixer for tlv320aic3104 */
static const struct snd_kcontrol_new aic3104_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("Mic2L Switch", MIC3LR_2_LADC_CTRL, 4, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Mic2R Switch", MIC3LR_2_LADC_CTRL, 0, 1, 1),
};
/* Right PGA Mixer for tlv320aic3104 */
static const struct snd_kcontrol_new aic3104_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("Mic2L Switch", MIC3LR_2_RADC_CTRL, 4, 1, 1),
SOC_DAPM_SINGLE_AIC3X("Mic2R 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_line1l_2_l_enum);
static const struct snd_kcontrol_new aic3x_right_line1l_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_line1l_2_r_enum);
/* Right Line1 Mux */
static const struct snd_kcontrol_new aic3x_right_line1r_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_line1r_2_r_enum);
static const struct snd_kcontrol_new aic3x_left_line1r_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_line1r_2_l_enum);
/* Left Line2 Mux */
static const struct snd_kcontrol_new aic3x_left_line2_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_line2l_2_ldac_enum);
/* Right Line2 Mux */
static const struct snd_kcontrol_new aic3x_right_line2_mux_controls =
SOC_DAPM_ENUM("Route", aic3x_line2r_2_rdac_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),
/* Inputs to Left ADC */
SND_SOC_DAPM_ADC("Left ADC", "Left Capture", LINE1L_2_LADC_CTRL, 2, 0),
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),
/* Inputs to Right ADC */
SND_SOC_DAPM_ADC("Right ADC", "Right Capture",
LINE1R_2_RADC_CTRL, 2, 0),
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),
/* 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),
SND_SOC_DAPM_OUTPUT("LLOUT"),
SND_SOC_DAPM_OUTPUT("RLOUT"),
SND_SOC_DAPM_OUTPUT("HPLOUT"),
SND_SOC_DAPM_OUTPUT("HPROUT"),
SND_SOC_DAPM_OUTPUT("HPLCOM"),
SND_SOC_DAPM_OUTPUT("HPRCOM"),
SND_SOC_DAPM_INPUT("LINE1L"),
SND_SOC_DAPM_INPUT("LINE1R"),
/*
* 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"),
};
/* For other than tlv320aic3104 */
static const struct snd_soc_dapm_widget aic3x_extra_dapm_widgets[] = {
/* Inputs to Left ADC */
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 Line2L Mux", SND_SOC_NOPM, 0, 0,
&aic3x_left_line2_mux_controls),
/* Inputs to Right ADC */
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 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),
/* 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("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_INPUT("MIC3L"),
SND_SOC_DAPM_INPUT("MIC3R"),
SND_SOC_DAPM_INPUT("LINE2L"),
SND_SOC_DAPM_INPUT("LINE2R"),
};
/* For tlv320aic3104 */
static const struct snd_soc_dapm_widget aic3104_extra_dapm_widgets[] = {
/* Inputs to Left ADC */
SND_SOC_DAPM_MIXER("Left PGA Mixer", SND_SOC_NOPM, 0, 0,
&aic3104_left_pga_mixer_controls[0],
ARRAY_SIZE(aic3104_left_pga_mixer_controls)),
/* Inputs to Right ADC */
SND_SOC_DAPM_MIXER("Right PGA Mixer", SND_SOC_NOPM, 0, 0,
&aic3104_right_pga_mixer_controls[0],
ARRAY_SIZE(aic3104_right_pga_mixer_controls)),
/* 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) - 2),
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) - 2),
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) - 2),
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) - 2),
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) - 2),
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) - 2),
SND_SOC_DAPM_INPUT("MIC2L"),
SND_SOC_DAPM_INPUT("MIC2R"),
};
static const struct snd_soc_dapm_widget aic3x_dapm_mono_widgets[] = {
/* Mono Output */
SND_SOC_DAPM_PGA("Mono Out", MONOLOPM_CTRL, 0, 0, NULL, 0),
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_OUTPUT("MONO_LOUT"),
};
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 Line1R Mux", "single-ended", "LINE1R"},
{"Left Line1R Mux", "differential", "LINE1R"},
{"Left PGA Mixer", "Line1L Switch", "Left Line1L Mux"},
{"Left PGA Mixer", "Line1R Switch", "Left Line1R Mux"},
{"Left ADC", NULL, "Left PGA Mixer"},
/* Right Input */
{"Right Line1R Mux", "single-ended", "LINE1R"},
{"Right Line1R Mux", "differential", "LINE1R"},
{"Right Line1L Mux", "single-ended", "LINE1L"},
{"Right Line1L Mux", "differential", "LINE1L"},
{"Right PGA Mixer", "Line1L Switch", "Right Line1L Mux"},
{"Right PGA Mixer", "Line1R Switch", "Right Line1R Mux"},
{"Right ADC", NULL, "Right PGA Mixer"},
/* 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", "PGAL Bypass Switch", "Left PGA Mixer"},
{"Left Line Mixer", "DACL1 Switch", "Left DAC 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", "PGAL Bypass Switch", "Left PGA Mixer"},
{"Right Line Mixer", "DACL1 Switch", "Left DAC 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"},
/* Left HP Output */
{"Left HP Mixer", "PGAL Bypass Switch", "Left PGA Mixer"},
{"Left HP Mixer", "DACL1 Switch", "Left DAC 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", "PGAL Bypass Switch", "Left PGA Mixer"},
{"Right HP Mixer", "DACL1 Switch", "Left DAC 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", "PGAL Bypass Switch", "Left PGA Mixer"},
{"Left HPCOM Mixer", "DACL1 Switch", "Left DAC 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", "PGAL Bypass Switch", "Left PGA Mixer"},
{"Right HPCOM Mixer", "DACL1 Switch", "Left DAC 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"},
};
/* For other than tlv320aic3104 */
static const struct snd_soc_dapm_route intercon_extra[] = {
/* Left Input */
{"Left Line2L Mux", "single-ended", "LINE2L"},
{"Left Line2L Mux", "differential", "LINE2L"},
{"Left PGA Mixer", "Line2L Switch", "Left Line2L Mux"},
{"Left PGA Mixer", "Mic3L Switch", "MIC3L"},
{"Left PGA Mixer", "Mic3R Switch", "MIC3R"},
{"Left ADC", NULL, "GPIO1 dmic modclk"},
/* Right Input */
{"Right Line2R Mux", "single-ended", "LINE2R"},
{"Right Line2R Mux", "differential", "LINE2R"},
{"Right PGA Mixer", "Line2R Switch", "Right Line2R Mux"},
{"Right PGA Mixer", "Mic3L Switch", "MIC3L"},
{"Right PGA Mixer", "Mic3R Switch", "MIC3R"},
{"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 Line Output */
{"Left Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
{"Left Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
/* Right Line Output */
{"Right Line Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
{"Right Line Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
/* Left HP Output */
{"Left HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
{"Left HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
/* Right HP Output */
{"Right HP Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
{"Right HP Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
/* Left HPCOM Output */
{"Left HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
{"Left HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
/* Right HPCOM Output */
{"Right HPCOM Mixer", "Line2L Bypass Switch", "Left Line2L Mux"},
{"Right HPCOM Mixer", "Line2R Bypass Switch", "Right Line2R Mux"},
};
/* For tlv320aic3104 */
static const struct snd_soc_dapm_route intercon_extra_3104[] = {
/* Left Input */
{"Left PGA Mixer", "Mic2L Switch", "MIC2L"},
{"Left PGA Mixer", "Mic2R Switch", "MIC2R"},
/* Right Input */
{"Right PGA Mixer", "Mic2L Switch", "MIC2L"},
{"Right PGA Mixer", "Mic2R Switch", "MIC2R"},
};
static const struct snd_soc_dapm_route intercon_mono[] = {
/* 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"},
};
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_component *component)
{
struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component);
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
switch (aic3x->model) {
case AIC3X_MODEL_3X:
case AIC3X_MODEL_33:
snd_soc_dapm_new_controls(dapm, aic3x_extra_dapm_widgets,
ARRAY_SIZE(aic3x_extra_dapm_widgets));
snd_soc_dapm_add_routes(dapm, intercon_extra,
ARRAY_SIZE(intercon_extra));
snd_soc_dapm_new_controls(dapm, aic3x_dapm_mono_widgets,
ARRAY_SIZE(aic3x_dapm_mono_widgets));
snd_soc_dapm_add_routes(dapm, intercon_mono,
ARRAY_SIZE(intercon_mono));
break;
case AIC3X_MODEL_3007:
snd_soc_dapm_new_controls(dapm, aic3x_extra_dapm_widgets,
ARRAY_SIZE(aic3x_extra_dapm_widgets));
snd_soc_dapm_add_routes(dapm, intercon_extra,
ARRAY_SIZE(intercon_extra));
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 07:53:46 -06:00
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 07:53:46 -06:00
snd_soc_dapm_add_routes(dapm, intercon_3007,
ARRAY_SIZE(intercon_3007));
break;
case AIC3X_MODEL_3104:
snd_soc_dapm_new_controls(dapm, aic3104_extra_dapm_widgets,
ARRAY_SIZE(aic3104_extra_dapm_widgets));
snd_soc_dapm_add_routes(dapm, intercon_extra_3104,
ARRAY_SIZE(intercon_extra_3104));
break;
}
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_component *component = dai->component;
struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component);
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;
int width = aic3x->slot_width;
if (!width)
width = params_width(params);
/* select data word length */
data = snd_soc_component_read32(component, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
switch (width) {
case 16:
break;
case 20:
data |= (0x01 << 4);
break;
case 24:
data |= (0x02 << 4);
break;
case 32:
data |= (0x03 << 4);
break;
}
snd_soc_component_write(component, 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_component_write(component, AIC3X_PLL_PROGA_REG, pll_q << PLLQ_SHIFT);
snd_soc_component_write(component, AIC3X_GPIOB_REG, CODEC_CLKIN_CLKDIV);
/* disable PLL if it is bypassed */
snd_soc_component_update_bits(component, AIC3X_PLL_PROGA_REG, PLL_ENABLE, 0);
} else {
snd_soc_component_write(component, AIC3X_GPIOB_REG, CODEC_CLKIN_PLLDIV);
/* enable PLL when it is used */
snd_soc_component_update_bits(component, 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_component_write(component, 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_component_write(component, 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_component_update_bits(component, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p);
snd_soc_component_write(component, AIC3X_OVRF_STATUS_AND_PLLR_REG,
pll_r << PLLR_SHIFT);
snd_soc_component_write(component, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT);
snd_soc_component_write(component, AIC3X_PLL_PROGC_REG,
(pll_d >> 6) << PLLD_MSB_SHIFT);
snd_soc_component_write(component, AIC3X_PLL_PROGD_REG,
(pll_d & 0x3F) << PLLD_LSB_SHIFT);
return 0;
}
static int aic3x_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component);
int delay = 0;
int width = aic3x->slot_width;
if (!width)
width = substream->runtime->sample_bits;
/* TDM slot selection only valid in DSP_A/_B mode */
if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_A)
delay += (aic3x->tdm_delay*width + 1);
else if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_B)
delay += aic3x->tdm_delay*width;
/* Configure data delay */
snd_soc_component_write(component, AIC3X_ASD_INTF_CTRLC, delay);
return 0;
}
static int aic3x_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_component *component = dai->component;
u8 ldac_reg = snd_soc_component_read32(component, LDAC_VOL) & ~MUTE_ON;
u8 rdac_reg = snd_soc_component_read32(component, RDAC_VOL) & ~MUTE_ON;
if (mute) {
snd_soc_component_write(component, LDAC_VOL, ldac_reg | MUTE_ON);
snd_soc_component_write(component, RDAC_VOL, rdac_reg | MUTE_ON);
} else {
snd_soc_component_write(component, LDAC_VOL, ldac_reg);
snd_soc_component_write(component, 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_component *component = codec_dai->component;
struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component);
/* set clock on MCLK or GPIO2 or BCLK */
snd_soc_component_update_bits(component, AIC3X_CLKGEN_CTRL_REG, PLLCLK_IN_MASK,
clk_id << PLLCLK_IN_SHIFT);
snd_soc_component_update_bits(component, 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_component *component = codec_dai->component;
struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component);
u8 iface_areg, iface_breg;
iface_areg = snd_soc_component_read32(component, AIC3X_ASD_INTF_CTRLA) & 0x3f;
iface_breg = snd_soc_component_read32(component, 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;
case SND_SOC_DAIFMT_CBM_CFS:
aic3x->master = 1;
iface_areg |= BIT_CLK_MASTER;
iface_areg &= ~WORD_CLK_MASTER;
break;
case SND_SOC_DAIFMT_CBS_CFM:
aic3x->master = 1;
iface_areg |= WORD_CLK_MASTER;
iface_areg &= ~BIT_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):
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;
}
aic3x->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
/* set iface */
snd_soc_component_write(component, AIC3X_ASD_INTF_CTRLA, iface_areg);
snd_soc_component_write(component, AIC3X_ASD_INTF_CTRLB, iface_breg);
return 0;
}
static int aic3x_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
unsigned int tx_mask, unsigned int rx_mask,
int slots, int slot_width)
{
struct snd_soc_component *component = codec_dai->component;
struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component);
unsigned int lsb;
if (tx_mask != rx_mask) {
dev_err(component->dev, "tx and rx masks must be symmetric\n");
return -EINVAL;
}
if (unlikely(!tx_mask)) {
dev_err(component->dev, "tx and rx masks need to be non 0\n");
return -EINVAL;
}
/* TDM based on DSP mode requires slots to be adjacent */
lsb = __ffs(tx_mask);
if ((lsb + 1) != __fls(tx_mask)) {
dev_err(component->dev, "Invalid mask, slots must be adjacent\n");
return -EINVAL;
}
switch (slot_width) {
case 16:
case 20:
case 24:
case 32:
break;
default:
dev_err(component->dev, "Unsupported slot width %d\n", slot_width);
return -EINVAL;
}
aic3x->tdm_delay = lsb;
aic3x->slot_width = slot_width;
/* DOUT in high-impedance on inactive bit clocks */
snd_soc_component_update_bits(component, AIC3X_ASD_INTF_CTRLA,
DOUT_TRISTATE, DOUT_TRISTATE);
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);
regcache_mark_dirty(aic3x->regmap);
}
return 0;
}
static int aic3x_set_power(struct snd_soc_component *component, int power)
{
struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component);
unsigned int pll_c, pll_d;
int ret;
if (power) {
ret = regulator_bulk_enable(ARRAY_SIZE(aic3x->supplies),
aic3x->supplies);
if (ret)
goto out;
aic3x->power = 1;
if (gpio_is_valid(aic3x->gpio_reset)) {
udelay(1);
gpio_set_value(aic3x->gpio_reset, 1);
}
/* Sync reg_cache with the hardware */
regcache_cache_only(aic3x->regmap, false);
regcache_sync(aic3x->regmap);
/* Rewrite paired PLL D registers in case cached sync skipped
* writing one of them and thus caused other one also not
* being written
*/
pll_c = snd_soc_component_read32(component, AIC3X_PLL_PROGC_REG);
pll_d = snd_soc_component_read32(component, AIC3X_PLL_PROGD_REG);
if (pll_c == aic3x_reg[AIC3X_PLL_PROGC_REG].def ||
pll_d == aic3x_reg[AIC3X_PLL_PROGD_REG].def) {
snd_soc_component_write(component, AIC3X_PLL_PROGC_REG, pll_c);
snd_soc_component_write(component, AIC3X_PLL_PROGD_REG, pll_d);
}
/*
* Delay is needed to reduce pop-noise after syncing back the
* registers
*/
mdelay(50);
} 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_component_write(component, AIC3X_RESET, SOFT_RESET);
regcache_mark_dirty(aic3x->regmap);
aic3x->power = 0;
/* HW writes are needless when bias is off */
regcache_cache_only(aic3x->regmap, true);
ret = regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies),
aic3x->supplies);
}
out:
return ret;
}
static int aic3x_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component);
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_STANDBY &&
aic3x->master) {
/* enable pll */
snd_soc_component_update_bits(component, AIC3X_PLL_PROGA_REG,
PLL_ENABLE, PLL_ENABLE);
}
break;
case SND_SOC_BIAS_STANDBY:
if (!aic3x->power)
aic3x_set_power(component, 1);
if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_PREPARE &&
aic3x->master) {
/* disable pll */
snd_soc_component_update_bits(component, AIC3X_PLL_PROGA_REG,
PLL_ENABLE, 0);
}
break;
case SND_SOC_BIAS_OFF:
if (aic3x->power)
aic3x_set_power(component, 0);
break;
}
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_S24_LE | \
SNDRV_PCM_FMTBIT_S32_LE)
static const struct snd_soc_dai_ops aic3x_dai_ops = {
.hw_params = aic3x_hw_params,
.prepare = aic3x_prepare,
.digital_mute = aic3x_mute,
.set_sysclk = aic3x_set_dai_sysclk,
.set_fmt = aic3x_set_dai_fmt,
.set_tdm_slot = aic3x_set_dai_tdm_slot,
};
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 14:15:21 -06:00
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 void aic3x_mono_init(struct snd_soc_component *component)
{
/* DAC to Mono Line Out default volume and route to Output mixer */
snd_soc_component_write(component, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
snd_soc_component_write(component, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
/* unmute all outputs */
snd_soc_component_update_bits(component, MONOLOPM_CTRL, UNMUTE, UNMUTE);
/* PGA to Mono Line Out default volume, disconnect from Output Mixer */
snd_soc_component_write(component, PGAL_2_MONOLOPM_VOL, DEFAULT_VOL);
snd_soc_component_write(component, PGAR_2_MONOLOPM_VOL, DEFAULT_VOL);
/* Line2 to Mono Out default volume, disconnect from Output Mixer */
snd_soc_component_write(component, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL);
snd_soc_component_write(component, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL);
}
/*
* initialise the AIC3X driver
* register the mixer and dsp interfaces with the kernel
*/
static int aic3x_init(struct snd_soc_component *component)
{
struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component);
snd_soc_component_write(component, AIC3X_PAGE_SELECT, PAGE0_SELECT);
snd_soc_component_write(component, AIC3X_RESET, SOFT_RESET);
/* DAC default volume and mute */
snd_soc_component_write(component, LDAC_VOL, DEFAULT_VOL | MUTE_ON);
snd_soc_component_write(component, RDAC_VOL, DEFAULT_VOL | MUTE_ON);
/* DAC to HP default volume and route to Output mixer */
snd_soc_component_write(component, DACL1_2_HPLOUT_VOL, DEFAULT_VOL | ROUTE_ON);
snd_soc_component_write(component, DACR1_2_HPROUT_VOL, DEFAULT_VOL | ROUTE_ON);
snd_soc_component_write(component, DACL1_2_HPLCOM_VOL, DEFAULT_VOL | ROUTE_ON);
snd_soc_component_write(component, DACR1_2_HPRCOM_VOL, DEFAULT_VOL | ROUTE_ON);
/* DAC to Line Out default volume and route to Output mixer */
snd_soc_component_write(component, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
snd_soc_component_write(component, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON);
/* unmute all outputs */
snd_soc_component_update_bits(component, LLOPM_CTRL, UNMUTE, UNMUTE);
snd_soc_component_update_bits(component, RLOPM_CTRL, UNMUTE, UNMUTE);
snd_soc_component_update_bits(component, HPLOUT_CTRL, UNMUTE, UNMUTE);
snd_soc_component_update_bits(component, HPROUT_CTRL, UNMUTE, UNMUTE);
snd_soc_component_update_bits(component, HPLCOM_CTRL, UNMUTE, UNMUTE);
snd_soc_component_update_bits(component, HPRCOM_CTRL, UNMUTE, UNMUTE);
/* ADC default volume and unmute */
snd_soc_component_write(component, LADC_VOL, DEFAULT_GAIN);
snd_soc_component_write(component, RADC_VOL, DEFAULT_GAIN);
/* By default route Line1 to ADC PGA mixer */
snd_soc_component_write(component, LINE1L_2_LADC_CTRL, 0x0);
snd_soc_component_write(component, LINE1R_2_RADC_CTRL, 0x0);
/* PGA to HP Bypass default volume, disconnect from Output Mixer */
snd_soc_component_write(component, PGAL_2_HPLOUT_VOL, DEFAULT_VOL);
snd_soc_component_write(component, PGAR_2_HPROUT_VOL, DEFAULT_VOL);
snd_soc_component_write(component, PGAL_2_HPLCOM_VOL, DEFAULT_VOL);
snd_soc_component_write(component, PGAR_2_HPRCOM_VOL, DEFAULT_VOL);
/* PGA to Line Out default volume, disconnect from Output Mixer */
snd_soc_component_write(component, PGAL_2_LLOPM_VOL, DEFAULT_VOL);
snd_soc_component_write(component, PGAR_2_RLOPM_VOL, DEFAULT_VOL);
/* On tlv320aic3104, these registers are reserved and must not be written */
if (aic3x->model != AIC3X_MODEL_3104) {
/* Line2 to HP Bypass default volume, disconnect from Output Mixer */
snd_soc_component_write(component, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL);
snd_soc_component_write(component, LINE2R_2_HPROUT_VOL, DEFAULT_VOL);
snd_soc_component_write(component, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL);
snd_soc_component_write(component, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL);
/* Line2 Line Out default volume, disconnect from Output Mixer */
snd_soc_component_write(component, LINE2L_2_LLOPM_VOL, DEFAULT_VOL);
snd_soc_component_write(component, LINE2R_2_RLOPM_VOL, DEFAULT_VOL);
}
switch (aic3x->model) {
case AIC3X_MODEL_3X:
case AIC3X_MODEL_33:
aic3x_mono_init(component);
break;
case AIC3X_MODEL_3007:
snd_soc_component_write(component, CLASSD_CTRL, 0);
break;
}
/* Output common-mode voltage = 1.5 V */
snd_soc_component_update_bits(component, HPOUT_SC, HPOUT_SC_OCMV_MASK,
aic3x->ocmv << HPOUT_SC_OCMV_SHIFT);
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;
}
static int aic3x_probe(struct snd_soc_component *component)
{
struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component);
int ret, i;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 14:15:21 -06:00
aic3x->component = component;
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 = devm_regulator_register_notifier(
aic3x->supplies[i].consumer,
&aic3x->disable_nb[i].nb);
if (ret) {
dev_err(component->dev,
"Failed to request regulator notifier: %d\n",
ret);
return ret;
}
}
regcache_mark_dirty(aic3x->regmap);
aic3x_init(component);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 14:15:21 -06:00
if (aic3x->setup) {
if (aic3x->model != AIC3X_MODEL_3104) {
/* setup GPIO functions */
snd_soc_component_write(component, AIC3X_GPIO1_REG,
(aic3x->setup->gpio_func[0] & 0xf) << 4);
snd_soc_component_write(component, AIC3X_GPIO2_REG,
(aic3x->setup->gpio_func[1] & 0xf) << 4);
} else {
dev_warn(component->dev, "GPIO functionality is not supported on tlv320aic3104\n");
}
}
switch (aic3x->model) {
case AIC3X_MODEL_3X:
case AIC3X_MODEL_33:
snd_soc_add_component_controls(component, aic3x_extra_snd_controls,
ARRAY_SIZE(aic3x_extra_snd_controls));
snd_soc_add_component_controls(component, aic3x_mono_controls,
ARRAY_SIZE(aic3x_mono_controls));
break;
case AIC3X_MODEL_3007:
snd_soc_add_component_controls(component, aic3x_extra_snd_controls,
ARRAY_SIZE(aic3x_extra_snd_controls));
snd_soc_add_component_controls(component,
&aic3x_classd_amp_gain_ctrl, 1);
break;
case AIC3X_MODEL_3104:
break;
}
/* set mic bias voltage */
switch (aic3x->micbias_vg) {
case AIC3X_MICBIAS_2_0V:
case AIC3X_MICBIAS_2_5V:
case AIC3X_MICBIAS_AVDDV:
snd_soc_component_update_bits(component, 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;
}
aic3x_add_widgets(component);
return 0;
}
static const struct snd_soc_component_driver soc_component_dev_aic3x = {
.set_bias_level = aic3x_set_bias_level,
.probe = aic3x_probe,
.controls = aic3x_snd_controls,
.num_controls = ARRAY_SIZE(aic3x_snd_controls),
.dapm_widgets = aic3x_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets),
.dapm_routes = intercon,
.num_dapm_routes = ARRAY_SIZE(intercon),
.use_pmdown_time = 1,
.endianness = 1,
.non_legacy_dai_naming = 1,
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 14:15:21 -06:00
};
static void aic3x_configure_ocmv(struct i2c_client *client)
{
struct device_node *np = client->dev.of_node;
struct aic3x_priv *aic3x = i2c_get_clientdata(client);
u32 value;
int dvdd, avdd;
if (np && !of_property_read_u32(np, "ai3x-ocmv", &value)) {
/* OCMV setting is forced by DT */
if (value <= 3) {
aic3x->ocmv = value;
return;
}
}
dvdd = regulator_get_voltage(aic3x->supplies[1].consumer);
avdd = regulator_get_voltage(aic3x->supplies[2].consumer);
if (avdd > 3600000 || dvdd > 1950000) {
dev_warn(&client->dev,
"Too high supply voltage(s) AVDD: %d, DVDD: %d\n",
avdd, dvdd);
} else if (avdd == 3600000 && dvdd == 1950000) {
aic3x->ocmv = HPOUT_SC_OCMV_1_8V;
} else if (avdd > 3300000 && dvdd > 1800000) {
aic3x->ocmv = HPOUT_SC_OCMV_1_65V;
} else if (avdd > 3000000 && dvdd > 1650000) {
aic3x->ocmv = HPOUT_SC_OCMV_1_5V;
} else if (avdd >= 2700000 && dvdd >= 1525000) {
aic3x->ocmv = HPOUT_SC_OCMV_1_35V;
} else {
dev_warn(&client->dev,
"Invalid supply voltage(s) AVDD: %d, DVDD: %d\n",
avdd, dvdd);
}
}
/*
* 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 },
{ "tlv320aic3106", AIC3X_MODEL_3X },
{ "tlv320aic3104", AIC3X_MODEL_3104 },
{ }
};
MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
static const struct reg_sequence aic3007_class_d[] = {
/* Class-D speaker driver init; datasheet p. 46 */
{ AIC3X_PAGE_SELECT, 0x0D },
{ 0xD, 0x0D },
{ 0x8, 0x5C },
{ 0x8, 0x5D },
{ 0x8, 0x5C },
{ AIC3X_PAGE_SELECT, 0x00 },
};
/*
* 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>
2010-03-17 14:15:21 -06:00
struct aic3x_priv *aic3x;
struct aic3x_setup_data *ai3x_setup;
struct device_node *np = i2c->dev.of_node;
int ret, i;
u32 value;
aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL);
if (!aic3x)
return -ENOMEM;
aic3x->regmap = devm_regmap_init_i2c(i2c, &aic3x_regmap);
if (IS_ERR(aic3x->regmap)) {
ret = PTR_ERR(aic3x->regmap);
return ret;
}
regcache_cache_only(aic3x->regmap, true);
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)
return -ENOMEM;
ret = of_get_named_gpio(np, "reset-gpios", 0);
if (ret >= 0) {
aic3x->gpio_reset = ret;
} else {
ret = of_get_named_gpio(np, "gpio-reset", 0);
if (ret > 0) {
dev_warn(&i2c->dev, "Using deprecated property \"gpio-reset\", please update your DT");
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;
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_direction_output(aic3x->gpio_reset, 0);
}
for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++)
aic3x->supplies[i].supply = aic3x_supply_names[i];
ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(aic3x->supplies),
aic3x->supplies);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret);
goto err_gpio;
}
aic3x_configure_ocmv(i2c);
if (aic3x->model == AIC3X_MODEL_3007) {
ret = regmap_register_patch(aic3x->regmap, aic3007_class_d,
ARRAY_SIZE(aic3007_class_d));
if (ret != 0)
dev_err(&i2c->dev, "Failed to init class D: %d\n",
ret);
}
ret = devm_snd_soc_register_component(&i2c->dev,
&soc_component_dev_aic3x, &aic3x_dai, 1);
if (ret != 0)
goto err_gpio;
INIT_LIST_HEAD(&aic3x->list);
list_add(&aic3x->list, &reset_list);
return 0;
err_gpio:
if (gpio_is_valid(aic3x->gpio_reset) &&
!aic3x_is_shared_reset(aic3x))
gpio_free(aic3x->gpio_reset);
err:
return ret;
}
static int aic3x_i2c_remove(struct i2c_client *client)
{
struct aic3x_priv *aic3x = i2c_get_clientdata(client);
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);
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 14:15:21 -06:00
return 0;
}
#if defined(CONFIG_OF)
static const struct of_device_id tlv320aic3x_of_match[] = {
{ .compatible = "ti,tlv320aic3x", },
{ .compatible = "ti,tlv320aic33" },
{ .compatible = "ti,tlv320aic3007" },
{ .compatible = "ti,tlv320aic3106" },
{ .compatible = "ti,tlv320aic3104" },
{},
};
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>
2010-03-17 14:15:21 -06:00
.name = "tlv320aic3x-codec",
.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");