kernel-ark/sound/soc/codecs/wm8904.c

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/*
* wm8904.c -- WM8904 ALSA SoC Audio driver
*
* Copyright 2009-12 Wolfson Microelectronics plc
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/regmap.h>
#include <linux/regulator/consumer.h>
include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo <tj@kernel.org> Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org> Cc: Ingo Molnar <mingo@redhat.com> Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
2010-03-24 08:04:11 +00:00
#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/wm8904.h>
#include "wm8904.h"
enum wm8904_type {
WM8904,
WM8912,
};
#define WM8904_NUM_DCS_CHANNELS 4
#define WM8904_NUM_SUPPLIES 5
static const char *wm8904_supply_names[WM8904_NUM_SUPPLIES] = {
"DCVDD",
"DBVDD",
"AVDD",
"CPVDD",
"MICVDD",
};
/* codec private data */
struct wm8904_priv {
struct regmap *regmap;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
enum wm8904_type devtype;
struct regulator_bulk_data supplies[WM8904_NUM_SUPPLIES];
struct wm8904_pdata *pdata;
int deemph;
/* Platform provided DRC configuration */
const char **drc_texts;
int drc_cfg;
struct soc_enum drc_enum;
/* Platform provided ReTune mobile configuration */
int num_retune_mobile_texts;
const char **retune_mobile_texts;
int retune_mobile_cfg;
struct soc_enum retune_mobile_enum;
/* FLL setup */
int fll_src;
int fll_fref;
int fll_fout;
/* Clocking configuration */
unsigned int mclk_rate;
int sysclk_src;
unsigned int sysclk_rate;
int tdm_width;
int tdm_slots;
int bclk;
int fs;
/* DC servo configuration - cached offset values */
int dcs_state[WM8904_NUM_DCS_CHANNELS];
};
static const struct reg_default wm8904_reg_defaults[] = {
{ 4, 0x0018 }, /* R4 - Bias Control 0 */
{ 5, 0x0000 }, /* R5 - VMID Control 0 */
{ 6, 0x0000 }, /* R6 - Mic Bias Control 0 */
{ 7, 0x0000 }, /* R7 - Mic Bias Control 1 */
{ 8, 0x0001 }, /* R8 - Analogue DAC 0 */
{ 9, 0x9696 }, /* R9 - mic Filter Control */
{ 10, 0x0001 }, /* R10 - Analogue ADC 0 */
{ 12, 0x0000 }, /* R12 - Power Management 0 */
{ 14, 0x0000 }, /* R14 - Power Management 2 */
{ 15, 0x0000 }, /* R15 - Power Management 3 */
{ 18, 0x0000 }, /* R18 - Power Management 6 */
{ 19, 0x945E }, /* R20 - Clock Rates 0 */
{ 21, 0x0C05 }, /* R21 - Clock Rates 1 */
{ 22, 0x0006 }, /* R22 - Clock Rates 2 */
{ 24, 0x0050 }, /* R24 - Audio Interface 0 */
{ 25, 0x000A }, /* R25 - Audio Interface 1 */
{ 26, 0x00E4 }, /* R26 - Audio Interface 2 */
{ 27, 0x0040 }, /* R27 - Audio Interface 3 */
{ 30, 0x00C0 }, /* R30 - DAC Digital Volume Left */
{ 31, 0x00C0 }, /* R31 - DAC Digital Volume Right */
{ 32, 0x0000 }, /* R32 - DAC Digital 0 */
{ 33, 0x0008 }, /* R33 - DAC Digital 1 */
{ 36, 0x00C0 }, /* R36 - ADC Digital Volume Left */
{ 37, 0x00C0 }, /* R37 - ADC Digital Volume Right */
{ 38, 0x0010 }, /* R38 - ADC Digital 0 */
{ 39, 0x0000 }, /* R39 - Digital Microphone 0 */
{ 40, 0x01AF }, /* R40 - DRC 0 */
{ 41, 0x3248 }, /* R41 - DRC 1 */
{ 42, 0x0000 }, /* R42 - DRC 2 */
{ 43, 0x0000 }, /* R43 - DRC 3 */
{ 44, 0x0085 }, /* R44 - Analogue Left Input 0 */
{ 45, 0x0085 }, /* R45 - Analogue Right Input 0 */
{ 46, 0x0044 }, /* R46 - Analogue Left Input 1 */
{ 47, 0x0044 }, /* R47 - Analogue Right Input 1 */
{ 57, 0x002D }, /* R57 - Analogue OUT1 Left */
{ 58, 0x002D }, /* R58 - Analogue OUT1 Right */
{ 59, 0x0039 }, /* R59 - Analogue OUT2 Left */
{ 60, 0x0039 }, /* R60 - Analogue OUT2 Right */
{ 61, 0x0000 }, /* R61 - Analogue OUT12 ZC */
{ 67, 0x0000 }, /* R67 - DC Servo 0 */
{ 69, 0xAAAA }, /* R69 - DC Servo 2 */
{ 71, 0xAAAA }, /* R71 - DC Servo 4 */
{ 72, 0xAAAA }, /* R72 - DC Servo 5 */
{ 90, 0x0000 }, /* R90 - Analogue HP 0 */
{ 94, 0x0000 }, /* R94 - Analogue Lineout 0 */
{ 98, 0x0000 }, /* R98 - Charge Pump 0 */
{ 104, 0x0004 }, /* R104 - Class W 0 */
{ 108, 0x0000 }, /* R108 - Write Sequencer 0 */
{ 109, 0x0000 }, /* R109 - Write Sequencer 1 */
{ 110, 0x0000 }, /* R110 - Write Sequencer 2 */
{ 111, 0x0000 }, /* R111 - Write Sequencer 3 */
{ 112, 0x0000 }, /* R112 - Write Sequencer 4 */
{ 116, 0x0000 }, /* R116 - FLL Control 1 */
{ 117, 0x0007 }, /* R117 - FLL Control 2 */
{ 118, 0x0000 }, /* R118 - FLL Control 3 */
{ 119, 0x2EE0 }, /* R119 - FLL Control 4 */
{ 120, 0x0004 }, /* R120 - FLL Control 5 */
{ 121, 0x0014 }, /* R121 - GPIO Control 1 */
{ 122, 0x0010 }, /* R122 - GPIO Control 2 */
{ 123, 0x0010 }, /* R123 - GPIO Control 3 */
{ 124, 0x0000 }, /* R124 - GPIO Control 4 */
{ 126, 0x0000 }, /* R126 - Digital Pulls */
{ 128, 0xFFFF }, /* R128 - Interrupt Status Mask */
{ 129, 0x0000 }, /* R129 - Interrupt Polarity */
{ 130, 0x0000 }, /* R130 - Interrupt Debounce */
{ 134, 0x0000 }, /* R134 - EQ1 */
{ 135, 0x000C }, /* R135 - EQ2 */
{ 136, 0x000C }, /* R136 - EQ3 */
{ 137, 0x000C }, /* R137 - EQ4 */
{ 138, 0x000C }, /* R138 - EQ5 */
{ 139, 0x000C }, /* R139 - EQ6 */
{ 140, 0x0FCA }, /* R140 - EQ7 */
{ 141, 0x0400 }, /* R141 - EQ8 */
{ 142, 0x00D8 }, /* R142 - EQ9 */
{ 143, 0x1EB5 }, /* R143 - EQ10 */
{ 144, 0xF145 }, /* R144 - EQ11 */
{ 145, 0x0B75 }, /* R145 - EQ12 */
{ 146, 0x01C5 }, /* R146 - EQ13 */
{ 147, 0x1C58 }, /* R147 - EQ14 */
{ 148, 0xF373 }, /* R148 - EQ15 */
{ 149, 0x0A54 }, /* R149 - EQ16 */
{ 150, 0x0558 }, /* R150 - EQ17 */
{ 151, 0x168E }, /* R151 - EQ18 */
{ 152, 0xF829 }, /* R152 - EQ19 */
{ 153, 0x07AD }, /* R153 - EQ20 */
{ 154, 0x1103 }, /* R154 - EQ21 */
{ 155, 0x0564 }, /* R155 - EQ22 */
{ 156, 0x0559 }, /* R156 - EQ23 */
{ 157, 0x4000 }, /* R157 - EQ24 */
{ 161, 0x0000 }, /* R161 - Control Interface Test 1 */
{ 204, 0x0000 }, /* R204 - Analogue Output Bias 0 */
{ 247, 0x0000 }, /* R247 - FLL NCO Test 0 */
{ 248, 0x0019 }, /* R248 - FLL NCO Test 1 */
};
static bool wm8904_volatile_register(struct device *dev, unsigned int reg)
{
switch (reg) {
case WM8904_SW_RESET_AND_ID:
case WM8904_REVISION:
case WM8904_DC_SERVO_1:
case WM8904_DC_SERVO_6:
case WM8904_DC_SERVO_7:
case WM8904_DC_SERVO_8:
case WM8904_DC_SERVO_9:
case WM8904_DC_SERVO_READBACK_0:
case WM8904_INTERRUPT_STATUS:
return true;
default:
return false;
}
}
static bool wm8904_readable_register(struct device *dev, unsigned int reg)
{
switch (reg) {
case WM8904_SW_RESET_AND_ID:
case WM8904_REVISION:
case WM8904_BIAS_CONTROL_0:
case WM8904_VMID_CONTROL_0:
case WM8904_MIC_BIAS_CONTROL_0:
case WM8904_MIC_BIAS_CONTROL_1:
case WM8904_ANALOGUE_DAC_0:
case WM8904_MIC_FILTER_CONTROL:
case WM8904_ANALOGUE_ADC_0:
case WM8904_POWER_MANAGEMENT_0:
case WM8904_POWER_MANAGEMENT_2:
case WM8904_POWER_MANAGEMENT_3:
case WM8904_POWER_MANAGEMENT_6:
case WM8904_CLOCK_RATES_0:
case WM8904_CLOCK_RATES_1:
case WM8904_CLOCK_RATES_2:
case WM8904_AUDIO_INTERFACE_0:
case WM8904_AUDIO_INTERFACE_1:
case WM8904_AUDIO_INTERFACE_2:
case WM8904_AUDIO_INTERFACE_3:
case WM8904_DAC_DIGITAL_VOLUME_LEFT:
case WM8904_DAC_DIGITAL_VOLUME_RIGHT:
case WM8904_DAC_DIGITAL_0:
case WM8904_DAC_DIGITAL_1:
case WM8904_ADC_DIGITAL_VOLUME_LEFT:
case WM8904_ADC_DIGITAL_VOLUME_RIGHT:
case WM8904_ADC_DIGITAL_0:
case WM8904_DIGITAL_MICROPHONE_0:
case WM8904_DRC_0:
case WM8904_DRC_1:
case WM8904_DRC_2:
case WM8904_DRC_3:
case WM8904_ANALOGUE_LEFT_INPUT_0:
case WM8904_ANALOGUE_RIGHT_INPUT_0:
case WM8904_ANALOGUE_LEFT_INPUT_1:
case WM8904_ANALOGUE_RIGHT_INPUT_1:
case WM8904_ANALOGUE_OUT1_LEFT:
case WM8904_ANALOGUE_OUT1_RIGHT:
case WM8904_ANALOGUE_OUT2_LEFT:
case WM8904_ANALOGUE_OUT2_RIGHT:
case WM8904_ANALOGUE_OUT12_ZC:
case WM8904_DC_SERVO_0:
case WM8904_DC_SERVO_1:
case WM8904_DC_SERVO_2:
case WM8904_DC_SERVO_4:
case WM8904_DC_SERVO_5:
case WM8904_DC_SERVO_6:
case WM8904_DC_SERVO_7:
case WM8904_DC_SERVO_8:
case WM8904_DC_SERVO_9:
case WM8904_DC_SERVO_READBACK_0:
case WM8904_ANALOGUE_HP_0:
case WM8904_ANALOGUE_LINEOUT_0:
case WM8904_CHARGE_PUMP_0:
case WM8904_CLASS_W_0:
case WM8904_WRITE_SEQUENCER_0:
case WM8904_WRITE_SEQUENCER_1:
case WM8904_WRITE_SEQUENCER_2:
case WM8904_WRITE_SEQUENCER_3:
case WM8904_WRITE_SEQUENCER_4:
case WM8904_FLL_CONTROL_1:
case WM8904_FLL_CONTROL_2:
case WM8904_FLL_CONTROL_3:
case WM8904_FLL_CONTROL_4:
case WM8904_FLL_CONTROL_5:
case WM8904_GPIO_CONTROL_1:
case WM8904_GPIO_CONTROL_2:
case WM8904_GPIO_CONTROL_3:
case WM8904_GPIO_CONTROL_4:
case WM8904_DIGITAL_PULLS:
case WM8904_INTERRUPT_STATUS:
case WM8904_INTERRUPT_STATUS_MASK:
case WM8904_INTERRUPT_POLARITY:
case WM8904_INTERRUPT_DEBOUNCE:
case WM8904_EQ1:
case WM8904_EQ2:
case WM8904_EQ3:
case WM8904_EQ4:
case WM8904_EQ5:
case WM8904_EQ6:
case WM8904_EQ7:
case WM8904_EQ8:
case WM8904_EQ9:
case WM8904_EQ10:
case WM8904_EQ11:
case WM8904_EQ12:
case WM8904_EQ13:
case WM8904_EQ14:
case WM8904_EQ15:
case WM8904_EQ16:
case WM8904_EQ17:
case WM8904_EQ18:
case WM8904_EQ19:
case WM8904_EQ20:
case WM8904_EQ21:
case WM8904_EQ22:
case WM8904_EQ23:
case WM8904_EQ24:
case WM8904_CONTROL_INTERFACE_TEST_1:
case WM8904_ADC_TEST_0:
case WM8904_ANALOGUE_OUTPUT_BIAS_0:
case WM8904_FLL_NCO_TEST_0:
case WM8904_FLL_NCO_TEST_1:
return true;
default:
return true;
}
}
static int wm8904_configure_clocking(struct snd_soc_codec *codec)
{
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
unsigned int clock0, clock2, rate;
/* Gate the clock while we're updating to avoid misclocking */
clock2 = snd_soc_read(codec, WM8904_CLOCK_RATES_2);
snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2,
WM8904_SYSCLK_SRC, 0);
/* This should be done on init() for bypass paths */
switch (wm8904->sysclk_src) {
case WM8904_CLK_MCLK:
dev_dbg(codec->dev, "Using %dHz MCLK\n", wm8904->mclk_rate);
clock2 &= ~WM8904_SYSCLK_SRC;
rate = wm8904->mclk_rate;
/* Ensure the FLL is stopped */
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1,
WM8904_FLL_OSC_ENA | WM8904_FLL_ENA, 0);
break;
case WM8904_CLK_FLL:
dev_dbg(codec->dev, "Using %dHz FLL clock\n",
wm8904->fll_fout);
clock2 |= WM8904_SYSCLK_SRC;
rate = wm8904->fll_fout;
break;
default:
dev_err(codec->dev, "System clock not configured\n");
return -EINVAL;
}
/* SYSCLK shouldn't be over 13.5MHz */
if (rate > 13500000) {
clock0 = WM8904_MCLK_DIV;
wm8904->sysclk_rate = rate / 2;
} else {
clock0 = 0;
wm8904->sysclk_rate = rate;
}
snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0, WM8904_MCLK_DIV,
clock0);
snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2,
WM8904_CLK_SYS_ENA | WM8904_SYSCLK_SRC, clock2);
dev_dbg(codec->dev, "CLK_SYS is %dHz\n", wm8904->sysclk_rate);
return 0;
}
static void wm8904_set_drc(struct snd_soc_codec *codec)
{
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
struct wm8904_pdata *pdata = wm8904->pdata;
int save, i;
/* Save any enables; the configuration should clear them. */
save = snd_soc_read(codec, WM8904_DRC_0);
for (i = 0; i < WM8904_DRC_REGS; i++)
snd_soc_update_bits(codec, WM8904_DRC_0 + i, 0xffff,
pdata->drc_cfgs[wm8904->drc_cfg].regs[i]);
/* Reenable the DRC */
snd_soc_update_bits(codec, WM8904_DRC_0,
WM8904_DRC_ENA | WM8904_DRC_DAC_PATH, save);
}
static int wm8904_put_drc_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
struct wm8904_pdata *pdata = wm8904->pdata;
int value = ucontrol->value.integer.value[0];
if (value >= pdata->num_drc_cfgs)
return -EINVAL;
wm8904->drc_cfg = value;
wm8904_set_drc(codec);
return 0;
}
static int wm8904_get_drc_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
ucontrol->value.enumerated.item[0] = wm8904->drc_cfg;
return 0;
}
static void wm8904_set_retune_mobile(struct snd_soc_codec *codec)
{
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
struct wm8904_pdata *pdata = wm8904->pdata;
int best, best_val, save, i, cfg;
if (!pdata || !wm8904->num_retune_mobile_texts)
return;
/* Find the version of the currently selected configuration
* with the nearest sample rate. */
cfg = wm8904->retune_mobile_cfg;
best = 0;
best_val = INT_MAX;
for (i = 0; i < pdata->num_retune_mobile_cfgs; i++) {
if (strcmp(pdata->retune_mobile_cfgs[i].name,
wm8904->retune_mobile_texts[cfg]) == 0 &&
abs(pdata->retune_mobile_cfgs[i].rate
- wm8904->fs) < best_val) {
best = i;
best_val = abs(pdata->retune_mobile_cfgs[i].rate
- wm8904->fs);
}
}
dev_dbg(codec->dev, "ReTune Mobile %s/%dHz for %dHz sample rate\n",
pdata->retune_mobile_cfgs[best].name,
pdata->retune_mobile_cfgs[best].rate,
wm8904->fs);
/* The EQ will be disabled while reconfiguring it, remember the
* current configuration.
*/
save = snd_soc_read(codec, WM8904_EQ1);
for (i = 0; i < WM8904_EQ_REGS; i++)
snd_soc_update_bits(codec, WM8904_EQ1 + i, 0xffff,
pdata->retune_mobile_cfgs[best].regs[i]);
snd_soc_update_bits(codec, WM8904_EQ1, WM8904_EQ_ENA, save);
}
static int wm8904_put_retune_mobile_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
struct wm8904_pdata *pdata = wm8904->pdata;
int value = ucontrol->value.integer.value[0];
if (value >= pdata->num_retune_mobile_cfgs)
return -EINVAL;
wm8904->retune_mobile_cfg = value;
wm8904_set_retune_mobile(codec);
return 0;
}
static int wm8904_get_retune_mobile_enum(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
ucontrol->value.enumerated.item[0] = wm8904->retune_mobile_cfg;
return 0;
}
static int deemph_settings[] = { 0, 32000, 44100, 48000 };
static int wm8904_set_deemph(struct snd_soc_codec *codec)
{
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
int val, i, best;
/* If we're using deemphasis select the nearest available sample
* rate.
*/
if (wm8904->deemph) {
best = 1;
for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) {
if (abs(deemph_settings[i] - wm8904->fs) <
abs(deemph_settings[best] - wm8904->fs))
best = i;
}
val = best << WM8904_DEEMPH_SHIFT;
} else {
val = 0;
}
dev_dbg(codec->dev, "Set deemphasis %d\n", val);
return snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_1,
WM8904_DEEMPH_MASK, val);
}
static int wm8904_get_deemph(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
ucontrol->value.enumerated.item[0] = wm8904->deemph;
return 0;
}
static int wm8904_put_deemph(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
int deemph = ucontrol->value.enumerated.item[0];
if (deemph > 1)
return -EINVAL;
wm8904->deemph = deemph;
return wm8904_set_deemph(codec);
}
static const DECLARE_TLV_DB_SCALE(dac_boost_tlv, 0, 600, 0);
static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1);
static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0);
static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -3600, 300, 0);
static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
static const char *input_mode_text[] = {
"Single-Ended", "Differential Line", "Differential Mic"
};
static const struct soc_enum lin_mode =
SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 0, 3, input_mode_text);
static const struct soc_enum rin_mode =
SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 0, 3, input_mode_text);
static const char *hpf_mode_text[] = {
"Hi-fi", "Voice 1", "Voice 2", "Voice 3"
};
static const struct soc_enum hpf_mode =
SOC_ENUM_SINGLE(WM8904_ADC_DIGITAL_0, 5, 4, hpf_mode_text);
static int wm8904_adc_osr_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
unsigned int val;
int ret;
ret = snd_soc_put_volsw(kcontrol, ucontrol);
if (ret < 0)
return ret;
if (ucontrol->value.integer.value[0])
val = 0;
else
val = WM8904_ADC_128_OSR_TST_MODE | WM8904_ADC_BIASX1P5;
snd_soc_update_bits(codec, WM8904_ADC_TEST_0,
WM8904_ADC_128_OSR_TST_MODE | WM8904_ADC_BIASX1P5,
val);
return ret;
}
static const struct snd_kcontrol_new wm8904_adc_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8904_ADC_DIGITAL_VOLUME_LEFT,
WM8904_ADC_DIGITAL_VOLUME_RIGHT, 1, 119, 0, digital_tlv),
SOC_ENUM("Left Caputure Mode", lin_mode),
SOC_ENUM("Right Capture Mode", rin_mode),
/* No TLV since it depends on mode */
SOC_DOUBLE_R("Capture Volume", WM8904_ANALOGUE_LEFT_INPUT_0,
WM8904_ANALOGUE_RIGHT_INPUT_0, 0, 31, 0),
SOC_DOUBLE_R("Capture Switch", WM8904_ANALOGUE_LEFT_INPUT_0,
WM8904_ANALOGUE_RIGHT_INPUT_0, 7, 1, 1),
SOC_SINGLE("High Pass Filter Switch", WM8904_ADC_DIGITAL_0, 4, 1, 0),
SOC_ENUM("High Pass Filter Mode", hpf_mode),
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "ADC 128x OSR Switch",
.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,
.put = wm8904_adc_osr_put,
.private_value = SOC_SINGLE_VALUE(WM8904_ANALOGUE_ADC_0, 0, 1, 0),
},
};
static const char *drc_path_text[] = {
"ADC", "DAC"
};
static const struct soc_enum drc_path =
SOC_ENUM_SINGLE(WM8904_DRC_0, 14, 2, drc_path_text);
static const struct snd_kcontrol_new wm8904_dac_snd_controls[] = {
SOC_SINGLE_TLV("Digital Playback Boost Volume",
WM8904_AUDIO_INTERFACE_0, 9, 3, 0, dac_boost_tlv),
SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8904_DAC_DIGITAL_VOLUME_LEFT,
WM8904_DAC_DIGITAL_VOLUME_RIGHT, 1, 96, 0, digital_tlv),
SOC_DOUBLE_R_TLV("Headphone Volume", WM8904_ANALOGUE_OUT1_LEFT,
WM8904_ANALOGUE_OUT1_RIGHT, 0, 63, 0, out_tlv),
SOC_DOUBLE_R("Headphone Switch", WM8904_ANALOGUE_OUT1_LEFT,
WM8904_ANALOGUE_OUT1_RIGHT, 8, 1, 1),
SOC_DOUBLE_R("Headphone ZC Switch", WM8904_ANALOGUE_OUT1_LEFT,
WM8904_ANALOGUE_OUT1_RIGHT, 6, 1, 0),
SOC_DOUBLE_R_TLV("Line Output Volume", WM8904_ANALOGUE_OUT2_LEFT,
WM8904_ANALOGUE_OUT2_RIGHT, 0, 63, 0, out_tlv),
SOC_DOUBLE_R("Line Output Switch", WM8904_ANALOGUE_OUT2_LEFT,
WM8904_ANALOGUE_OUT2_RIGHT, 8, 1, 1),
SOC_DOUBLE_R("Line Output ZC Switch", WM8904_ANALOGUE_OUT2_LEFT,
WM8904_ANALOGUE_OUT2_RIGHT, 6, 1, 0),
SOC_SINGLE("EQ Switch", WM8904_EQ1, 0, 1, 0),
SOC_SINGLE("DRC Switch", WM8904_DRC_0, 15, 1, 0),
SOC_ENUM("DRC Path", drc_path),
SOC_SINGLE("DAC OSRx2 Switch", WM8904_DAC_DIGITAL_1, 6, 1, 0),
SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0,
wm8904_get_deemph, wm8904_put_deemph),
};
static const struct snd_kcontrol_new wm8904_snd_controls[] = {
SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8904_DAC_DIGITAL_0, 4, 8, 15, 0,
sidetone_tlv),
};
static const struct snd_kcontrol_new wm8904_eq_controls[] = {
SOC_SINGLE_TLV("EQ1 Volume", WM8904_EQ2, 0, 24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ2 Volume", WM8904_EQ3, 0, 24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ3 Volume", WM8904_EQ4, 0, 24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ4 Volume", WM8904_EQ5, 0, 24, 0, eq_tlv),
SOC_SINGLE_TLV("EQ5 Volume", WM8904_EQ6, 0, 24, 0, eq_tlv),
};
static int cp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
BUG_ON(event != SND_SOC_DAPM_POST_PMU);
/* Maximum startup time */
udelay(500);
return 0;
}
static int sysclk_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
/* If we're using the FLL then we only start it when
* required; we assume that the configuration has been
* done previously and all we need to do is kick it
* off.
*/
switch (wm8904->sysclk_src) {
case WM8904_CLK_FLL:
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1,
WM8904_FLL_OSC_ENA,
WM8904_FLL_OSC_ENA);
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1,
WM8904_FLL_ENA,
WM8904_FLL_ENA);
break;
default:
break;
}
break;
case SND_SOC_DAPM_POST_PMD:
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1,
WM8904_FLL_OSC_ENA | WM8904_FLL_ENA, 0);
break;
}
return 0;
}
static int out_pga_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
int reg, val;
int dcs_mask;
int dcs_l, dcs_r;
int dcs_l_reg, dcs_r_reg;
int timeout;
int pwr_reg;
/* This code is shared between HP and LINEOUT; we do all our
* power management in stereo pairs to avoid latency issues so
* we reuse shift to identify which rather than strcmp() the
* name. */
reg = w->shift;
switch (reg) {
case WM8904_ANALOGUE_HP_0:
pwr_reg = WM8904_POWER_MANAGEMENT_2;
dcs_mask = WM8904_DCS_ENA_CHAN_0 | WM8904_DCS_ENA_CHAN_1;
dcs_r_reg = WM8904_DC_SERVO_8;
dcs_l_reg = WM8904_DC_SERVO_9;
dcs_l = 0;
dcs_r = 1;
break;
case WM8904_ANALOGUE_LINEOUT_0:
pwr_reg = WM8904_POWER_MANAGEMENT_3;
dcs_mask = WM8904_DCS_ENA_CHAN_2 | WM8904_DCS_ENA_CHAN_3;
dcs_r_reg = WM8904_DC_SERVO_6;
dcs_l_reg = WM8904_DC_SERVO_7;
dcs_l = 2;
dcs_r = 3;
break;
default:
BUG();
return -EINVAL;
}
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
/* Power on the PGAs */
snd_soc_update_bits(codec, pwr_reg,
WM8904_HPL_PGA_ENA | WM8904_HPR_PGA_ENA,
WM8904_HPL_PGA_ENA | WM8904_HPR_PGA_ENA);
/* Power on the amplifier */
snd_soc_update_bits(codec, reg,
WM8904_HPL_ENA | WM8904_HPR_ENA,
WM8904_HPL_ENA | WM8904_HPR_ENA);
/* Enable the first stage */
snd_soc_update_bits(codec, reg,
WM8904_HPL_ENA_DLY | WM8904_HPR_ENA_DLY,
WM8904_HPL_ENA_DLY | WM8904_HPR_ENA_DLY);
/* Power up the DC servo */
snd_soc_update_bits(codec, WM8904_DC_SERVO_0,
dcs_mask, dcs_mask);
/* Either calibrate the DC servo or restore cached state
* if we have that.
*/
if (wm8904->dcs_state[dcs_l] || wm8904->dcs_state[dcs_r]) {
dev_dbg(codec->dev, "Restoring DC servo state\n");
snd_soc_write(codec, dcs_l_reg,
wm8904->dcs_state[dcs_l]);
snd_soc_write(codec, dcs_r_reg,
wm8904->dcs_state[dcs_r]);
snd_soc_write(codec, WM8904_DC_SERVO_1, dcs_mask);
timeout = 20;
} else {
dev_dbg(codec->dev, "Calibrating DC servo\n");
snd_soc_write(codec, WM8904_DC_SERVO_1,
dcs_mask << WM8904_DCS_TRIG_STARTUP_0_SHIFT);
timeout = 500;
}
/* Wait for DC servo to complete */
dcs_mask <<= WM8904_DCS_CAL_COMPLETE_SHIFT;
do {
val = snd_soc_read(codec, WM8904_DC_SERVO_READBACK_0);
if ((val & dcs_mask) == dcs_mask)
break;
msleep(1);
} while (--timeout);
if ((val & dcs_mask) != dcs_mask)
dev_warn(codec->dev, "DC servo timed out\n");
else
dev_dbg(codec->dev, "DC servo ready\n");
/* Enable the output stage */
snd_soc_update_bits(codec, reg,
WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP,
WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP);
break;
case SND_SOC_DAPM_POST_PMU:
/* Unshort the output itself */
snd_soc_update_bits(codec, reg,
WM8904_HPL_RMV_SHORT |
WM8904_HPR_RMV_SHORT,
WM8904_HPL_RMV_SHORT |
WM8904_HPR_RMV_SHORT);
break;
case SND_SOC_DAPM_PRE_PMD:
/* Short the output */
snd_soc_update_bits(codec, reg,
WM8904_HPL_RMV_SHORT |
WM8904_HPR_RMV_SHORT, 0);
break;
case SND_SOC_DAPM_POST_PMD:
/* Cache the DC servo configuration; this will be
* invalidated if we change the configuration. */
wm8904->dcs_state[dcs_l] = snd_soc_read(codec, dcs_l_reg);
wm8904->dcs_state[dcs_r] = snd_soc_read(codec, dcs_r_reg);
snd_soc_update_bits(codec, WM8904_DC_SERVO_0,
dcs_mask, 0);
/* Disable the amplifier input and output stages */
snd_soc_update_bits(codec, reg,
WM8904_HPL_ENA | WM8904_HPR_ENA |
WM8904_HPL_ENA_DLY | WM8904_HPR_ENA_DLY |
WM8904_HPL_ENA_OUTP | WM8904_HPR_ENA_OUTP,
0);
/* PGAs too */
snd_soc_update_bits(codec, pwr_reg,
WM8904_HPL_PGA_ENA | WM8904_HPR_PGA_ENA,
0);
break;
}
return 0;
}
static const char *lin_text[] = {
"IN1L", "IN2L", "IN3L"
};
static const struct soc_enum lin_enum =
SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 2, 3, lin_text);
static const struct snd_kcontrol_new lin_mux =
SOC_DAPM_ENUM("Left Capture Mux", lin_enum);
static const struct soc_enum lin_inv_enum =
SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 4, 3, lin_text);
static const struct snd_kcontrol_new lin_inv_mux =
SOC_DAPM_ENUM("Left Capture Inveting Mux", lin_inv_enum);
static const char *rin_text[] = {
"IN1R", "IN2R", "IN3R"
};
static const struct soc_enum rin_enum =
SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 2, 3, rin_text);
static const struct snd_kcontrol_new rin_mux =
SOC_DAPM_ENUM("Right Capture Mux", rin_enum);
static const struct soc_enum rin_inv_enum =
SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 4, 3, rin_text);
static const struct snd_kcontrol_new rin_inv_mux =
SOC_DAPM_ENUM("Right Capture Inveting Mux", rin_inv_enum);
static const char *aif_text[] = {
"Left", "Right"
};
static const struct soc_enum aifoutl_enum =
SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 7, 2, aif_text);
static const struct snd_kcontrol_new aifoutl_mux =
SOC_DAPM_ENUM("AIFOUTL Mux", aifoutl_enum);
static const struct soc_enum aifoutr_enum =
SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 6, 2, aif_text);
static const struct snd_kcontrol_new aifoutr_mux =
SOC_DAPM_ENUM("AIFOUTR Mux", aifoutr_enum);
static const struct soc_enum aifinl_enum =
SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 5, 2, aif_text);
static const struct snd_kcontrol_new aifinl_mux =
SOC_DAPM_ENUM("AIFINL Mux", aifinl_enum);
static const struct soc_enum aifinr_enum =
SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 4, 2, aif_text);
static const struct snd_kcontrol_new aifinr_mux =
SOC_DAPM_ENUM("AIFINR Mux", aifinr_enum);
static const struct snd_soc_dapm_widget wm8904_core_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("SYSCLK", WM8904_CLOCK_RATES_2, 2, 0, sysclk_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8904_CLOCK_RATES_2, 1, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("TOCLK", WM8904_CLOCK_RATES_2, 0, 0, NULL, 0),
};
static const struct snd_soc_dapm_widget wm8904_adc_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("IN1L"),
SND_SOC_DAPM_INPUT("IN1R"),
SND_SOC_DAPM_INPUT("IN2L"),
SND_SOC_DAPM_INPUT("IN2R"),
SND_SOC_DAPM_INPUT("IN3L"),
SND_SOC_DAPM_INPUT("IN3R"),
SND_SOC_DAPM_SUPPLY("MICBIAS", WM8904_MIC_BIAS_CONTROL_0, 0, 0, NULL, 0),
SND_SOC_DAPM_MUX("Left Capture Mux", SND_SOC_NOPM, 0, 0, &lin_mux),
SND_SOC_DAPM_MUX("Left Capture Inverting Mux", SND_SOC_NOPM, 0, 0,
&lin_inv_mux),
SND_SOC_DAPM_MUX("Right Capture Mux", SND_SOC_NOPM, 0, 0, &rin_mux),
SND_SOC_DAPM_MUX("Right Capture Inverting Mux", SND_SOC_NOPM, 0, 0,
&rin_inv_mux),
SND_SOC_DAPM_PGA("Left Capture PGA", WM8904_POWER_MANAGEMENT_0, 1, 0,
NULL, 0),
SND_SOC_DAPM_PGA("Right Capture PGA", WM8904_POWER_MANAGEMENT_0, 0, 0,
NULL, 0),
SND_SOC_DAPM_ADC("ADCL", NULL, WM8904_POWER_MANAGEMENT_6, 1, 0),
SND_SOC_DAPM_ADC("ADCR", NULL, WM8904_POWER_MANAGEMENT_6, 0, 0),
SND_SOC_DAPM_MUX("AIFOUTL Mux", SND_SOC_NOPM, 0, 0, &aifoutl_mux),
SND_SOC_DAPM_MUX("AIFOUTR Mux", SND_SOC_NOPM, 0, 0, &aifoutr_mux),
SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0),
};
static const struct snd_soc_dapm_widget wm8904_dac_dapm_widgets[] = {
SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &aifinl_mux),
SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &aifinr_mux),
SND_SOC_DAPM_DAC("DACL", NULL, WM8904_POWER_MANAGEMENT_6, 3, 0),
SND_SOC_DAPM_DAC("DACR", NULL, WM8904_POWER_MANAGEMENT_6, 2, 0),
SND_SOC_DAPM_SUPPLY("Charge pump", WM8904_CHARGE_PUMP_0, 0, 0, cp_event,
SND_SOC_DAPM_POST_PMU),
SND_SOC_DAPM_PGA("HPL PGA", SND_SOC_NOPM, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("HPR PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("LINEL PGA", SND_SOC_NOPM, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("LINER PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA_E("Headphone Output", SND_SOC_NOPM, WM8904_ANALOGUE_HP_0,
0, NULL, 0, out_pga_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_PGA_E("Line Output", SND_SOC_NOPM, WM8904_ANALOGUE_LINEOUT_0,
0, NULL, 0, out_pga_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_OUTPUT("HPOUTL"),
SND_SOC_DAPM_OUTPUT("HPOUTR"),
SND_SOC_DAPM_OUTPUT("LINEOUTL"),
SND_SOC_DAPM_OUTPUT("LINEOUTR"),
};
static const char *out_mux_text[] = {
"DAC", "Bypass"
};
static const struct soc_enum hpl_enum =
SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 3, 2, out_mux_text);
static const struct snd_kcontrol_new hpl_mux =
SOC_DAPM_ENUM("HPL Mux", hpl_enum);
static const struct soc_enum hpr_enum =
SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 2, 2, out_mux_text);
static const struct snd_kcontrol_new hpr_mux =
SOC_DAPM_ENUM("HPR Mux", hpr_enum);
static const struct soc_enum linel_enum =
SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 1, 2, out_mux_text);
static const struct snd_kcontrol_new linel_mux =
SOC_DAPM_ENUM("LINEL Mux", linel_enum);
static const struct soc_enum liner_enum =
SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text);
static const struct snd_kcontrol_new liner_mux =
SOC_DAPM_ENUM("LINEL Mux", liner_enum);
static const char *sidetone_text[] = {
"None", "Left", "Right"
};
static const struct soc_enum dacl_sidetone_enum =
SOC_ENUM_SINGLE(WM8904_DAC_DIGITAL_0, 2, 3, sidetone_text);
static const struct snd_kcontrol_new dacl_sidetone_mux =
SOC_DAPM_ENUM("Left Sidetone Mux", dacl_sidetone_enum);
static const struct soc_enum dacr_sidetone_enum =
SOC_ENUM_SINGLE(WM8904_DAC_DIGITAL_0, 0, 3, sidetone_text);
static const struct snd_kcontrol_new dacr_sidetone_mux =
SOC_DAPM_ENUM("Right Sidetone Mux", dacr_sidetone_enum);
static const struct snd_soc_dapm_widget wm8904_dapm_widgets[] = {
SND_SOC_DAPM_SUPPLY("Class G", WM8904_CLASS_W_0, 0, 1, NULL, 0),
SND_SOC_DAPM_PGA("Left Bypass", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Bypass", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MUX("Left Sidetone", SND_SOC_NOPM, 0, 0, &dacl_sidetone_mux),
SND_SOC_DAPM_MUX("Right Sidetone", SND_SOC_NOPM, 0, 0, &dacr_sidetone_mux),
SND_SOC_DAPM_MUX("HPL Mux", SND_SOC_NOPM, 0, 0, &hpl_mux),
SND_SOC_DAPM_MUX("HPR Mux", SND_SOC_NOPM, 0, 0, &hpr_mux),
SND_SOC_DAPM_MUX("LINEL Mux", SND_SOC_NOPM, 0, 0, &linel_mux),
SND_SOC_DAPM_MUX("LINER Mux", SND_SOC_NOPM, 0, 0, &liner_mux),
};
static const struct snd_soc_dapm_route core_intercon[] = {
{ "CLK_DSP", NULL, "SYSCLK" },
{ "TOCLK", NULL, "SYSCLK" },
};
static const struct snd_soc_dapm_route adc_intercon[] = {
{ "Left Capture Mux", "IN1L", "IN1L" },
{ "Left Capture Mux", "IN2L", "IN2L" },
{ "Left Capture Mux", "IN3L", "IN3L" },
{ "Left Capture Inverting Mux", "IN1L", "IN1L" },
{ "Left Capture Inverting Mux", "IN2L", "IN2L" },
{ "Left Capture Inverting Mux", "IN3L", "IN3L" },
{ "Right Capture Mux", "IN1R", "IN1R" },
{ "Right Capture Mux", "IN2R", "IN2R" },
{ "Right Capture Mux", "IN3R", "IN3R" },
{ "Right Capture Inverting Mux", "IN1R", "IN1R" },
{ "Right Capture Inverting Mux", "IN2R", "IN2R" },
{ "Right Capture Inverting Mux", "IN3R", "IN3R" },
{ "Left Capture PGA", NULL, "Left Capture Mux" },
{ "Left Capture PGA", NULL, "Left Capture Inverting Mux" },
{ "Right Capture PGA", NULL, "Right Capture Mux" },
{ "Right Capture PGA", NULL, "Right Capture Inverting Mux" },
{ "AIFOUTL", "Left", "ADCL" },
{ "AIFOUTL", "Right", "ADCR" },
{ "AIFOUTR", "Left", "ADCL" },
{ "AIFOUTR", "Right", "ADCR" },
{ "ADCL", NULL, "CLK_DSP" },
{ "ADCL", NULL, "Left Capture PGA" },
{ "ADCR", NULL, "CLK_DSP" },
{ "ADCR", NULL, "Right Capture PGA" },
};
static const struct snd_soc_dapm_route dac_intercon[] = {
{ "DACL", "Right", "AIFINR" },
{ "DACL", "Left", "AIFINL" },
{ "DACL", NULL, "CLK_DSP" },
{ "DACR", "Right", "AIFINR" },
{ "DACR", "Left", "AIFINL" },
{ "DACR", NULL, "CLK_DSP" },
{ "Charge pump", NULL, "SYSCLK" },
{ "Headphone Output", NULL, "HPL PGA" },
{ "Headphone Output", NULL, "HPR PGA" },
{ "Headphone Output", NULL, "Charge pump" },
{ "Headphone Output", NULL, "TOCLK" },
{ "Line Output", NULL, "LINEL PGA" },
{ "Line Output", NULL, "LINER PGA" },
{ "Line Output", NULL, "Charge pump" },
{ "Line Output", NULL, "TOCLK" },
{ "HPOUTL", NULL, "Headphone Output" },
{ "HPOUTR", NULL, "Headphone Output" },
{ "LINEOUTL", NULL, "Line Output" },
{ "LINEOUTR", NULL, "Line Output" },
};
static const struct snd_soc_dapm_route wm8904_intercon[] = {
{ "Left Sidetone", "Left", "ADCL" },
{ "Left Sidetone", "Right", "ADCR" },
{ "DACL", NULL, "Left Sidetone" },
{ "Right Sidetone", "Left", "ADCL" },
{ "Right Sidetone", "Right", "ADCR" },
{ "DACR", NULL, "Right Sidetone" },
{ "Left Bypass", NULL, "Class G" },
{ "Left Bypass", NULL, "Left Capture PGA" },
{ "Right Bypass", NULL, "Class G" },
{ "Right Bypass", NULL, "Right Capture PGA" },
{ "HPL Mux", "DAC", "DACL" },
{ "HPL Mux", "Bypass", "Left Bypass" },
{ "HPR Mux", "DAC", "DACR" },
{ "HPR Mux", "Bypass", "Right Bypass" },
{ "LINEL Mux", "DAC", "DACL" },
{ "LINEL Mux", "Bypass", "Left Bypass" },
{ "LINER Mux", "DAC", "DACR" },
{ "LINER Mux", "Bypass", "Right Bypass" },
{ "HPL PGA", NULL, "HPL Mux" },
{ "HPR PGA", NULL, "HPR Mux" },
{ "LINEL PGA", NULL, "LINEL Mux" },
{ "LINER PGA", NULL, "LINER Mux" },
};
static const struct snd_soc_dapm_route wm8912_intercon[] = {
{ "HPL PGA", NULL, "DACL" },
{ "HPR PGA", NULL, "DACR" },
{ "LINEL PGA", NULL, "DACL" },
{ "LINER PGA", NULL, "DACR" },
};
static int wm8904_add_widgets(struct snd_soc_codec *codec)
{
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 13:53:46 +00:00
struct snd_soc_dapm_context *dapm = &codec->dapm;
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 13:53:46 +00:00
snd_soc_dapm_new_controls(dapm, wm8904_core_dapm_widgets,
ARRAY_SIZE(wm8904_core_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 13:53:46 +00:00
snd_soc_dapm_add_routes(dapm, core_intercon,
ARRAY_SIZE(core_intercon));
switch (wm8904->devtype) {
case WM8904:
snd_soc_add_codec_controls(codec, wm8904_adc_snd_controls,
ARRAY_SIZE(wm8904_adc_snd_controls));
snd_soc_add_codec_controls(codec, wm8904_dac_snd_controls,
ARRAY_SIZE(wm8904_dac_snd_controls));
snd_soc_add_codec_controls(codec, wm8904_snd_controls,
ARRAY_SIZE(wm8904_snd_controls));
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 13:53:46 +00:00
snd_soc_dapm_new_controls(dapm, wm8904_adc_dapm_widgets,
ARRAY_SIZE(wm8904_adc_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 13:53:46 +00:00
snd_soc_dapm_new_controls(dapm, wm8904_dac_dapm_widgets,
ARRAY_SIZE(wm8904_dac_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 13:53:46 +00:00
snd_soc_dapm_new_controls(dapm, wm8904_dapm_widgets,
ARRAY_SIZE(wm8904_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 13:53:46 +00:00
snd_soc_dapm_add_routes(dapm, adc_intercon,
ARRAY_SIZE(adc_intercon));
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 13:53:46 +00:00
snd_soc_dapm_add_routes(dapm, dac_intercon,
ARRAY_SIZE(dac_intercon));
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 13:53:46 +00:00
snd_soc_dapm_add_routes(dapm, wm8904_intercon,
ARRAY_SIZE(wm8904_intercon));
break;
case WM8912:
snd_soc_add_codec_controls(codec, wm8904_dac_snd_controls,
ARRAY_SIZE(wm8904_dac_snd_controls));
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 13:53:46 +00:00
snd_soc_dapm_new_controls(dapm, wm8904_dac_dapm_widgets,
ARRAY_SIZE(wm8904_dac_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 13:53:46 +00:00
snd_soc_dapm_add_routes(dapm, dac_intercon,
ARRAY_SIZE(dac_intercon));
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 13:53:46 +00:00
snd_soc_dapm_add_routes(dapm, wm8912_intercon,
ARRAY_SIZE(wm8912_intercon));
break;
}
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 13:53:46 +00:00
snd_soc_dapm_new_widgets(dapm);
return 0;
}
static struct {
int ratio;
unsigned int clk_sys_rate;
} clk_sys_rates[] = {
{ 64, 0 },
{ 128, 1 },
{ 192, 2 },
{ 256, 3 },
{ 384, 4 },
{ 512, 5 },
{ 786, 6 },
{ 1024, 7 },
{ 1408, 8 },
{ 1536, 9 },
};
static struct {
int rate;
int sample_rate;
} sample_rates[] = {
{ 8000, 0 },
{ 11025, 1 },
{ 12000, 1 },
{ 16000, 2 },
{ 22050, 3 },
{ 24000, 3 },
{ 32000, 4 },
{ 44100, 5 },
{ 48000, 5 },
};
static struct {
int div; /* *10 due to .5s */
int bclk_div;
} bclk_divs[] = {
{ 10, 0 },
{ 15, 1 },
{ 20, 2 },
{ 30, 3 },
{ 40, 4 },
{ 50, 5 },
{ 55, 6 },
{ 60, 7 },
{ 80, 8 },
{ 100, 9 },
{ 110, 10 },
{ 120, 11 },
{ 160, 12 },
{ 200, 13 },
{ 220, 14 },
{ 240, 16 },
{ 200, 17 },
{ 320, 18 },
{ 440, 19 },
{ 480, 20 },
};
static int wm8904_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
int ret, i, best, best_val, cur_val;
unsigned int aif1 = 0;
unsigned int aif2 = 0;
unsigned int aif3 = 0;
unsigned int clock1 = 0;
unsigned int dac_digital1 = 0;
/* What BCLK do we need? */
wm8904->fs = params_rate(params);
if (wm8904->tdm_slots) {
dev_dbg(codec->dev, "Configuring for %d %d bit TDM slots\n",
wm8904->tdm_slots, wm8904->tdm_width);
wm8904->bclk = snd_soc_calc_bclk(wm8904->fs,
wm8904->tdm_width, 2,
wm8904->tdm_slots);
} else {
wm8904->bclk = snd_soc_params_to_bclk(params);
}
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
aif1 |= 0x40;
break;
case SNDRV_PCM_FORMAT_S24_LE:
aif1 |= 0x80;
break;
case SNDRV_PCM_FORMAT_S32_LE:
aif1 |= 0xc0;
break;
default:
return -EINVAL;
}
dev_dbg(codec->dev, "Target BCLK is %dHz\n", wm8904->bclk);
ret = wm8904_configure_clocking(codec);
if (ret != 0)
return ret;
/* Select nearest CLK_SYS_RATE */
best = 0;
best_val = abs((wm8904->sysclk_rate / clk_sys_rates[0].ratio)
- wm8904->fs);
for (i = 1; i < ARRAY_SIZE(clk_sys_rates); i++) {
cur_val = abs((wm8904->sysclk_rate /
clk_sys_rates[i].ratio) - wm8904->fs);
if (cur_val < best_val) {
best = i;
best_val = cur_val;
}
}
dev_dbg(codec->dev, "Selected CLK_SYS_RATIO of %d\n",
clk_sys_rates[best].ratio);
clock1 |= (clk_sys_rates[best].clk_sys_rate
<< WM8904_CLK_SYS_RATE_SHIFT);
/* SAMPLE_RATE */
best = 0;
best_val = abs(wm8904->fs - sample_rates[0].rate);
for (i = 1; i < ARRAY_SIZE(sample_rates); i++) {
/* Closest match */
cur_val = abs(wm8904->fs - sample_rates[i].rate);
if (cur_val < best_val) {
best = i;
best_val = cur_val;
}
}
dev_dbg(codec->dev, "Selected SAMPLE_RATE of %dHz\n",
sample_rates[best].rate);
clock1 |= (sample_rates[best].sample_rate
<< WM8904_SAMPLE_RATE_SHIFT);
/* Enable sloping stopband filter for low sample rates */
if (wm8904->fs <= 24000)
dac_digital1 |= WM8904_DAC_SB_FILT;
/* BCLK_DIV */
best = 0;
best_val = INT_MAX;
for (i = 0; i < ARRAY_SIZE(bclk_divs); i++) {
cur_val = ((wm8904->sysclk_rate * 10) / bclk_divs[i].div)
- wm8904->bclk;
if (cur_val < 0) /* Table is sorted */
break;
if (cur_val < best_val) {
best = i;
best_val = cur_val;
}
}
wm8904->bclk = (wm8904->sysclk_rate * 10) / bclk_divs[best].div;
dev_dbg(codec->dev, "Selected BCLK_DIV of %d for %dHz BCLK\n",
bclk_divs[best].div, wm8904->bclk);
aif2 |= bclk_divs[best].bclk_div;
/* LRCLK is a simple fraction of BCLK */
dev_dbg(codec->dev, "LRCLK_RATE is %d\n", wm8904->bclk / wm8904->fs);
aif3 |= wm8904->bclk / wm8904->fs;
/* Apply the settings */
snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_1,
WM8904_DAC_SB_FILT, dac_digital1);
snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_1,
WM8904_AIF_WL_MASK, aif1);
snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_2,
WM8904_BCLK_DIV_MASK, aif2);
snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_3,
WM8904_LRCLK_RATE_MASK, aif3);
snd_soc_update_bits(codec, WM8904_CLOCK_RATES_1,
WM8904_SAMPLE_RATE_MASK |
WM8904_CLK_SYS_RATE_MASK, clock1);
/* Update filters for the new settings */
wm8904_set_retune_mobile(codec);
wm8904_set_deemph(codec);
return 0;
}
static int wm8904_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
struct snd_soc_codec *codec = dai->codec;
struct wm8904_priv *priv = snd_soc_codec_get_drvdata(codec);
switch (clk_id) {
case WM8904_CLK_MCLK:
priv->sysclk_src = clk_id;
priv->mclk_rate = freq;
break;
case WM8904_CLK_FLL:
priv->sysclk_src = clk_id;
break;
default:
return -EINVAL;
}
dev_dbg(dai->dev, "Clock source is %d at %uHz\n", clk_id, freq);
wm8904_configure_clocking(codec);
return 0;
}
static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct snd_soc_codec *codec = dai->codec;
unsigned int aif1 = 0;
unsigned int aif3 = 0;
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
break;
case SND_SOC_DAIFMT_CBS_CFM:
aif3 |= WM8904_LRCLK_DIR;
break;
case SND_SOC_DAIFMT_CBM_CFS:
aif1 |= WM8904_BCLK_DIR;
break;
case SND_SOC_DAIFMT_CBM_CFM:
aif1 |= WM8904_BCLK_DIR;
aif3 |= WM8904_LRCLK_DIR;
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_B:
aif1 |= WM8904_AIF_LRCLK_INV;
case SND_SOC_DAIFMT_DSP_A:
aif1 |= 0x3;
break;
case SND_SOC_DAIFMT_I2S:
aif1 |= 0x2;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
aif1 |= 0x1;
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_A:
case SND_SOC_DAIFMT_DSP_B:
/* frame inversion not valid for DSP modes */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_IB_NF:
aif1 |= WM8904_AIF_BCLK_INV;
break;
default:
return -EINVAL;
}
break;
case SND_SOC_DAIFMT_I2S:
case SND_SOC_DAIFMT_RIGHT_J:
case SND_SOC_DAIFMT_LEFT_J:
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_IB_IF:
aif1 |= WM8904_AIF_BCLK_INV | WM8904_AIF_LRCLK_INV;
break;
case SND_SOC_DAIFMT_IB_NF:
aif1 |= WM8904_AIF_BCLK_INV;
break;
case SND_SOC_DAIFMT_NB_IF:
aif1 |= WM8904_AIF_LRCLK_INV;
break;
default:
return -EINVAL;
}
break;
default:
return -EINVAL;
}
snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_1,
WM8904_AIF_BCLK_INV | WM8904_AIF_LRCLK_INV |
WM8904_AIF_FMT_MASK | WM8904_BCLK_DIR, aif1);
snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_3,
WM8904_LRCLK_DIR, aif3);
return 0;
}
static int wm8904_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
unsigned int rx_mask, int slots, int slot_width)
{
struct snd_soc_codec *codec = dai->codec;
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
int aif1 = 0;
/* Don't need to validate anything if we're turning off TDM */
if (slots == 0)
goto out;
/* Note that we allow configurations we can't handle ourselves -
* for example, we can generate clocks for slots 2 and up even if
* we can't use those slots ourselves.
*/
aif1 |= WM8904_AIFADC_TDM | WM8904_AIFDAC_TDM;
switch (rx_mask) {
case 3:
break;
case 0xc:
aif1 |= WM8904_AIFADC_TDM_CHAN;
break;
default:
return -EINVAL;
}
switch (tx_mask) {
case 3:
break;
case 0xc:
aif1 |= WM8904_AIFDAC_TDM_CHAN;
break;
default:
return -EINVAL;
}
out:
wm8904->tdm_width = slot_width;
wm8904->tdm_slots = slots / 2;
snd_soc_update_bits(codec, WM8904_AUDIO_INTERFACE_1,
WM8904_AIFADC_TDM | WM8904_AIFADC_TDM_CHAN |
WM8904_AIFDAC_TDM | WM8904_AIFDAC_TDM_CHAN, aif1);
return 0;
}
struct _fll_div {
u16 fll_fratio;
u16 fll_outdiv;
u16 fll_clk_ref_div;
u16 n;
u16 k;
};
/* The size in bits of the FLL divide multiplied by 10
* to allow rounding later */
#define FIXED_FLL_SIZE ((1 << 16) * 10)
static struct {
unsigned int min;
unsigned int max;
u16 fll_fratio;
int ratio;
} fll_fratios[] = {
{ 0, 64000, 4, 16 },
{ 64000, 128000, 3, 8 },
{ 128000, 256000, 2, 4 },
{ 256000, 1000000, 1, 2 },
{ 1000000, 13500000, 0, 1 },
};
static int fll_factors(struct _fll_div *fll_div, unsigned int Fref,
unsigned int Fout)
{
u64 Kpart;
unsigned int K, Ndiv, Nmod, target;
unsigned int div;
int i;
/* Fref must be <=13.5MHz */
div = 1;
fll_div->fll_clk_ref_div = 0;
while ((Fref / div) > 13500000) {
div *= 2;
fll_div->fll_clk_ref_div++;
if (div > 8) {
pr_err("Can't scale %dMHz input down to <=13.5MHz\n",
Fref);
return -EINVAL;
}
}
pr_debug("Fref=%u Fout=%u\n", Fref, Fout);
/* Apply the division for our remaining calculations */
Fref /= div;
/* Fvco should be 90-100MHz; don't check the upper bound */
div = 4;
while (Fout * div < 90000000) {
div++;
if (div > 64) {
pr_err("Unable to find FLL_OUTDIV for Fout=%uHz\n",
Fout);
return -EINVAL;
}
}
target = Fout * div;
fll_div->fll_outdiv = div - 1;
pr_debug("Fvco=%dHz\n", target);
/* Find an appropriate FLL_FRATIO and factor it out of the target */
for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
fll_div->fll_fratio = fll_fratios[i].fll_fratio;
target /= fll_fratios[i].ratio;
break;
}
}
if (i == ARRAY_SIZE(fll_fratios)) {
pr_err("Unable to find FLL_FRATIO for Fref=%uHz\n", Fref);
return -EINVAL;
}
/* Now, calculate N.K */
Ndiv = target / Fref;
fll_div->n = Ndiv;
Nmod = target % Fref;
pr_debug("Nmod=%d\n", Nmod);
/* Calculate fractional part - scale up so we can round. */
Kpart = FIXED_FLL_SIZE * (long long)Nmod;
do_div(Kpart, Fref);
K = Kpart & 0xFFFFFFFF;
if ((K % 10) >= 5)
K += 5;
/* Move down to proper range now rounding is done */
fll_div->k = K / 10;
pr_debug("N=%x K=%x FLL_FRATIO=%x FLL_OUTDIV=%x FLL_CLK_REF_DIV=%x\n",
fll_div->n, fll_div->k,
fll_div->fll_fratio, fll_div->fll_outdiv,
fll_div->fll_clk_ref_div);
return 0;
}
static int wm8904_set_fll(struct snd_soc_dai *dai, int fll_id, int source,
unsigned int Fref, unsigned int Fout)
{
struct snd_soc_codec *codec = dai->codec;
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
struct _fll_div fll_div;
int ret, val;
int clock2, fll1;
/* Any change? */
if (source == wm8904->fll_src && Fref == wm8904->fll_fref &&
Fout == wm8904->fll_fout)
return 0;
clock2 = snd_soc_read(codec, WM8904_CLOCK_RATES_2);
if (Fout == 0) {
dev_dbg(codec->dev, "FLL disabled\n");
wm8904->fll_fref = 0;
wm8904->fll_fout = 0;
/* Gate SYSCLK to avoid glitches */
snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2,
WM8904_CLK_SYS_ENA, 0);
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1,
WM8904_FLL_OSC_ENA | WM8904_FLL_ENA, 0);
goto out;
}
/* Validate the FLL ID */
switch (source) {
case WM8904_FLL_MCLK:
case WM8904_FLL_LRCLK:
case WM8904_FLL_BCLK:
ret = fll_factors(&fll_div, Fref, Fout);
if (ret != 0)
return ret;
break;
case WM8904_FLL_FREE_RUNNING:
dev_dbg(codec->dev, "Using free running FLL\n");
/* Force 12MHz and output/4 for now */
Fout = 12000000;
Fref = 12000000;
memset(&fll_div, 0, sizeof(fll_div));
fll_div.fll_outdiv = 3;
break;
default:
dev_err(codec->dev, "Unknown FLL ID %d\n", fll_id);
return -EINVAL;
}
/* Save current state then disable the FLL and SYSCLK to avoid
* misclocking */
fll1 = snd_soc_read(codec, WM8904_FLL_CONTROL_1);
snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2,
WM8904_CLK_SYS_ENA, 0);
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1,
WM8904_FLL_OSC_ENA | WM8904_FLL_ENA, 0);
/* Unlock forced oscilator control to switch it on/off */
snd_soc_update_bits(codec, WM8904_CONTROL_INTERFACE_TEST_1,
WM8904_USER_KEY, WM8904_USER_KEY);
if (fll_id == WM8904_FLL_FREE_RUNNING) {
val = WM8904_FLL_FRC_NCO;
} else {
val = 0;
}
snd_soc_update_bits(codec, WM8904_FLL_NCO_TEST_1, WM8904_FLL_FRC_NCO,
val);
snd_soc_update_bits(codec, WM8904_CONTROL_INTERFACE_TEST_1,
WM8904_USER_KEY, 0);
switch (fll_id) {
case WM8904_FLL_MCLK:
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_5,
WM8904_FLL_CLK_REF_SRC_MASK, 0);
break;
case WM8904_FLL_LRCLK:
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_5,
WM8904_FLL_CLK_REF_SRC_MASK, 1);
break;
case WM8904_FLL_BCLK:
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_5,
WM8904_FLL_CLK_REF_SRC_MASK, 2);
break;
}
if (fll_div.k)
val = WM8904_FLL_FRACN_ENA;
else
val = 0;
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1,
WM8904_FLL_FRACN_ENA, val);
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_2,
WM8904_FLL_OUTDIV_MASK | WM8904_FLL_FRATIO_MASK,
(fll_div.fll_outdiv << WM8904_FLL_OUTDIV_SHIFT) |
(fll_div.fll_fratio << WM8904_FLL_FRATIO_SHIFT));
snd_soc_write(codec, WM8904_FLL_CONTROL_3, fll_div.k);
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_4, WM8904_FLL_N_MASK,
fll_div.n << WM8904_FLL_N_SHIFT);
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_5,
WM8904_FLL_CLK_REF_DIV_MASK,
fll_div.fll_clk_ref_div
<< WM8904_FLL_CLK_REF_DIV_SHIFT);
dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout);
wm8904->fll_fref = Fref;
wm8904->fll_fout = Fout;
wm8904->fll_src = source;
/* Enable the FLL if it was previously active */
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1,
WM8904_FLL_OSC_ENA, fll1);
snd_soc_update_bits(codec, WM8904_FLL_CONTROL_1,
WM8904_FLL_ENA, fll1);
out:
/* Reenable SYSCLK if it was previously active */
snd_soc_update_bits(codec, WM8904_CLOCK_RATES_2,
WM8904_CLK_SYS_ENA, clock2);
return 0;
}
static int wm8904_digital_mute(struct snd_soc_dai *codec_dai, int mute)
{
struct snd_soc_codec *codec = codec_dai->codec;
int val;
if (mute)
val = WM8904_DAC_MUTE;
else
val = 0;
snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_1, WM8904_DAC_MUTE, val);
return 0;
}
static int wm8904_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
int ret;
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
/* VMID resistance 2*50k */
snd_soc_update_bits(codec, WM8904_VMID_CONTROL_0,
WM8904_VMID_RES_MASK,
0x1 << WM8904_VMID_RES_SHIFT);
/* Normal bias current */
snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0,
WM8904_ISEL_MASK, 2 << WM8904_ISEL_SHIFT);
break;
case SND_SOC_BIAS_STANDBY:
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 13:53:46 +00:00
if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies),
wm8904->supplies);
if (ret != 0) {
dev_err(codec->dev,
"Failed to enable supplies: %d\n",
ret);
return ret;
}
regcache_cache_only(wm8904->regmap, false);
regcache_sync(wm8904->regmap);
/* Enable bias */
snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0,
WM8904_BIAS_ENA, WM8904_BIAS_ENA);
/* Enable VMID, VMID buffering, 2*5k resistance */
snd_soc_update_bits(codec, WM8904_VMID_CONTROL_0,
WM8904_VMID_ENA |
WM8904_VMID_RES_MASK,
WM8904_VMID_ENA |
0x3 << WM8904_VMID_RES_SHIFT);
/* Let VMID ramp */
msleep(1);
}
/* Maintain VMID with 2*250k */
snd_soc_update_bits(codec, WM8904_VMID_CONTROL_0,
WM8904_VMID_RES_MASK,
0x2 << WM8904_VMID_RES_SHIFT);
/* Bias current *0.5 */
snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0,
WM8904_ISEL_MASK, 0);
break;
case SND_SOC_BIAS_OFF:
/* Turn off VMID */
snd_soc_update_bits(codec, WM8904_VMID_CONTROL_0,
WM8904_VMID_RES_MASK | WM8904_VMID_ENA, 0);
/* Stop bias generation */
snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0,
WM8904_BIAS_ENA, 0);
regcache_cache_only(wm8904->regmap, true);
regcache_mark_dirty(wm8904->regmap);
regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies),
wm8904->supplies);
break;
}
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 13:53:46 +00:00
codec->dapm.bias_level = level;
return 0;
}
#define WM8904_RATES SNDRV_PCM_RATE_8000_96000
#define WM8904_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
static const struct snd_soc_dai_ops wm8904_dai_ops = {
.set_sysclk = wm8904_set_sysclk,
.set_fmt = wm8904_set_fmt,
.set_tdm_slot = wm8904_set_tdm_slot,
.set_pll = wm8904_set_fll,
.hw_params = wm8904_hw_params,
.digital_mute = wm8904_digital_mute,
};
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
static struct snd_soc_dai_driver wm8904_dai = {
.name = "wm8904-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
.rates = WM8904_RATES,
.formats = WM8904_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = WM8904_RATES,
.formats = WM8904_FORMATS,
},
.ops = &wm8904_dai_ops,
.symmetric_rates = 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 20:15:21 +00:00
static void wm8904_handle_retune_mobile_pdata(struct snd_soc_codec *codec)
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
struct wm8904_pdata *pdata = wm8904->pdata;
struct snd_kcontrol_new control =
SOC_ENUM_EXT("EQ Mode",
wm8904->retune_mobile_enum,
wm8904_get_retune_mobile_enum,
wm8904_put_retune_mobile_enum);
int ret, i, j;
const char **t;
/* We need an array of texts for the enum API but the number
* of texts is likely to be less than the number of
* configurations due to the sample rate dependency of the
* configurations. */
wm8904->num_retune_mobile_texts = 0;
wm8904->retune_mobile_texts = NULL;
for (i = 0; i < pdata->num_retune_mobile_cfgs; i++) {
for (j = 0; j < wm8904->num_retune_mobile_texts; j++) {
if (strcmp(pdata->retune_mobile_cfgs[i].name,
wm8904->retune_mobile_texts[j]) == 0)
break;
}
if (j != wm8904->num_retune_mobile_texts)
continue;
/* Expand the array... */
t = krealloc(wm8904->retune_mobile_texts,
sizeof(char *) *
(wm8904->num_retune_mobile_texts + 1),
GFP_KERNEL);
if (t == NULL)
continue;
/* ...store the new entry... */
t[wm8904->num_retune_mobile_texts] =
pdata->retune_mobile_cfgs[i].name;
/* ...and remember the new version. */
wm8904->num_retune_mobile_texts++;
wm8904->retune_mobile_texts = t;
}
dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n",
wm8904->num_retune_mobile_texts);
wm8904->retune_mobile_enum.max = wm8904->num_retune_mobile_texts;
wm8904->retune_mobile_enum.texts = wm8904->retune_mobile_texts;
ret = snd_soc_add_codec_controls(codec, &control, 1);
if (ret != 0)
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
dev_err(codec->dev,
"Failed to add ReTune Mobile control: %d\n", ret);
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
static void wm8904_handle_pdata(struct snd_soc_codec *codec)
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
struct wm8904_pdata *pdata = wm8904->pdata;
int ret, i;
if (!pdata) {
snd_soc_add_codec_controls(codec, wm8904_eq_controls,
ARRAY_SIZE(wm8904_eq_controls));
return;
}
dev_dbg(codec->dev, "%d DRC configurations\n", pdata->num_drc_cfgs);
if (pdata->num_drc_cfgs) {
struct snd_kcontrol_new control =
SOC_ENUM_EXT("DRC Mode", wm8904->drc_enum,
wm8904_get_drc_enum, wm8904_put_drc_enum);
/* We need an array of texts for the enum API */
wm8904->drc_texts = kmalloc(sizeof(char *)
* pdata->num_drc_cfgs, GFP_KERNEL);
if (!wm8904->drc_texts) {
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
dev_err(codec->dev,
"Failed to allocate %d DRC config texts\n",
pdata->num_drc_cfgs);
return;
}
for (i = 0; i < pdata->num_drc_cfgs; i++)
wm8904->drc_texts[i] = pdata->drc_cfgs[i].name;
wm8904->drc_enum.max = pdata->num_drc_cfgs;
wm8904->drc_enum.texts = wm8904->drc_texts;
ret = snd_soc_add_codec_controls(codec, &control, 1);
if (ret != 0)
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
dev_err(codec->dev,
"Failed to add DRC mode control: %d\n", ret);
wm8904_set_drc(codec);
}
dev_dbg(codec->dev, "%d ReTune Mobile configurations\n",
pdata->num_retune_mobile_cfgs);
if (pdata->num_retune_mobile_cfgs)
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
wm8904_handle_retune_mobile_pdata(codec);
else
snd_soc_add_codec_controls(codec, wm8904_eq_controls,
ARRAY_SIZE(wm8904_eq_controls));
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
static int wm8904_probe(struct snd_soc_codec *codec)
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
int ret;
codec->control_data = wm8904->regmap;
switch (wm8904->devtype) {
case WM8904:
break;
case WM8912:
memset(&wm8904_dai.capture, 0, sizeof(wm8904_dai.capture));
break;
default:
dev_err(codec->dev, "Unknown device type %d\n",
wm8904->devtype);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
return -EINVAL;
}
ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
if (ret != 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
return ret;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
wm8904_handle_pdata(codec);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
wm8904_add_widgets(codec);
return 0;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
static int wm8904_remove(struct snd_soc_codec *codec)
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
kfree(wm8904->retune_mobile_texts);
kfree(wm8904->drc_texts);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
return 0;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
static struct snd_soc_codec_driver soc_codec_dev_wm8904 = {
.probe = wm8904_probe,
.remove = wm8904_remove,
.set_bias_level = wm8904_set_bias_level,
.idle_bias_off = true,
};
static const struct regmap_config wm8904_regmap = {
.reg_bits = 8,
.val_bits = 16,
.max_register = WM8904_MAX_REGISTER,
.volatile_reg = wm8904_volatile_register,
.readable_reg = wm8904_readable_register,
.cache_type = REGCACHE_RBTREE,
.reg_defaults = wm8904_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(wm8904_reg_defaults),
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
};
static __devinit int wm8904_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct wm8904_priv *wm8904;
unsigned int val;
int ret, i;
wm8904 = devm_kzalloc(&i2c->dev, sizeof(struct wm8904_priv),
GFP_KERNEL);
if (wm8904 == NULL)
return -ENOMEM;
wm8904->regmap = devm_regmap_init_i2c(i2c, &wm8904_regmap);
if (IS_ERR(wm8904->regmap)) {
ret = PTR_ERR(wm8904->regmap);
dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
ret);
return ret;
}
wm8904->devtype = id->driver_data;
i2c_set_clientdata(i2c, wm8904);
wm8904->pdata = i2c->dev.platform_data;
for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++)
wm8904->supplies[i].supply = wm8904_supply_names[i];
ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8904->supplies),
wm8904->supplies);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret);
return ret;
}
ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies),
wm8904->supplies);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret);
return ret;
}
ret = regmap_read(wm8904->regmap, WM8904_SW_RESET_AND_ID, &val);
if (ret < 0) {
dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret);
goto err_enable;
}
if (val != 0x8904) {
dev_err(&i2c->dev, "Device is not a WM8904, ID is %x\n", val);
ret = -EINVAL;
goto err_enable;
}
ret = regmap_read(wm8904->regmap, WM8904_REVISION, &val);
if (ret < 0) {
dev_err(&i2c->dev, "Failed to read device revision: %d\n",
ret);
goto err_enable;
}
dev_info(&i2c->dev, "revision %c\n", val + 'A');
ret = regmap_write(wm8904->regmap, WM8904_SW_RESET_AND_ID, 0);
if (ret < 0) {
dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
goto err_enable;
}
/* Change some default settings - latch VU and enable ZC */
regmap_update_bits(wm8904->regmap, WM8904_ADC_DIGITAL_VOLUME_LEFT,
WM8904_ADC_VU, WM8904_ADC_VU);
regmap_update_bits(wm8904->regmap, WM8904_ADC_DIGITAL_VOLUME_RIGHT,
WM8904_ADC_VU, WM8904_ADC_VU);
regmap_update_bits(wm8904->regmap, WM8904_DAC_DIGITAL_VOLUME_LEFT,
WM8904_DAC_VU, WM8904_DAC_VU);
regmap_update_bits(wm8904->regmap, WM8904_DAC_DIGITAL_VOLUME_RIGHT,
WM8904_DAC_VU, WM8904_DAC_VU);
regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT1_LEFT,
WM8904_HPOUT_VU | WM8904_HPOUTLZC,
WM8904_HPOUT_VU | WM8904_HPOUTLZC);
regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT1_RIGHT,
WM8904_HPOUT_VU | WM8904_HPOUTRZC,
WM8904_HPOUT_VU | WM8904_HPOUTRZC);
regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT2_LEFT,
WM8904_LINEOUT_VU | WM8904_LINEOUTLZC,
WM8904_LINEOUT_VU | WM8904_LINEOUTLZC);
regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT2_RIGHT,
WM8904_LINEOUT_VU | WM8904_LINEOUTRZC,
WM8904_LINEOUT_VU | WM8904_LINEOUTRZC);
regmap_update_bits(wm8904->regmap, WM8904_CLOCK_RATES_0,
WM8904_SR_MODE, 0);
/* Apply configuration from the platform data. */
if (wm8904->pdata) {
for (i = 0; i < WM8904_GPIO_REGS; i++) {
if (!wm8904->pdata->gpio_cfg[i])
continue;
regmap_update_bits(wm8904->regmap,
WM8904_GPIO_CONTROL_1 + i,
0xffff,
wm8904->pdata->gpio_cfg[i]);
}
/* Zero is the default value for these anyway */
for (i = 0; i < WM8904_MIC_REGS; i++)
regmap_update_bits(wm8904->regmap,
WM8904_MIC_BIAS_CONTROL_0 + i,
0xffff,
wm8904->pdata->mic_cfg[i]);
}
/* Set Class W by default - this will be managed by the Class
* G widget at runtime where bypass paths are available.
*/
regmap_update_bits(wm8904->regmap, WM8904_CLASS_W_0,
WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR);
/* Use normal bias source */
regmap_update_bits(wm8904->regmap, WM8904_BIAS_CONTROL_0,
WM8904_POBCTRL, 0);
/* Can leave the device powered off until we need it */
regcache_cache_only(wm8904->regmap, true);
regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_wm8904, &wm8904_dai, 1);
if (ret != 0)
return ret;
return 0;
err_enable:
regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
return ret;
}
static __devexit int wm8904_i2c_remove(struct i2c_client *client)
{
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 20:15:21 +00:00
snd_soc_unregister_codec(&client->dev);
return 0;
}
static const struct i2c_device_id wm8904_i2c_id[] = {
{ "wm8904", WM8904 },
{ "wm8912", WM8912 },
{ "wm8918", WM8904 }, /* Actually a subset, updates to follow */
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8904_i2c_id);
static struct i2c_driver wm8904_i2c_driver = {
.driver = {
.name = "wm8904",
.owner = THIS_MODULE,
},
.probe = wm8904_i2c_probe,
.remove = __devexit_p(wm8904_i2c_remove),
.id_table = wm8904_i2c_id,
};
module_i2c_driver(wm8904_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8904 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_LICENSE("GPL");