alistair23-linux/sound/soc/codecs/wm8978.c

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// SPDX-License-Identifier: GPL-2.0-only
/*
* wm8978.c -- WM8978 ALSA SoC Audio Codec driver
*
* Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
* Copyright (C) 2007 Carlos Munoz <carlos@kenati.com>
* Copyright 2006-2009 Wolfson Microelectronics PLC.
* Based on wm8974 and wm8990 by Liam Girdwood <lrg@slimlogic.co.uk>
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/regmap.h>
include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo <tj@kernel.org> Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org> Cc: Ingo Molnar <mingo@redhat.com> Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
2010-03-24 02:04:11 -06:00
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include <asm/div64.h>
#include "wm8978.h"
static const struct reg_default wm8978_reg_defaults[] = {
{ 1, 0x0000 },
{ 2, 0x0000 },
{ 3, 0x0000 },
{ 4, 0x0050 },
{ 5, 0x0000 },
{ 6, 0x0140 },
{ 7, 0x0000 },
{ 8, 0x0000 },
{ 9, 0x0000 },
{ 10, 0x0000 },
{ 11, 0x00ff },
{ 12, 0x00ff },
{ 13, 0x0000 },
{ 14, 0x0100 },
{ 15, 0x00ff },
{ 16, 0x00ff },
{ 17, 0x0000 },
{ 18, 0x012c },
{ 19, 0x002c },
{ 20, 0x002c },
{ 21, 0x002c },
{ 22, 0x002c },
{ 23, 0x0000 },
{ 24, 0x0032 },
{ 25, 0x0000 },
{ 26, 0x0000 },
{ 27, 0x0000 },
{ 28, 0x0000 },
{ 29, 0x0000 },
{ 30, 0x0000 },
{ 31, 0x0000 },
{ 32, 0x0038 },
{ 33, 0x000b },
{ 34, 0x0032 },
{ 35, 0x0000 },
{ 36, 0x0008 },
{ 37, 0x000c },
{ 38, 0x0093 },
{ 39, 0x00e9 },
{ 40, 0x0000 },
{ 41, 0x0000 },
{ 42, 0x0000 },
{ 43, 0x0000 },
{ 44, 0x0033 },
{ 45, 0x0010 },
{ 46, 0x0010 },
{ 47, 0x0100 },
{ 48, 0x0100 },
{ 49, 0x0002 },
{ 50, 0x0001 },
{ 51, 0x0001 },
{ 52, 0x0039 },
{ 53, 0x0039 },
{ 54, 0x0039 },
{ 55, 0x0039 },
{ 56, 0x0001 },
{ 57, 0x0001 },
};
static bool wm8978_volatile(struct device *dev, unsigned int reg)
{
return reg == WM8978_RESET;
}
/* codec private data */
struct wm8978_priv {
struct regmap *regmap;
unsigned int f_pllout;
unsigned int f_mclk;
unsigned int f_256fs;
unsigned int f_opclk;
int mclk_idx;
enum wm8978_sysclk_src sysclk;
};
static const char *wm8978_companding[] = {"Off", "NC", "u-law", "A-law"};
static const char *wm8978_eqmode[] = {"Capture", "Playback"};
static const char *wm8978_bw[] = {"Narrow", "Wide"};
static const char *wm8978_eq1[] = {"80Hz", "105Hz", "135Hz", "175Hz"};
static const char *wm8978_eq2[] = {"230Hz", "300Hz", "385Hz", "500Hz"};
static const char *wm8978_eq3[] = {"650Hz", "850Hz", "1.1kHz", "1.4kHz"};
static const char *wm8978_eq4[] = {"1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz"};
static const char *wm8978_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz"};
static const char *wm8978_alc3[] = {"ALC", "Limiter"};
static const char *wm8978_alc1[] = {"Off", "Right", "Left", "Both"};
static SOC_ENUM_SINGLE_DECL(adc_compand, WM8978_COMPANDING_CONTROL, 1,
wm8978_companding);
static SOC_ENUM_SINGLE_DECL(dac_compand, WM8978_COMPANDING_CONTROL, 3,
wm8978_companding);
static SOC_ENUM_SINGLE_DECL(eqmode, WM8978_EQ1, 8, wm8978_eqmode);
static SOC_ENUM_SINGLE_DECL(eq1, WM8978_EQ1, 5, wm8978_eq1);
static SOC_ENUM_SINGLE_DECL(eq2bw, WM8978_EQ2, 8, wm8978_bw);
static SOC_ENUM_SINGLE_DECL(eq2, WM8978_EQ2, 5, wm8978_eq2);
static SOC_ENUM_SINGLE_DECL(eq3bw, WM8978_EQ3, 8, wm8978_bw);
static SOC_ENUM_SINGLE_DECL(eq3, WM8978_EQ3, 5, wm8978_eq3);
static SOC_ENUM_SINGLE_DECL(eq4bw, WM8978_EQ4, 8, wm8978_bw);
static SOC_ENUM_SINGLE_DECL(eq4, WM8978_EQ4, 5, wm8978_eq4);
static SOC_ENUM_SINGLE_DECL(eq5, WM8978_EQ5, 5, wm8978_eq5);
static SOC_ENUM_SINGLE_DECL(alc3, WM8978_ALC_CONTROL_3, 8, wm8978_alc3);
static SOC_ENUM_SINGLE_DECL(alc1, WM8978_ALC_CONTROL_1, 7, wm8978_alc1);
static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1);
static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0);
static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0);
static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1);
static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0);
static const struct snd_kcontrol_new wm8978_snd_controls[] = {
SOC_SINGLE("Digital Loopback Switch",
WM8978_COMPANDING_CONTROL, 0, 1, 0),
SOC_ENUM("ADC Companding", adc_compand),
SOC_ENUM("DAC Companding", dac_compand),
SOC_DOUBLE("DAC Inversion Switch", WM8978_DAC_CONTROL, 0, 1, 1, 0),
SOC_DOUBLE_R_TLV("PCM Volume",
WM8978_LEFT_DAC_DIGITAL_VOLUME, WM8978_RIGHT_DAC_DIGITAL_VOLUME,
0, 255, 0, digital_tlv),
SOC_SINGLE("High Pass Filter Switch", WM8978_ADC_CONTROL, 8, 1, 0),
SOC_SINGLE("High Pass Cut Off", WM8978_ADC_CONTROL, 4, 7, 0),
SOC_DOUBLE("ADC Inversion Switch", WM8978_ADC_CONTROL, 0, 1, 1, 0),
SOC_DOUBLE_R_TLV("ADC Volume",
WM8978_LEFT_ADC_DIGITAL_VOLUME, WM8978_RIGHT_ADC_DIGITAL_VOLUME,
0, 255, 0, digital_tlv),
SOC_ENUM("Equaliser Function", eqmode),
SOC_ENUM("EQ1 Cut Off", eq1),
SOC_SINGLE_TLV("EQ1 Volume", WM8978_EQ1, 0, 24, 1, eq_tlv),
SOC_ENUM("Equaliser EQ2 Bandwidth", eq2bw),
SOC_ENUM("EQ2 Cut Off", eq2),
SOC_SINGLE_TLV("EQ2 Volume", WM8978_EQ2, 0, 24, 1, eq_tlv),
SOC_ENUM("Equaliser EQ3 Bandwidth", eq3bw),
SOC_ENUM("EQ3 Cut Off", eq3),
SOC_SINGLE_TLV("EQ3 Volume", WM8978_EQ3, 0, 24, 1, eq_tlv),
SOC_ENUM("Equaliser EQ4 Bandwidth", eq4bw),
SOC_ENUM("EQ4 Cut Off", eq4),
SOC_SINGLE_TLV("EQ4 Volume", WM8978_EQ4, 0, 24, 1, eq_tlv),
SOC_ENUM("EQ5 Cut Off", eq5),
SOC_SINGLE_TLV("EQ5 Volume", WM8978_EQ5, 0, 24, 1, eq_tlv),
SOC_SINGLE("DAC Playback Limiter Switch",
WM8978_DAC_LIMITER_1, 8, 1, 0),
SOC_SINGLE("DAC Playback Limiter Decay",
WM8978_DAC_LIMITER_1, 4, 15, 0),
SOC_SINGLE("DAC Playback Limiter Attack",
WM8978_DAC_LIMITER_1, 0, 15, 0),
SOC_SINGLE("DAC Playback Limiter Threshold",
WM8978_DAC_LIMITER_2, 4, 7, 0),
SOC_SINGLE_TLV("DAC Playback Limiter Volume",
WM8978_DAC_LIMITER_2, 0, 12, 0, limiter_tlv),
SOC_ENUM("ALC Enable Switch", alc1),
SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0),
SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0),
SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 10, 0),
SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0),
SOC_ENUM("ALC Capture Mode", alc3),
SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 10, 0),
SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 10, 0),
SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0),
SOC_SINGLE("ALC Capture Noise Gate Threshold",
WM8978_NOISE_GATE, 0, 7, 0),
SOC_DOUBLE_R("Capture PGA ZC Switch",
WM8978_LEFT_INP_PGA_CONTROL, WM8978_RIGHT_INP_PGA_CONTROL,
7, 1, 0),
/* OUT1 - Headphones */
SOC_DOUBLE_R("Headphone Playback ZC Switch",
WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, 7, 1, 0),
SOC_DOUBLE_R_TLV("Headphone Playback Volume",
WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL,
0, 63, 0, spk_tlv),
/* OUT2 - Speakers */
SOC_DOUBLE_R("Speaker Playback ZC Switch",
WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 7, 1, 0),
SOC_DOUBLE_R_TLV("Speaker Playback Volume",
WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL,
0, 63, 0, spk_tlv),
/* OUT3/4 - Line Output */
SOC_DOUBLE_R("Line Playback Switch",
WM8978_OUT3_MIXER_CONTROL, WM8978_OUT4_MIXER_CONTROL, 6, 1, 1),
/* Mixer #3: Boost (Input) mixer */
SOC_DOUBLE_R("PGA Boost (+20dB)",
WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL,
8, 1, 0),
SOC_DOUBLE_R_TLV("L2/R2 Boost Volume",
WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL,
4, 7, 0, boost_tlv),
SOC_DOUBLE_R_TLV("Aux Boost Volume",
WM8978_LEFT_ADC_BOOST_CONTROL, WM8978_RIGHT_ADC_BOOST_CONTROL,
0, 7, 0, boost_tlv),
/* Input PGA volume */
SOC_DOUBLE_R_TLV("Input PGA Volume",
WM8978_LEFT_INP_PGA_CONTROL, WM8978_RIGHT_INP_PGA_CONTROL,
0, 63, 0, inpga_tlv),
/* Headphone */
SOC_DOUBLE_R("Headphone Switch",
WM8978_LOUT1_HP_CONTROL, WM8978_ROUT1_HP_CONTROL, 6, 1, 1),
/* Speaker */
SOC_DOUBLE_R("Speaker Switch",
WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1),
/* DAC / ADC oversampling */
SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL,
5, 1, 0),
SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL,
5, 1, 0),
};
/* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */
static const struct snd_kcontrol_new wm8978_left_out_mixer[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", WM8978_LEFT_MIXER_CONTROL, 1, 1, 0),
SOC_DAPM_SINGLE("Aux Playback Switch", WM8978_LEFT_MIXER_CONTROL, 5, 1, 0),
SOC_DAPM_SINGLE("PCM Playback Switch", WM8978_LEFT_MIXER_CONTROL, 0, 1, 0),
};
static const struct snd_kcontrol_new wm8978_right_out_mixer[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", WM8978_RIGHT_MIXER_CONTROL, 1, 1, 0),
SOC_DAPM_SINGLE("Aux Playback Switch", WM8978_RIGHT_MIXER_CONTROL, 5, 1, 0),
SOC_DAPM_SINGLE("PCM Playback Switch", WM8978_RIGHT_MIXER_CONTROL, 0, 1, 0),
};
/* OUT3/OUT4 Mixer not implemented */
/* Mixer #2: Input PGA Mute */
static const struct snd_kcontrol_new wm8978_left_input_mixer[] = {
SOC_DAPM_SINGLE("L2 Switch", WM8978_INPUT_CONTROL, 2, 1, 0),
SOC_DAPM_SINGLE("MicN Switch", WM8978_INPUT_CONTROL, 1, 1, 0),
SOC_DAPM_SINGLE("MicP Switch", WM8978_INPUT_CONTROL, 0, 1, 0),
};
static const struct snd_kcontrol_new wm8978_right_input_mixer[] = {
SOC_DAPM_SINGLE("R2 Switch", WM8978_INPUT_CONTROL, 6, 1, 0),
SOC_DAPM_SINGLE("MicN Switch", WM8978_INPUT_CONTROL, 5, 1, 0),
SOC_DAPM_SINGLE("MicP Switch", WM8978_INPUT_CONTROL, 4, 1, 0),
};
static const struct snd_soc_dapm_widget wm8978_dapm_widgets[] = {
SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
WM8978_POWER_MANAGEMENT_3, 0, 0),
SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
WM8978_POWER_MANAGEMENT_3, 1, 0),
SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
WM8978_POWER_MANAGEMENT_2, 0, 0),
SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
WM8978_POWER_MANAGEMENT_2, 1, 0),
/* Mixer #1: OUT1,2 */
SOC_MIXER_ARRAY("Left Output Mixer", WM8978_POWER_MANAGEMENT_3,
2, 0, wm8978_left_out_mixer),
SOC_MIXER_ARRAY("Right Output Mixer", WM8978_POWER_MANAGEMENT_3,
3, 0, wm8978_right_out_mixer),
SOC_MIXER_ARRAY("Left Input Mixer", WM8978_POWER_MANAGEMENT_2,
2, 0, wm8978_left_input_mixer),
SOC_MIXER_ARRAY("Right Input Mixer", WM8978_POWER_MANAGEMENT_2,
3, 0, wm8978_right_input_mixer),
SND_SOC_DAPM_PGA("Left Boost Mixer", WM8978_POWER_MANAGEMENT_2,
4, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Boost Mixer", WM8978_POWER_MANAGEMENT_2,
5, 0, NULL, 0),
SND_SOC_DAPM_PGA("Left Capture PGA", WM8978_LEFT_INP_PGA_CONTROL,
6, 1, NULL, 0),
SND_SOC_DAPM_PGA("Right Capture PGA", WM8978_RIGHT_INP_PGA_CONTROL,
6, 1, NULL, 0),
SND_SOC_DAPM_PGA("Left Headphone Out", WM8978_POWER_MANAGEMENT_2,
7, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Headphone Out", WM8978_POWER_MANAGEMENT_2,
8, 0, NULL, 0),
SND_SOC_DAPM_PGA("Left Speaker Out", WM8978_POWER_MANAGEMENT_3,
6, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Speaker Out", WM8978_POWER_MANAGEMENT_3,
5, 0, NULL, 0),
SND_SOC_DAPM_MIXER("OUT4 VMID", WM8978_POWER_MANAGEMENT_3,
8, 0, NULL, 0),
SND_SOC_DAPM_MICBIAS("Mic Bias", WM8978_POWER_MANAGEMENT_1, 4, 0),
SND_SOC_DAPM_INPUT("LMICN"),
SND_SOC_DAPM_INPUT("LMICP"),
SND_SOC_DAPM_INPUT("RMICN"),
SND_SOC_DAPM_INPUT("RMICP"),
SND_SOC_DAPM_INPUT("LAUX"),
SND_SOC_DAPM_INPUT("RAUX"),
SND_SOC_DAPM_INPUT("L2"),
SND_SOC_DAPM_INPUT("R2"),
SND_SOC_DAPM_OUTPUT("LHP"),
SND_SOC_DAPM_OUTPUT("RHP"),
SND_SOC_DAPM_OUTPUT("LSPK"),
SND_SOC_DAPM_OUTPUT("RSPK"),
};
static const struct snd_soc_dapm_route wm8978_dapm_routes[] = {
/* Output mixer */
{"Right Output Mixer", "PCM Playback Switch", "Right DAC"},
{"Right Output Mixer", "Aux Playback Switch", "RAUX"},
{"Right Output Mixer", "Line Bypass Switch", "Right Boost Mixer"},
{"Left Output Mixer", "PCM Playback Switch", "Left DAC"},
{"Left Output Mixer", "Aux Playback Switch", "LAUX"},
{"Left Output Mixer", "Line Bypass Switch", "Left Boost Mixer"},
/* Outputs */
{"Right Headphone Out", NULL, "Right Output Mixer"},
{"RHP", NULL, "Right Headphone Out"},
{"Left Headphone Out", NULL, "Left Output Mixer"},
{"LHP", NULL, "Left Headphone Out"},
{"Right Speaker Out", NULL, "Right Output Mixer"},
{"RSPK", NULL, "Right Speaker Out"},
{"Left Speaker Out", NULL, "Left Output Mixer"},
{"LSPK", NULL, "Left Speaker Out"},
/* Boost Mixer */
{"Right ADC", NULL, "Right Boost Mixer"},
{"Right Boost Mixer", NULL, "RAUX"},
{"Right Boost Mixer", NULL, "Right Capture PGA"},
{"Right Boost Mixer", NULL, "R2"},
{"Left ADC", NULL, "Left Boost Mixer"},
{"Left Boost Mixer", NULL, "LAUX"},
{"Left Boost Mixer", NULL, "Left Capture PGA"},
{"Left Boost Mixer", NULL, "L2"},
/* Input PGA */
{"Right Capture PGA", NULL, "Right Input Mixer"},
{"Left Capture PGA", NULL, "Left Input Mixer"},
{"Right Input Mixer", "R2 Switch", "R2"},
{"Right Input Mixer", "MicN Switch", "RMICN"},
{"Right Input Mixer", "MicP Switch", "RMICP"},
{"Left Input Mixer", "L2 Switch", "L2"},
{"Left Input Mixer", "MicN Switch", "LMICN"},
{"Left Input Mixer", "MicP Switch", "LMICP"},
};
/* PLL divisors */
struct wm8978_pll_div {
u32 k;
u8 n;
u8 div2;
};
#define FIXED_PLL_SIZE (1 << 24)
static void pll_factors(struct snd_soc_component *component,
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 14:15:21 -06:00
struct wm8978_pll_div *pll_div, unsigned int target, unsigned int source)
{
u64 k_part;
unsigned int k, n_div, n_mod;
n_div = target / source;
if (n_div < 6) {
source >>= 1;
pll_div->div2 = 1;
n_div = target / source;
} else {
pll_div->div2 = 0;
}
if (n_div < 6 || n_div > 12)
dev_warn(component->dev,
"WM8978 N value exceeds recommended range! N = %u\n",
n_div);
pll_div->n = n_div;
n_mod = target - source * n_div;
k_part = FIXED_PLL_SIZE * (long long)n_mod + source / 2;
do_div(k_part, source);
k = k_part & 0xFFFFFFFF;
pll_div->k = k;
}
/* MCLK dividers */
static const int mclk_numerator[] = {1, 3, 2, 3, 4, 6, 8, 12};
static const int mclk_denominator[] = {1, 2, 1, 1, 1, 1, 1, 1};
/*
* find index >= idx, such that, for a given f_out,
* 3 * f_mclk / 4 <= f_PLLOUT < 13 * f_mclk / 4
* f_out can be f_256fs or f_opclk, currently only used for f_256fs. Can be
* generalised for f_opclk with suitable coefficient arrays, but currently
* the OPCLK divisor is calculated directly, not iteratively.
*/
static int wm8978_enum_mclk(unsigned int f_out, unsigned int f_mclk,
unsigned int *f_pllout)
{
int i;
for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) {
unsigned int f_pllout_x4 = 4 * f_out * mclk_numerator[i] /
mclk_denominator[i];
if (3 * f_mclk <= f_pllout_x4 && f_pllout_x4 < 13 * f_mclk) {
*f_pllout = f_pllout_x4 / 4;
return i;
}
}
return -EINVAL;
}
/*
* Calculate internal frequencies and dividers, according to Figure 40
* "PLL and Clock Select Circuit" in WM8978 datasheet Rev. 2.6
*/
static int wm8978_configure_pll(struct snd_soc_component *component)
{
struct wm8978_priv *wm8978 = snd_soc_component_get_drvdata(component);
struct wm8978_pll_div pll_div;
unsigned int f_opclk = wm8978->f_opclk, f_mclk = wm8978->f_mclk,
f_256fs = wm8978->f_256fs;
unsigned int f2;
if (!f_mclk)
return -EINVAL;
if (f_opclk) {
unsigned int opclk_div;
/* Cannot set up MCLK divider now, do later */
wm8978->mclk_idx = -1;
/*
* The user needs OPCLK. Choose OPCLKDIV to put
* 6 <= R = f2 / f1 < 13, 1 <= OPCLKDIV <= 4.
* f_opclk = f_mclk * prescale * R / 4 / OPCLKDIV, where
* prescale = 1, or prescale = 2. Prescale is calculated inside
* pll_factors(). We have to select f_PLLOUT, such that
* f_mclk * 3 / 4 <= f_PLLOUT < f_mclk * 13 / 4. Must be
* f_mclk * 3 / 16 <= f_opclk < f_mclk * 13 / 4.
*/
if (16 * f_opclk < 3 * f_mclk || 4 * f_opclk >= 13 * f_mclk)
return -EINVAL;
if (4 * f_opclk < 3 * f_mclk)
/* Have to use OPCLKDIV */
opclk_div = (3 * f_mclk / 4 + f_opclk - 1) / f_opclk;
else
opclk_div = 1;
dev_dbg(component->dev, "%s: OPCLKDIV=%d\n", __func__, opclk_div);
snd_soc_component_update_bits(component, WM8978_GPIO_CONTROL, 0x30,
(opclk_div - 1) << 4);
wm8978->f_pllout = f_opclk * opclk_div;
} else if (f_256fs) {
/*
* Not using OPCLK, but PLL is used for the codec, choose R:
* 6 <= R = f2 / f1 < 13, to put 1 <= MCLKDIV <= 12.
* f_256fs = f_mclk * prescale * R / 4 / MCLKDIV, where
* prescale = 1, or prescale = 2. Prescale is calculated inside
* pll_factors(). We have to select f_PLLOUT, such that
* f_mclk * 3 / 4 <= f_PLLOUT < f_mclk * 13 / 4. Must be
* f_mclk * 3 / 48 <= f_256fs < f_mclk * 13 / 4. This means MCLK
* must be 3.781MHz <= f_MCLK <= 32.768MHz
*/
int idx = wm8978_enum_mclk(f_256fs, f_mclk, &wm8978->f_pllout);
if (idx < 0)
return idx;
wm8978->mclk_idx = idx;
} else {
return -EINVAL;
}
f2 = wm8978->f_pllout * 4;
dev_dbg(component->dev, "%s: f_MCLK=%uHz, f_PLLOUT=%uHz\n", __func__,
wm8978->f_mclk, wm8978->f_pllout);
pll_factors(component, &pll_div, f2, wm8978->f_mclk);
dev_dbg(component->dev, "%s: calculated PLL N=0x%x, K=0x%x, div2=%d\n",
__func__, pll_div.n, pll_div.k, pll_div.div2);
/* Turn PLL off for configuration... */
snd_soc_component_update_bits(component, WM8978_POWER_MANAGEMENT_1, 0x20, 0);
snd_soc_component_write(component, WM8978_PLL_N, (pll_div.div2 << 4) | pll_div.n);
snd_soc_component_write(component, WM8978_PLL_K1, pll_div.k >> 18);
snd_soc_component_write(component, WM8978_PLL_K2, (pll_div.k >> 9) & 0x1ff);
snd_soc_component_write(component, WM8978_PLL_K3, pll_div.k & 0x1ff);
/* ...and on again */
snd_soc_component_update_bits(component, WM8978_POWER_MANAGEMENT_1, 0x20, 0x20);
if (f_opclk)
/* Output PLL (OPCLK) to GPIO1 */
snd_soc_component_update_bits(component, WM8978_GPIO_CONTROL, 7, 4);
return 0;
}
/*
* Configure WM8978 clock dividers.
*/
static int wm8978_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
int div_id, int div)
{
struct snd_soc_component *component = codec_dai->component;
struct wm8978_priv *wm8978 = snd_soc_component_get_drvdata(component);
int ret = 0;
switch (div_id) {
case WM8978_OPCLKRATE:
wm8978->f_opclk = div;
if (wm8978->f_mclk)
/*
* We know the MCLK frequency, the user has requested
* OPCLK, configure the PLL based on that and start it
* and OPCLK immediately. We will configure PLL to match
* user-requested OPCLK frquency as good as possible.
* In fact, it is likely, that matching the sampling
* rate, when it becomes known, is more important, and
* we will not be reconfiguring PLL then, because we
* must not interrupt OPCLK. But it should be fine,
* because typically the user will request OPCLK to run
* at 256fs or 512fs, and for these cases we will also
* find an exact MCLK divider configuration - it will
* be equal to or double the OPCLK divisor.
*/
ret = wm8978_configure_pll(component);
break;
case WM8978_BCLKDIV:
if (div & ~0x1c)
return -EINVAL;
snd_soc_component_update_bits(component, WM8978_CLOCKING, 0x1c, div);
break;
default:
return -EINVAL;
}
dev_dbg(component->dev, "%s: ID %d, value %u\n", __func__, div_id, div);
return ret;
}
/*
* @freq: when .set_pll() us not used, freq is codec MCLK input frequency
*/
static int wm8978_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id,
unsigned int freq, int dir)
{
struct snd_soc_component *component = codec_dai->component;
struct wm8978_priv *wm8978 = snd_soc_component_get_drvdata(component);
int ret = 0;
dev_dbg(component->dev, "%s: ID %d, freq %u\n", __func__, clk_id, freq);
if (freq) {
wm8978->f_mclk = freq;
/* Even if MCLK is used for system clock, might have to drive OPCLK */
if (wm8978->f_opclk)
ret = wm8978_configure_pll(component);
/* Our sysclk is fixed to 256 * fs, will configure in .hw_params() */
if (!ret)
wm8978->sysclk = clk_id;
}
if (wm8978->sysclk == WM8978_PLL && (!freq || clk_id == WM8978_MCLK)) {
/* Clock CODEC directly from MCLK */
snd_soc_component_update_bits(component, WM8978_CLOCKING, 0x100, 0);
/* GPIO1 into default mode as input - before configuring PLL */
snd_soc_component_update_bits(component, WM8978_GPIO_CONTROL, 7, 0);
/* Turn off PLL */
snd_soc_component_update_bits(component, WM8978_POWER_MANAGEMENT_1, 0x20, 0);
wm8978->sysclk = WM8978_MCLK;
wm8978->f_pllout = 0;
wm8978->f_opclk = 0;
}
return ret;
}
/*
* Set ADC and Voice DAC format.
*/
static int wm8978_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
{
struct snd_soc_component *component = codec_dai->component;
/*
* BCLK polarity mask = 0x100, LRC clock polarity mask = 0x80,
* Data Format mask = 0x18: all will be calculated anew
*/
u16 iface = snd_soc_component_read32(component, WM8978_AUDIO_INTERFACE) & ~0x198;
u16 clk = snd_soc_component_read32(component, WM8978_CLOCKING);
dev_dbg(component->dev, "%s\n", __func__);
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
clk |= 1;
break;
case SND_SOC_DAIFMT_CBS_CFS:
clk &= ~1;
break;
default:
return -EINVAL;
}
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
iface |= 0x10;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
iface |= 0x8;
break;
case SND_SOC_DAIFMT_DSP_A:
iface |= 0x18;
break;
default:
return -EINVAL;
}
/* clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_IB_IF:
iface |= 0x180;
break;
case SND_SOC_DAIFMT_IB_NF:
iface |= 0x100;
break;
case SND_SOC_DAIFMT_NB_IF:
iface |= 0x80;
break;
default:
return -EINVAL;
}
snd_soc_component_write(component, WM8978_AUDIO_INTERFACE, iface);
snd_soc_component_write(component, WM8978_CLOCKING, clk);
return 0;
}
/*
* Set PCM DAI bit size and sample rate.
*/
static int wm8978_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct wm8978_priv *wm8978 = snd_soc_component_get_drvdata(component);
/* Word length mask = 0x60 */
u16 iface_ctl = snd_soc_component_read32(component, WM8978_AUDIO_INTERFACE) & ~0x60;
/* Sampling rate mask = 0xe (for filters) */
u16 add_ctl = snd_soc_component_read32(component, WM8978_ADDITIONAL_CONTROL) & ~0xe;
u16 clking = snd_soc_component_read32(component, WM8978_CLOCKING);
enum wm8978_sysclk_src current_clk_id = clking & 0x100 ?
WM8978_PLL : WM8978_MCLK;
unsigned int f_sel, diff, diff_best = INT_MAX;
int i, best = 0;
if (!wm8978->f_mclk)
return -EINVAL;
/* bit size */
switch (params_width(params)) {
case 16:
break;
case 20:
iface_ctl |= 0x20;
break;
case 24:
iface_ctl |= 0x40;
break;
case 32:
iface_ctl |= 0x60;
break;
}
/* filter coefficient */
switch (params_rate(params)) {
case 8000:
add_ctl |= 0x5 << 1;
break;
case 11025:
add_ctl |= 0x4 << 1;
break;
case 16000:
add_ctl |= 0x3 << 1;
break;
case 22050:
add_ctl |= 0x2 << 1;
break;
case 32000:
add_ctl |= 0x1 << 1;
break;
case 44100:
case 48000:
break;
}
/* Sampling rate is known now, can configure the MCLK divider */
wm8978->f_256fs = params_rate(params) * 256;
if (wm8978->sysclk == WM8978_MCLK) {
wm8978->mclk_idx = -1;
f_sel = wm8978->f_mclk;
} else {
if (!wm8978->f_opclk) {
/* We only enter here, if OPCLK is not used */
int ret = wm8978_configure_pll(component);
if (ret < 0)
return ret;
}
f_sel = wm8978->f_pllout;
}
if (wm8978->mclk_idx < 0) {
/* Either MCLK is used directly, or OPCLK is used */
if (f_sel < wm8978->f_256fs || f_sel > 12 * wm8978->f_256fs)
return -EINVAL;
for (i = 0; i < ARRAY_SIZE(mclk_numerator); i++) {
diff = abs(wm8978->f_256fs * 3 -
f_sel * 3 * mclk_denominator[i] / mclk_numerator[i]);
if (diff < diff_best) {
diff_best = diff;
best = i;
}
if (!diff)
break;
}
} else {
/* OPCLK not used, codec driven by PLL */
best = wm8978->mclk_idx;
diff = 0;
}
if (diff)
dev_warn(component->dev, "Imprecise sampling rate: %uHz%s\n",
f_sel * mclk_denominator[best] / mclk_numerator[best] / 256,
wm8978->sysclk == WM8978_MCLK ?
", consider using PLL" : "");
dev_dbg(component->dev, "%s: width %d, rate %u, MCLK divisor #%d\n", __func__,
params_width(params), params_rate(params), best);
/* MCLK divisor mask = 0xe0 */
snd_soc_component_update_bits(component, WM8978_CLOCKING, 0xe0, best << 5);
snd_soc_component_write(component, WM8978_AUDIO_INTERFACE, iface_ctl);
snd_soc_component_write(component, WM8978_ADDITIONAL_CONTROL, add_ctl);
if (wm8978->sysclk != current_clk_id) {
if (wm8978->sysclk == WM8978_PLL)
/* Run CODEC from PLL instead of MCLK */
snd_soc_component_update_bits(component, WM8978_CLOCKING,
0x100, 0x100);
else
/* Clock CODEC directly from MCLK */
snd_soc_component_update_bits(component, WM8978_CLOCKING, 0x100, 0);
}
return 0;
}
static int wm8978_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_component *component = dai->component;
dev_dbg(component->dev, "%s: %d\n", __func__, mute);
if (mute)
snd_soc_component_update_bits(component, WM8978_DAC_CONTROL, 0x40, 0x40);
else
snd_soc_component_update_bits(component, WM8978_DAC_CONTROL, 0x40, 0);
return 0;
}
static int wm8978_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
u16 power1 = snd_soc_component_read32(component, WM8978_POWER_MANAGEMENT_1) & ~3;
switch (level) {
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
power1 |= 1; /* VMID 75k */
snd_soc_component_write(component, WM8978_POWER_MANAGEMENT_1, power1);
break;
case SND_SOC_BIAS_STANDBY:
/* bit 3: enable bias, bit 2: enable I/O tie off buffer */
power1 |= 0xc;
if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) {
/* Initial cap charge at VMID 5k */
snd_soc_component_write(component, WM8978_POWER_MANAGEMENT_1,
power1 | 0x3);
mdelay(100);
}
power1 |= 0x2; /* VMID 500k */
snd_soc_component_write(component, WM8978_POWER_MANAGEMENT_1, power1);
break;
case SND_SOC_BIAS_OFF:
/* Preserve PLL - OPCLK may be used by someone */
snd_soc_component_update_bits(component, WM8978_POWER_MANAGEMENT_1, ~0x20, 0);
snd_soc_component_write(component, WM8978_POWER_MANAGEMENT_2, 0);
snd_soc_component_write(component, WM8978_POWER_MANAGEMENT_3, 0);
break;
}
dev_dbg(component->dev, "%s: %d, %x\n", __func__, level, power1);
return 0;
}
#define WM8978_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 wm8978_dai_ops = {
.hw_params = wm8978_hw_params,
.digital_mute = wm8978_mute,
.set_fmt = wm8978_set_dai_fmt,
.set_clkdiv = wm8978_set_dai_clkdiv,
.set_sysclk = wm8978_set_dai_sysclk,
};
/* Also supports 12kHz */
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 14:15:21 -06:00
static struct snd_soc_dai_driver wm8978_dai = {
.name = "wm8978-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = WM8978_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = WM8978_FORMATS,
},
.ops = &wm8978_dai_ops,
.symmetric_rates = 1,
};
static int wm8978_suspend(struct snd_soc_component *component)
{
struct wm8978_priv *wm8978 = snd_soc_component_get_drvdata(component);
snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF);
/* Also switch PLL off */
snd_soc_component_write(component, WM8978_POWER_MANAGEMENT_1, 0);
regcache_mark_dirty(wm8978->regmap);
return 0;
}
static int wm8978_resume(struct snd_soc_component *component)
{
struct wm8978_priv *wm8978 = snd_soc_component_get_drvdata(component);
/* Sync reg_cache with the hardware */
regcache_sync(wm8978->regmap);
snd_soc_component_force_bias_level(component, SND_SOC_BIAS_STANDBY);
if (wm8978->f_pllout)
/* Switch PLL on */
snd_soc_component_update_bits(component, WM8978_POWER_MANAGEMENT_1, 0x20, 0x20);
return 0;
}
/*
* These registers contain an "update" bit - bit 8. This means, for example,
* that one can write new DAC digital volume for both channels, but only when
* the update bit is set, will also the volume be updated - simultaneously for
* both channels.
*/
static const int update_reg[] = {
WM8978_LEFT_DAC_DIGITAL_VOLUME,
WM8978_RIGHT_DAC_DIGITAL_VOLUME,
WM8978_LEFT_ADC_DIGITAL_VOLUME,
WM8978_RIGHT_ADC_DIGITAL_VOLUME,
WM8978_LEFT_INP_PGA_CONTROL,
WM8978_RIGHT_INP_PGA_CONTROL,
WM8978_LOUT1_HP_CONTROL,
WM8978_ROUT1_HP_CONTROL,
WM8978_LOUT2_SPK_CONTROL,
WM8978_ROUT2_SPK_CONTROL,
};
static int wm8978_probe(struct snd_soc_component *component)
{
struct wm8978_priv *wm8978 = snd_soc_component_get_drvdata(component);
int i;
/*
* Set default system clock to PLL, it is more precise, this is also the
* default hardware setting
*/
wm8978->sysclk = WM8978_PLL;
/*
* Set the update bit in all registers, that have one. This way all
* writes to those registers will also cause the update bit to be
* written.
*/
for (i = 0; i < ARRAY_SIZE(update_reg); i++)
snd_soc_component_update_bits(component, update_reg[i], 0x100, 0x100);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 14:15:21 -06:00
return 0;
}
static const struct snd_soc_component_driver soc_component_dev_wm8978 = {
.probe = wm8978_probe,
.suspend = wm8978_suspend,
.resume = wm8978_resume,
.set_bias_level = wm8978_set_bias_level,
.controls = wm8978_snd_controls,
.num_controls = ARRAY_SIZE(wm8978_snd_controls),
.dapm_widgets = wm8978_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm8978_dapm_widgets),
.dapm_routes = wm8978_dapm_routes,
.num_dapm_routes = ARRAY_SIZE(wm8978_dapm_routes),
.idle_bias_on = 1,
.use_pmdown_time = 1,
.endianness = 1,
.non_legacy_dai_naming = 1,
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 14:15:21 -06:00
};
static const struct regmap_config wm8978_regmap_config = {
.reg_bits = 7,
.val_bits = 9,
.max_register = WM8978_MAX_REGISTER,
.volatile_reg = wm8978_volatile,
.cache_type = REGCACHE_RBTREE,
.reg_defaults = wm8978_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(wm8978_reg_defaults),
};
static int wm8978_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct wm8978_priv *wm8978;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 14:15:21 -06:00
int ret;
wm8978 = devm_kzalloc(&i2c->dev, sizeof(struct wm8978_priv),
GFP_KERNEL);
if (wm8978 == NULL)
return -ENOMEM;
wm8978->regmap = devm_regmap_init_i2c(i2c, &wm8978_regmap_config);
if (IS_ERR(wm8978->regmap)) {
ret = PTR_ERR(wm8978->regmap);
dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
return ret;
}
i2c_set_clientdata(i2c, wm8978);
/* Reset the codec */
ret = regmap_write(wm8978->regmap, WM8978_RESET, 0);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
return ret;
}
ret = devm_snd_soc_register_component(&i2c->dev,
&soc_component_dev_wm8978, &wm8978_dai, 1);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
return ret;
}
return 0;
}
static const struct i2c_device_id wm8978_i2c_id[] = {
{ "wm8978", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8978_i2c_id);
static const struct of_device_id wm8978_of_match[] = {
{ .compatible = "wlf,wm8978", },
{ }
};
MODULE_DEVICE_TABLE(of, wm8978_of_match);
static struct i2c_driver wm8978_i2c_driver = {
.driver = {
.name = "wm8978",
.of_match_table = wm8978_of_match,
},
.probe = wm8978_i2c_probe,
.id_table = wm8978_i2c_id,
};
module_i2c_driver(wm8978_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8978 codec driver");
MODULE_AUTHOR("Guennadi Liakhovetski <g.liakhovetski@gmx.de>");
MODULE_LICENSE("GPL");