OpenCloudOS-Kernel/sound/soc/codecs/wm8960.c

1493 lines
41 KiB
C
Raw Normal View History

/*
* wm8960.c -- WM8960 ALSA SoC Audio driver
*
* Copyright 2007-11 Wolfson Microelectronics, plc
*
* Author: Liam Girdwood
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/clk.h>
#include <linux/i2c.h>
include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo <tj@kernel.org> Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org> Cc: Ingo Molnar <mingo@redhat.com> Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
2010-03-24 16:04:11 +08:00
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include <sound/wm8960.h>
#include "wm8960.h"
/* R25 - Power 1 */
#define WM8960_VMID_MASK 0x180
#define WM8960_VREF 0x40
/* R26 - Power 2 */
#define WM8960_PWR2_LOUT1 0x40
#define WM8960_PWR2_ROUT1 0x20
#define WM8960_PWR2_OUT3 0x02
/* R28 - Anti-pop 1 */
#define WM8960_POBCTRL 0x80
#define WM8960_BUFDCOPEN 0x10
#define WM8960_BUFIOEN 0x08
#define WM8960_SOFT_ST 0x04
#define WM8960_HPSTBY 0x01
/* R29 - Anti-pop 2 */
#define WM8960_DISOP 0x40
#define WM8960_DRES_MASK 0x30
static bool is_pll_freq_available(unsigned int source, unsigned int target);
static int wm8960_set_pll(struct snd_soc_component *component,
unsigned int freq_in, unsigned int freq_out);
/*
* wm8960 register cache
* We can't read the WM8960 register space when we are
* using 2 wire for device control, so we cache them instead.
*/
static const struct reg_default wm8960_reg_defaults[] = {
{ 0x0, 0x00a7 },
{ 0x1, 0x00a7 },
{ 0x2, 0x0000 },
{ 0x3, 0x0000 },
{ 0x4, 0x0000 },
{ 0x5, 0x0008 },
{ 0x6, 0x0000 },
{ 0x7, 0x000a },
{ 0x8, 0x01c0 },
{ 0x9, 0x0000 },
{ 0xa, 0x00ff },
{ 0xb, 0x00ff },
{ 0x10, 0x0000 },
{ 0x11, 0x007b },
{ 0x12, 0x0100 },
{ 0x13, 0x0032 },
{ 0x14, 0x0000 },
{ 0x15, 0x00c3 },
{ 0x16, 0x00c3 },
{ 0x17, 0x01c0 },
{ 0x18, 0x0000 },
{ 0x19, 0x0000 },
{ 0x1a, 0x0000 },
{ 0x1b, 0x0000 },
{ 0x1c, 0x0000 },
{ 0x1d, 0x0000 },
{ 0x20, 0x0100 },
{ 0x21, 0x0100 },
{ 0x22, 0x0050 },
{ 0x25, 0x0050 },
{ 0x26, 0x0000 },
{ 0x27, 0x0000 },
{ 0x28, 0x0000 },
{ 0x29, 0x0000 },
{ 0x2a, 0x0040 },
{ 0x2b, 0x0000 },
{ 0x2c, 0x0000 },
{ 0x2d, 0x0050 },
{ 0x2e, 0x0050 },
{ 0x2f, 0x0000 },
{ 0x30, 0x0002 },
{ 0x31, 0x0037 },
{ 0x33, 0x0080 },
{ 0x34, 0x0008 },
{ 0x35, 0x0031 },
{ 0x36, 0x0026 },
{ 0x37, 0x00e9 },
};
static bool wm8960_volatile(struct device *dev, unsigned int reg)
{
switch (reg) {
case WM8960_RESET:
return true;
default:
return false;
}
}
struct wm8960_priv {
struct clk *mclk;
struct regmap *regmap;
int (*set_bias_level)(struct snd_soc_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>
2010-03-18 04:15:21 +08:00
enum snd_soc_bias_level level);
struct snd_soc_dapm_widget *lout1;
struct snd_soc_dapm_widget *rout1;
struct snd_soc_dapm_widget *out3;
bool deemph;
int lrclk;
int bclk;
int sysclk;
int clk_id;
int freq_in;
bool is_stream_in_use[2];
struct wm8960_data pdata;
};
#define wm8960_reset(c) regmap_write(c, WM8960_RESET, 0)
/* enumerated controls */
static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted",
"Right Inverted", "Stereo Inversion"};
static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"};
static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"};
static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"};
static const char *wm8960_alcmode[] = {"ALC", "Limiter"};
static const char *wm8960_adc_data_output_sel[] = {
"Left Data = Left ADC; Right Data = Right ADC",
"Left Data = Left ADC; Right Data = Left ADC",
"Left Data = Right ADC; Right Data = Right ADC",
"Left Data = Right ADC; Right Data = Left ADC",
};
static const char *wm8960_dmonomix[] = {"Stereo", "Mono"};
static const struct soc_enum wm8960_enum[] = {
SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity),
SOC_ENUM_SINGLE(WM8960_DACCTL2, 5, 4, wm8960_polarity),
SOC_ENUM_SINGLE(WM8960_3D, 6, 2, wm8960_3d_upper_cutoff),
SOC_ENUM_SINGLE(WM8960_3D, 5, 2, wm8960_3d_lower_cutoff),
SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc),
SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode),
SOC_ENUM_SINGLE(WM8960_ADDCTL1, 2, 4, wm8960_adc_data_output_sel),
SOC_ENUM_SINGLE(WM8960_ADDCTL1, 4, 2, wm8960_dmonomix),
};
static const int deemph_settings[] = { 0, 32000, 44100, 48000 };
static int wm8960_set_deemph(struct snd_soc_component *component)
{
struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component);
int val, i, best;
/* If we're using deemphasis select the nearest available sample
* rate.
*/
if (wm8960->deemph) {
best = 1;
for (i = 2; i < ARRAY_SIZE(deemph_settings); i++) {
if (abs(deemph_settings[i] - wm8960->lrclk) <
abs(deemph_settings[best] - wm8960->lrclk))
best = i;
}
val = best << 1;
} else {
val = 0;
}
dev_dbg(component->dev, "Set deemphasis %d\n", val);
return snd_soc_component_update_bits(component, WM8960_DACCTL1,
0x6, val);
}
static int wm8960_get_deemph(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component);
ucontrol->value.integer.value[0] = wm8960->deemph;
return 0;
}
static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component);
unsigned int deemph = ucontrol->value.integer.value[0];
if (deemph > 1)
return -EINVAL;
wm8960->deemph = deemph;
return wm8960_set_deemph(component);
}
static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1725, 75, 0);
static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
static const DECLARE_TLV_DB_SCALE(lineinboost_tlv, -1500, 300, 1);
static const SNDRV_CTL_TLVD_DECLARE_DB_RANGE(micboost_tlv,
0, 1, TLV_DB_SCALE_ITEM(0, 1300, 0),
2, 3, TLV_DB_SCALE_ITEM(2000, 900, 0),
);
static const struct snd_kcontrol_new wm8960_snd_controls[] = {
SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
0, 63, 0, inpga_tlv),
SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
6, 1, 0),
SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
7, 1, 1),
SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume",
WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv),
SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume",
WM8960_INBMIX1, 1, 7, 0, lineinboost_tlv),
SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
WM8960_INBMIX2, 4, 7, 0, lineinboost_tlv),
SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume",
WM8960_INBMIX2, 1, 7, 0, lineinboost_tlv),
SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT1 Volume",
WM8960_RINPATH, 4, 3, 0, micboost_tlv),
SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT1 Volume",
WM8960_LINPATH, 4, 3, 0, micboost_tlv),
SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC,
0, 255, 0, dac_tlv),
SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8960_LOUT1, WM8960_ROUT1,
0, 127, 0, out_tlv),
SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8960_LOUT1, WM8960_ROUT1,
7, 1, 0),
SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8960_LOUT2, WM8960_ROUT2,
0, 127, 0, out_tlv),
SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8960_LOUT2, WM8960_ROUT2,
7, 1, 0),
SOC_SINGLE("Speaker DC Volume", WM8960_CLASSD3, 3, 5, 0),
SOC_SINGLE("Speaker AC Volume", WM8960_CLASSD3, 0, 5, 0),
SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0),
SOC_ENUM("ADC Polarity", wm8960_enum[0]),
SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0),
SOC_ENUM("DAC Polarity", wm8960_enum[1]),
SOC_SINGLE_BOOL_EXT("DAC Deemphasis Switch", 0,
wm8960_get_deemph, wm8960_put_deemph),
SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[2]),
SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[3]),
SOC_SINGLE("3D Volume", WM8960_3D, 1, 15, 0),
SOC_SINGLE("3D Switch", WM8960_3D, 0, 1, 0),
SOC_ENUM("ALC Function", wm8960_enum[4]),
SOC_SINGLE("ALC Max Gain", WM8960_ALC1, 4, 7, 0),
SOC_SINGLE("ALC Target", WM8960_ALC1, 0, 15, 1),
SOC_SINGLE("ALC Min Gain", WM8960_ALC2, 4, 7, 0),
SOC_SINGLE("ALC Hold Time", WM8960_ALC2, 0, 15, 0),
SOC_ENUM("ALC Mode", wm8960_enum[5]),
SOC_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0),
SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0),
SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0),
SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0),
SOC_DOUBLE_R_TLV("ADC PCM Capture Volume", WM8960_LADC, WM8960_RADC,
0, 255, 0, adc_tlv),
SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume",
WM8960_BYPASS1, 4, 7, 1, bypass_tlv),
SOC_SINGLE_TLV("Left Output Mixer LINPUT3 Volume",
WM8960_LOUTMIX, 4, 7, 1, bypass_tlv),
SOC_SINGLE_TLV("Right Output Mixer Boost Bypass Volume",
WM8960_BYPASS2, 4, 7, 1, bypass_tlv),
SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume",
WM8960_ROUTMIX, 4, 7, 1, bypass_tlv),
SOC_ENUM("ADC Data Output Select", wm8960_enum[6]),
SOC_ENUM("DAC Mono Mix", wm8960_enum[7]),
};
static const struct snd_kcontrol_new wm8960_lin_boost[] = {
SOC_DAPM_SINGLE("LINPUT2 Switch", WM8960_LINPATH, 6, 1, 0),
SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LINPATH, 7, 1, 0),
SOC_DAPM_SINGLE("LINPUT1 Switch", WM8960_LINPATH, 8, 1, 0),
};
static const struct snd_kcontrol_new wm8960_lin[] = {
SOC_DAPM_SINGLE("Boost Switch", WM8960_LINPATH, 3, 1, 0),
};
static const struct snd_kcontrol_new wm8960_rin_boost[] = {
SOC_DAPM_SINGLE("RINPUT2 Switch", WM8960_RINPATH, 6, 1, 0),
SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_RINPATH, 7, 1, 0),
SOC_DAPM_SINGLE("RINPUT1 Switch", WM8960_RINPATH, 8, 1, 0),
};
static const struct snd_kcontrol_new wm8960_rin[] = {
SOC_DAPM_SINGLE("Boost Switch", WM8960_RINPATH, 3, 1, 0),
};
static const struct snd_kcontrol_new wm8960_loutput_mixer[] = {
SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_LOUTMIX, 8, 1, 0),
SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LOUTMIX, 7, 1, 0),
SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS1, 7, 1, 0),
};
static const struct snd_kcontrol_new wm8960_routput_mixer[] = {
SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_ROUTMIX, 8, 1, 0),
SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_ROUTMIX, 7, 1, 0),
SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS2, 7, 1, 0),
};
static const struct snd_kcontrol_new wm8960_mono_out[] = {
SOC_DAPM_SINGLE("Left Switch", WM8960_MONOMIX1, 7, 1, 0),
SOC_DAPM_SINGLE("Right Switch", WM8960_MONOMIX2, 7, 1, 0),
};
static const struct snd_soc_dapm_widget wm8960_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("LINPUT1"),
SND_SOC_DAPM_INPUT("RINPUT1"),
SND_SOC_DAPM_INPUT("LINPUT2"),
SND_SOC_DAPM_INPUT("RINPUT2"),
SND_SOC_DAPM_INPUT("LINPUT3"),
SND_SOC_DAPM_INPUT("RINPUT3"),
SND_SOC_DAPM_SUPPLY("MICB", WM8960_POWER1, 1, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8960_POWER1, 5, 0,
wm8960_lin_boost, ARRAY_SIZE(wm8960_lin_boost)),
SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8960_POWER1, 4, 0,
wm8960_rin_boost, ARRAY_SIZE(wm8960_rin_boost)),
SND_SOC_DAPM_MIXER("Left Input Mixer", WM8960_POWER3, 5, 0,
wm8960_lin, ARRAY_SIZE(wm8960_lin)),
SND_SOC_DAPM_MIXER("Right Input Mixer", WM8960_POWER3, 4, 0,
wm8960_rin, ARRAY_SIZE(wm8960_rin)),
SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER1, 3, 0),
SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER1, 2, 0),
SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0),
SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0),
SND_SOC_DAPM_MIXER("Left Output Mixer", WM8960_POWER3, 3, 0,
&wm8960_loutput_mixer[0],
ARRAY_SIZE(wm8960_loutput_mixer)),
SND_SOC_DAPM_MIXER("Right Output Mixer", WM8960_POWER3, 2, 0,
&wm8960_routput_mixer[0],
ARRAY_SIZE(wm8960_routput_mixer)),
SND_SOC_DAPM_PGA("LOUT1 PGA", WM8960_POWER2, 6, 0, NULL, 0),
SND_SOC_DAPM_PGA("ROUT1 PGA", WM8960_POWER2, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("Left Speaker PGA", WM8960_POWER2, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Speaker PGA", WM8960_POWER2, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Speaker Output", WM8960_CLASSD1, 7, 0, NULL, 0),
SND_SOC_DAPM_PGA("Left Speaker Output", WM8960_CLASSD1, 6, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("SPK_LP"),
SND_SOC_DAPM_OUTPUT("SPK_LN"),
SND_SOC_DAPM_OUTPUT("HP_L"),
SND_SOC_DAPM_OUTPUT("HP_R"),
SND_SOC_DAPM_OUTPUT("SPK_RP"),
SND_SOC_DAPM_OUTPUT("SPK_RN"),
SND_SOC_DAPM_OUTPUT("OUT3"),
};
static const struct snd_soc_dapm_widget wm8960_dapm_widgets_out3[] = {
SND_SOC_DAPM_MIXER("Mono Output Mixer", WM8960_POWER2, 1, 0,
&wm8960_mono_out[0],
ARRAY_SIZE(wm8960_mono_out)),
};
/* Represent OUT3 as a PGA so that it gets turned on with LOUT1/ROUT1 */
static const struct snd_soc_dapm_widget wm8960_dapm_widgets_capless[] = {
SND_SOC_DAPM_PGA("OUT3 VMID", WM8960_POWER2, 1, 0, NULL, 0),
};
static const struct snd_soc_dapm_route audio_paths[] = {
{ "Left Boost Mixer", "LINPUT1 Switch", "LINPUT1" },
{ "Left Boost Mixer", "LINPUT2 Switch", "LINPUT2" },
{ "Left Boost Mixer", "LINPUT3 Switch", "LINPUT3" },
{ "Left Input Mixer", "Boost Switch", "Left Boost Mixer" },
{ "Left Input Mixer", "Boost Switch", "LINPUT1" }, /* Really Boost Switch */
{ "Left Input Mixer", NULL, "LINPUT2" },
{ "Left Input Mixer", NULL, "LINPUT3" },
{ "Right Boost Mixer", "RINPUT1 Switch", "RINPUT1" },
{ "Right Boost Mixer", "RINPUT2 Switch", "RINPUT2" },
{ "Right Boost Mixer", "RINPUT3 Switch", "RINPUT3" },
{ "Right Input Mixer", "Boost Switch", "Right Boost Mixer" },
{ "Right Input Mixer", "Boost Switch", "RINPUT1" }, /* Really Boost Switch */
{ "Right Input Mixer", NULL, "RINPUT2" },
{ "Right Input Mixer", NULL, "RINPUT3" },
{ "Left ADC", NULL, "Left Input Mixer" },
{ "Right ADC", NULL, "Right Input Mixer" },
{ "Left Output Mixer", "LINPUT3 Switch", "LINPUT3" },
{ "Left Output Mixer", "Boost Bypass Switch", "Left Boost Mixer" },
{ "Left Output Mixer", "PCM Playback Switch", "Left DAC" },
{ "Right Output Mixer", "RINPUT3 Switch", "RINPUT3" },
{ "Right Output Mixer", "Boost Bypass Switch", "Right Boost Mixer" },
{ "Right Output Mixer", "PCM Playback Switch", "Right DAC" },
{ "LOUT1 PGA", NULL, "Left Output Mixer" },
{ "ROUT1 PGA", NULL, "Right Output Mixer" },
{ "HP_L", NULL, "LOUT1 PGA" },
{ "HP_R", NULL, "ROUT1 PGA" },
{ "Left Speaker PGA", NULL, "Left Output Mixer" },
{ "Right Speaker PGA", NULL, "Right Output Mixer" },
{ "Left Speaker Output", NULL, "Left Speaker PGA" },
{ "Right Speaker Output", NULL, "Right Speaker PGA" },
{ "SPK_LN", NULL, "Left Speaker Output" },
{ "SPK_LP", NULL, "Left Speaker Output" },
{ "SPK_RN", NULL, "Right Speaker Output" },
{ "SPK_RP", NULL, "Right Speaker Output" },
};
static const struct snd_soc_dapm_route audio_paths_out3[] = {
{ "Mono Output Mixer", "Left Switch", "Left Output Mixer" },
{ "Mono Output Mixer", "Right Switch", "Right Output Mixer" },
{ "OUT3", NULL, "Mono Output Mixer", }
};
static const struct snd_soc_dapm_route audio_paths_capless[] = {
{ "HP_L", NULL, "OUT3 VMID" },
{ "HP_R", NULL, "OUT3 VMID" },
{ "OUT3 VMID", NULL, "Left Output Mixer" },
{ "OUT3 VMID", NULL, "Right Output Mixer" },
};
static int wm8960_add_widgets(struct snd_soc_component *component)
{
struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component);
struct wm8960_data *pdata = &wm8960->pdata;
struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
struct snd_soc_dapm_widget *w;
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets,
ARRAY_SIZE(wm8960_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 21:53:46 +08:00
snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
/* In capless mode OUT3 is used to provide VMID for the
* headphone outputs, otherwise it is used as a mono mixer.
*/
if (pdata && pdata->capless) {
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets_capless,
ARRAY_SIZE(wm8960_dapm_widgets_capless));
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
snd_soc_dapm_add_routes(dapm, audio_paths_capless,
ARRAY_SIZE(audio_paths_capless));
} else {
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
snd_soc_dapm_new_controls(dapm, wm8960_dapm_widgets_out3,
ARRAY_SIZE(wm8960_dapm_widgets_out3));
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
snd_soc_dapm_add_routes(dapm, audio_paths_out3,
ARRAY_SIZE(audio_paths_out3));
}
/* We need to power up the headphone output stage out of
* sequence for capless mode. To save scanning the widget
* list each time to find the desired power state do so now
* and save the result.
*/
list_for_each_entry(w, &component->card->widgets, list) {
if (w->dapm != dapm)
continue;
if (strcmp(w->name, "LOUT1 PGA") == 0)
wm8960->lout1 = w;
if (strcmp(w->name, "ROUT1 PGA") == 0)
wm8960->rout1 = w;
if (strcmp(w->name, "OUT3 VMID") == 0)
wm8960->out3 = w;
}
return 0;
}
static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_component *component = codec_dai->component;
u16 iface = 0;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
iface |= 0x0040;
break;
case SND_SOC_DAIFMT_CBS_CFS:
break;
default:
return -EINVAL;
}
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
iface |= 0x0002;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
iface |= 0x0001;
break;
case SND_SOC_DAIFMT_DSP_A:
iface |= 0x0003;
break;
case SND_SOC_DAIFMT_DSP_B:
iface |= 0x0013;
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 |= 0x0090;
break;
case SND_SOC_DAIFMT_IB_NF:
iface |= 0x0080;
break;
case SND_SOC_DAIFMT_NB_IF:
iface |= 0x0010;
break;
default:
return -EINVAL;
}
/* set iface */
snd_soc_component_write(component, WM8960_IFACE1, iface);
return 0;
}
static struct {
int rate;
unsigned int val;
} alc_rates[] = {
{ 48000, 0 },
{ 44100, 0 },
{ 32000, 1 },
{ 22050, 2 },
{ 24000, 2 },
{ 16000, 3 },
{ 11025, 4 },
{ 12000, 4 },
{ 8000, 5 },
};
/* -1 for reserved value */
static const int sysclk_divs[] = { 1, -1, 2, -1 };
/* Multiply 256 for internal 256 div */
static const int dac_divs[] = { 256, 384, 512, 768, 1024, 1408, 1536 };
/* Multiply 10 to eliminate decimials */
static const int bclk_divs[] = {
10, 15, 20, 30, 40, 55, 60, 80, 110,
120, 160, 220, 240, 320, 320, 320
};
/**
* wm8960_configure_sysclk - checks if there is a sysclk frequency available
* The sysclk must be chosen such that:
* - sysclk = MCLK / sysclk_divs
* - lrclk = sysclk / dac_divs
* - 10 * bclk = sysclk / bclk_divs
*
* If we cannot find an exact match for (sysclk, lrclk, bclk)
* triplet, we relax the bclk such that bclk is chosen as the
* closest available frequency greater than expected bclk.
*
* @wm8960_priv: wm8960 codec private data
* @mclk: MCLK used to derive sysclk
* @sysclk_idx: sysclk_divs index for found sysclk
* @dac_idx: dac_divs index for found lrclk
* @bclk_idx: bclk_divs index for found bclk
*
* Returns:
* -1, in case no sysclk frequency available found
* >=0, in case we could derive bclk and lrclk from sysclk using
* (@sysclk_idx, @dac_idx, @bclk_idx) dividers
*/
static
int wm8960_configure_sysclk(struct wm8960_priv *wm8960, int mclk,
int *sysclk_idx, int *dac_idx, int *bclk_idx)
{
int sysclk, bclk, lrclk;
int i, j, k;
int diff, closest = mclk;
/* marker for no match */
*bclk_idx = -1;
bclk = wm8960->bclk;
lrclk = wm8960->lrclk;
/* check if the sysclk frequency is available. */
for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) {
if (sysclk_divs[i] == -1)
continue;
sysclk = mclk / sysclk_divs[i];
for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) {
if (sysclk != dac_divs[j] * lrclk)
continue;
for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) {
diff = sysclk - bclk * bclk_divs[k] / 10;
if (diff == 0) {
*sysclk_idx = i;
*dac_idx = j;
*bclk_idx = k;
break;
}
if (diff > 0 && closest > diff) {
*sysclk_idx = i;
*dac_idx = j;
*bclk_idx = k;
closest = diff;
}
}
if (k != ARRAY_SIZE(bclk_divs))
break;
}
if (j != ARRAY_SIZE(dac_divs))
break;
}
return *bclk_idx;
}
/**
* wm8960_configure_pll - checks if there is a PLL out frequency available
* The PLL out frequency must be chosen such that:
* - sysclk = lrclk * dac_divs
* - freq_out = sysclk * sysclk_divs
* - 10 * sysclk = bclk * bclk_divs
*
* If we cannot find an exact match for (sysclk, lrclk, bclk)
* triplet, we relax the bclk such that bclk is chosen as the
* closest available frequency greater than expected bclk.
*
* @component: component structure
* @freq_in: input frequency used to derive freq out via PLL
* @sysclk_idx: sysclk_divs index for found sysclk
* @dac_idx: dac_divs index for found lrclk
* @bclk_idx: bclk_divs index for found bclk
*
* Returns:
* < 0, in case no PLL frequency out available was found
* >=0, in case we could derive bclk, lrclk, sysclk from PLL out using
* (@sysclk_idx, @dac_idx, @bclk_idx) dividers
*/
static
int wm8960_configure_pll(struct snd_soc_component *component, int freq_in,
int *sysclk_idx, int *dac_idx, int *bclk_idx)
{
struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component);
int sysclk, bclk, lrclk, freq_out;
int diff, closest, best_freq_out;
int i, j, k;
bclk = wm8960->bclk;
lrclk = wm8960->lrclk;
closest = freq_in;
best_freq_out = -EINVAL;
*sysclk_idx = *dac_idx = *bclk_idx = -1;
for (i = 0; i < ARRAY_SIZE(sysclk_divs); ++i) {
if (sysclk_divs[i] == -1)
continue;
for (j = 0; j < ARRAY_SIZE(dac_divs); ++j) {
sysclk = lrclk * dac_divs[j];
freq_out = sysclk * sysclk_divs[i];
for (k = 0; k < ARRAY_SIZE(bclk_divs); ++k) {
if (!is_pll_freq_available(freq_in, freq_out))
continue;
diff = sysclk - bclk * bclk_divs[k] / 10;
if (diff == 0) {
*sysclk_idx = i;
*dac_idx = j;
*bclk_idx = k;
return freq_out;
}
if (diff > 0 && closest > diff) {
*sysclk_idx = i;
*dac_idx = j;
*bclk_idx = k;
closest = diff;
best_freq_out = freq_out;
}
}
}
}
return best_freq_out;
}
static int wm8960_configure_clocking(struct snd_soc_component *component)
{
struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component);
int freq_out, freq_in;
u16 iface1 = snd_soc_component_read32(component, WM8960_IFACE1);
int i, j, k;
int ret;
if (!(iface1 & (1<<6))) {
dev_dbg(component->dev,
"Codec is slave mode, no need to configure clock\n");
return 0;
}
if (wm8960->clk_id != WM8960_SYSCLK_MCLK && !wm8960->freq_in) {
dev_err(component->dev, "No MCLK configured\n");
return -EINVAL;
}
freq_in = wm8960->freq_in;
/*
* If it's sysclk auto mode, check if the MCLK can provide sysclk or
* not. If MCLK can provide sysclk, using MCLK to provide sysclk
* directly. Otherwise, auto select a available pll out frequency
* and set PLL.
*/
if (wm8960->clk_id == WM8960_SYSCLK_AUTO) {
/* disable the PLL and using MCLK to provide sysclk */
wm8960_set_pll(component, 0, 0);
freq_out = freq_in;
} else if (wm8960->sysclk) {
freq_out = wm8960->sysclk;
} else {
dev_err(component->dev, "No SYSCLK configured\n");
return -EINVAL;
}
if (wm8960->clk_id != WM8960_SYSCLK_PLL) {
ret = wm8960_configure_sysclk(wm8960, freq_out, &i, &j, &k);
if (ret >= 0) {
goto configure_clock;
} else if (wm8960->clk_id != WM8960_SYSCLK_AUTO) {
dev_err(component->dev, "failed to configure clock\n");
return -EINVAL;
}
}
freq_out = wm8960_configure_pll(component, freq_in, &i, &j, &k);
if (freq_out < 0) {
dev_err(component->dev, "failed to configure clock via PLL\n");
return freq_out;
}
wm8960_set_pll(component, freq_in, freq_out);
configure_clock:
/* configure sysclk clock */
snd_soc_component_update_bits(component, WM8960_CLOCK1, 3 << 1, i << 1);
/* configure frame clock */
snd_soc_component_update_bits(component, WM8960_CLOCK1, 0x7 << 3, j << 3);
snd_soc_component_update_bits(component, WM8960_CLOCK1, 0x7 << 6, j << 6);
/* configure bit clock */
snd_soc_component_update_bits(component, WM8960_CLOCK2, 0xf, k);
return 0;
}
static int wm8960_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 wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component);
u16 iface = snd_soc_component_read32(component, WM8960_IFACE1) & 0xfff3;
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
int i;
wm8960->bclk = snd_soc_params_to_bclk(params);
if (params_channels(params) == 1)
wm8960->bclk *= 2;
/* bit size */
switch (params_width(params)) {
case 16:
break;
case 20:
iface |= 0x0004;
break;
case 24:
iface |= 0x0008;
break;
case 32:
/* right justify mode does not support 32 word length */
if ((iface & 0x3) != 0) {
iface |= 0x000c;
break;
}
/* fall through */
default:
dev_err(component->dev, "unsupported width %d\n",
params_width(params));
return -EINVAL;
}
wm8960->lrclk = params_rate(params);
/* Update filters for the new rate */
if (tx) {
wm8960_set_deemph(component);
} else {
for (i = 0; i < ARRAY_SIZE(alc_rates); i++)
if (alc_rates[i].rate == params_rate(params))
snd_soc_component_update_bits(component,
WM8960_ADDCTL3, 0x7,
alc_rates[i].val);
}
/* set iface */
snd_soc_component_write(component, WM8960_IFACE1, iface);
wm8960->is_stream_in_use[tx] = true;
if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_ON &&
!wm8960->is_stream_in_use[!tx])
return wm8960_configure_clocking(component);
return 0;
}
static int wm8960_hw_free(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component);
bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
wm8960->is_stream_in_use[tx] = false;
return 0;
}
static int wm8960_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_component *component = dai->component;
if (mute)
snd_soc_component_update_bits(component, WM8960_DACCTL1, 0x8, 0x8);
else
snd_soc_component_update_bits(component, WM8960_DACCTL1, 0x8, 0);
return 0;
}
static int wm8960_set_bias_level_out3(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component);
u16 pm2 = snd_soc_component_read32(component, WM8960_POWER2);
int ret;
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
switch (snd_soc_component_get_bias_level(component)) {
case SND_SOC_BIAS_STANDBY:
if (!IS_ERR(wm8960->mclk)) {
ret = clk_prepare_enable(wm8960->mclk);
if (ret) {
dev_err(component->dev,
"Failed to enable MCLK: %d\n",
ret);
return ret;
}
}
ret = wm8960_configure_clocking(component);
if (ret)
return ret;
/* Set VMID to 2x50k */
snd_soc_component_update_bits(component, WM8960_POWER1, 0x180, 0x80);
break;
case SND_SOC_BIAS_ON:
/*
* If it's sysclk auto mode, and the pll is enabled,
* disable the pll
*/
if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1))
wm8960_set_pll(component, 0, 0);
if (!IS_ERR(wm8960->mclk))
clk_disable_unprepare(wm8960->mclk);
break;
default:
break;
}
break;
case SND_SOC_BIAS_STANDBY:
if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) {
regcache_sync(wm8960->regmap);
/* Enable anti-pop features */
snd_soc_component_write(component, WM8960_APOP1,
WM8960_POBCTRL | WM8960_SOFT_ST |
WM8960_BUFDCOPEN | WM8960_BUFIOEN);
/* Enable & ramp VMID at 2x50k */
snd_soc_component_update_bits(component, WM8960_POWER1, 0x80, 0x80);
msleep(100);
/* Enable VREF */
snd_soc_component_update_bits(component, WM8960_POWER1, WM8960_VREF,
WM8960_VREF);
/* Disable anti-pop features */
snd_soc_component_write(component, WM8960_APOP1, WM8960_BUFIOEN);
}
/* Set VMID to 2x250k */
snd_soc_component_update_bits(component, WM8960_POWER1, 0x180, 0x100);
break;
case SND_SOC_BIAS_OFF:
/* Enable anti-pop features */
snd_soc_component_write(component, WM8960_APOP1,
WM8960_POBCTRL | WM8960_SOFT_ST |
WM8960_BUFDCOPEN | WM8960_BUFIOEN);
/* Disable VMID and VREF, let them discharge */
snd_soc_component_write(component, WM8960_POWER1, 0);
msleep(600);
break;
}
return 0;
}
static int wm8960_set_bias_level_capless(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component);
u16 pm2 = snd_soc_component_read32(component, WM8960_POWER2);
int reg, ret;
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
switch (snd_soc_component_get_bias_level(component)) {
case SND_SOC_BIAS_STANDBY:
/* Enable anti pop mode */
snd_soc_component_update_bits(component, WM8960_APOP1,
WM8960_POBCTRL | WM8960_SOFT_ST |
WM8960_BUFDCOPEN,
WM8960_POBCTRL | WM8960_SOFT_ST |
WM8960_BUFDCOPEN);
/* Enable LOUT1, ROUT1 and OUT3 if they're enabled */
reg = 0;
if (wm8960->lout1 && wm8960->lout1->power)
reg |= WM8960_PWR2_LOUT1;
if (wm8960->rout1 && wm8960->rout1->power)
reg |= WM8960_PWR2_ROUT1;
if (wm8960->out3 && wm8960->out3->power)
reg |= WM8960_PWR2_OUT3;
snd_soc_component_update_bits(component, WM8960_POWER2,
WM8960_PWR2_LOUT1 |
WM8960_PWR2_ROUT1 |
WM8960_PWR2_OUT3, reg);
/* Enable VMID at 2*50k */
snd_soc_component_update_bits(component, WM8960_POWER1,
WM8960_VMID_MASK, 0x80);
/* Ramp */
msleep(100);
/* Enable VREF */
snd_soc_component_update_bits(component, WM8960_POWER1,
WM8960_VREF, WM8960_VREF);
msleep(100);
if (!IS_ERR(wm8960->mclk)) {
ret = clk_prepare_enable(wm8960->mclk);
if (ret) {
dev_err(component->dev,
"Failed to enable MCLK: %d\n",
ret);
return ret;
}
}
ret = wm8960_configure_clocking(component);
if (ret)
return ret;
break;
case SND_SOC_BIAS_ON:
/*
* If it's sysclk auto mode, and the pll is enabled,
* disable the pll
*/
if (wm8960->clk_id == WM8960_SYSCLK_AUTO && (pm2 & 0x1))
wm8960_set_pll(component, 0, 0);
if (!IS_ERR(wm8960->mclk))
clk_disable_unprepare(wm8960->mclk);
/* Enable anti-pop mode */
snd_soc_component_update_bits(component, WM8960_APOP1,
WM8960_POBCTRL | WM8960_SOFT_ST |
WM8960_BUFDCOPEN,
WM8960_POBCTRL | WM8960_SOFT_ST |
WM8960_BUFDCOPEN);
/* Disable VMID and VREF */
snd_soc_component_update_bits(component, WM8960_POWER1,
WM8960_VREF | WM8960_VMID_MASK, 0);
break;
case SND_SOC_BIAS_OFF:
regcache_sync(wm8960->regmap);
break;
default:
break;
}
break;
case SND_SOC_BIAS_STANDBY:
switch (snd_soc_component_get_bias_level(component)) {
case SND_SOC_BIAS_PREPARE:
/* Disable HP discharge */
snd_soc_component_update_bits(component, WM8960_APOP2,
WM8960_DISOP | WM8960_DRES_MASK,
0);
/* Disable anti-pop features */
snd_soc_component_update_bits(component, WM8960_APOP1,
WM8960_POBCTRL | WM8960_SOFT_ST |
WM8960_BUFDCOPEN,
WM8960_POBCTRL | WM8960_SOFT_ST |
WM8960_BUFDCOPEN);
break;
default:
break;
}
break;
case SND_SOC_BIAS_OFF:
break;
}
return 0;
}
/* PLL divisors */
struct _pll_div {
u32 pre_div:1;
u32 n:4;
u32 k:24;
};
static bool is_pll_freq_available(unsigned int source, unsigned int target)
{
unsigned int Ndiv;
if (source == 0 || target == 0)
return false;
/* Scale up target to PLL operating frequency */
target *= 4;
Ndiv = target / source;
if (Ndiv < 6) {
source >>= 1;
Ndiv = target / source;
}
if ((Ndiv < 6) || (Ndiv > 12))
return false;
return true;
}
/* The size in bits of the pll divide multiplied by 10
* to allow rounding later */
#define FIXED_PLL_SIZE ((1 << 24) * 10)
static int pll_factors(unsigned int source, unsigned int target,
struct _pll_div *pll_div)
{
unsigned long long Kpart;
unsigned int K, Ndiv, Nmod;
pr_debug("WM8960 PLL: setting %dHz->%dHz\n", source, target);
/* Scale up target to PLL operating frequency */
target *= 4;
Ndiv = target / source;
if (Ndiv < 6) {
source >>= 1;
pll_div->pre_div = 1;
Ndiv = target / source;
} else
pll_div->pre_div = 0;
if ((Ndiv < 6) || (Ndiv > 12)) {
pr_err("WM8960 PLL: Unsupported N=%d\n", Ndiv);
return -EINVAL;
}
pll_div->n = Ndiv;
Nmod = target % source;
Kpart = FIXED_PLL_SIZE * (long long)Nmod;
do_div(Kpart, source);
K = Kpart & 0xFFFFFFFF;
/* Check if we need to round */
if ((K % 10) >= 5)
K += 5;
/* Move down to proper range now rounding is done */
K /= 10;
pll_div->k = K;
pr_debug("WM8960 PLL: N=%x K=%x pre_div=%d\n",
pll_div->n, pll_div->k, pll_div->pre_div);
return 0;
}
static int wm8960_set_pll(struct snd_soc_component *component,
unsigned int freq_in, unsigned int freq_out)
{
u16 reg;
static struct _pll_div pll_div;
int ret;
if (freq_in && freq_out) {
ret = pll_factors(freq_in, freq_out, &pll_div);
if (ret != 0)
return ret;
}
/* Disable the PLL: even if we are changing the frequency the
* PLL needs to be disabled while we do so. */
snd_soc_component_update_bits(component, WM8960_CLOCK1, 0x1, 0);
snd_soc_component_update_bits(component, WM8960_POWER2, 0x1, 0);
if (!freq_in || !freq_out)
return 0;
reg = snd_soc_component_read32(component, WM8960_PLL1) & ~0x3f;
reg |= pll_div.pre_div << 4;
reg |= pll_div.n;
if (pll_div.k) {
reg |= 0x20;
snd_soc_component_write(component, WM8960_PLL2, (pll_div.k >> 16) & 0xff);
snd_soc_component_write(component, WM8960_PLL3, (pll_div.k >> 8) & 0xff);
snd_soc_component_write(component, WM8960_PLL4, pll_div.k & 0xff);
}
snd_soc_component_write(component, WM8960_PLL1, reg);
/* Turn it on */
snd_soc_component_update_bits(component, WM8960_POWER2, 0x1, 0x1);
msleep(250);
snd_soc_component_update_bits(component, WM8960_CLOCK1, 0x1, 0x1);
return 0;
}
static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
int source, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_component *component = codec_dai->component;
struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component);
wm8960->freq_in = freq_in;
if (pll_id == WM8960_SYSCLK_AUTO)
return 0;
return wm8960_set_pll(component, freq_in, freq_out);
}
static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
int div_id, int div)
{
struct snd_soc_component *component = codec_dai->component;
u16 reg;
switch (div_id) {
case WM8960_SYSCLKDIV:
reg = snd_soc_component_read32(component, WM8960_CLOCK1) & 0x1f9;
snd_soc_component_write(component, WM8960_CLOCK1, reg | div);
break;
case WM8960_DACDIV:
reg = snd_soc_component_read32(component, WM8960_CLOCK1) & 0x1c7;
snd_soc_component_write(component, WM8960_CLOCK1, reg | div);
break;
case WM8960_OPCLKDIV:
reg = snd_soc_component_read32(component, WM8960_PLL1) & 0x03f;
snd_soc_component_write(component, WM8960_PLL1, reg | div);
break;
case WM8960_DCLKDIV:
reg = snd_soc_component_read32(component, WM8960_CLOCK2) & 0x03f;
snd_soc_component_write(component, WM8960_CLOCK2, reg | div);
break;
case WM8960_TOCLKSEL:
reg = snd_soc_component_read32(component, WM8960_ADDCTL1) & 0x1fd;
snd_soc_component_write(component, WM8960_ADDCTL1, reg | div);
break;
default:
return -EINVAL;
}
return 0;
}
static int wm8960_set_bias_level(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>
2010-03-18 04:15:21 +08:00
enum snd_soc_bias_level level)
{
struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(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>
2010-03-18 04:15:21 +08:00
return wm8960->set_bias_level(component, level);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
}
static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
struct snd_soc_component *component = dai->component;
struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component);
switch (clk_id) {
case WM8960_SYSCLK_MCLK:
snd_soc_component_update_bits(component, WM8960_CLOCK1,
0x1, WM8960_SYSCLK_MCLK);
break;
case WM8960_SYSCLK_PLL:
snd_soc_component_update_bits(component, WM8960_CLOCK1,
0x1, WM8960_SYSCLK_PLL);
break;
case WM8960_SYSCLK_AUTO:
break;
default:
return -EINVAL;
}
wm8960->sysclk = freq;
wm8960->clk_id = clk_id;
return 0;
}
#define WM8960_RATES SNDRV_PCM_RATE_8000_48000
#define WM8960_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 wm8960_dai_ops = {
.hw_params = wm8960_hw_params,
.hw_free = wm8960_hw_free,
.digital_mute = wm8960_mute,
.set_fmt = wm8960_set_dai_fmt,
.set_clkdiv = wm8960_set_dai_clkdiv,
.set_pll = wm8960_set_dai_pll,
.set_sysclk = wm8960_set_dai_sysclk,
};
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
static struct snd_soc_dai_driver wm8960_dai = {
.name = "wm8960-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = WM8960_RATES,
.formats = WM8960_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8960_RATES,
.formats = WM8960_FORMATS,},
.ops = &wm8960_dai_ops,
.symmetric_rates = 1,
};
static int wm8960_probe(struct snd_soc_component *component)
{
struct wm8960_priv *wm8960 = snd_soc_component_get_drvdata(component);
struct wm8960_data *pdata = &wm8960->pdata;
if (pdata->capless)
wm8960->set_bias_level = wm8960_set_bias_level_capless;
else
wm8960->set_bias_level = wm8960_set_bias_level_out3;
snd_soc_add_component_controls(component, wm8960_snd_controls,
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
ARRAY_SIZE(wm8960_snd_controls));
wm8960_add_widgets(component);
return 0;
}
static const struct snd_soc_component_driver soc_component_dev_wm8960 = {
.probe = wm8960_probe,
.set_bias_level = wm8960_set_bias_level,
.suspend_bias_off = 1,
.idle_bias_on = 1,
.use_pmdown_time = 1,
.endianness = 1,
.non_legacy_dai_naming = 1,
};
static const struct regmap_config wm8960_regmap = {
.reg_bits = 7,
.val_bits = 9,
.max_register = WM8960_PLL4,
.reg_defaults = wm8960_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(wm8960_reg_defaults),
.cache_type = REGCACHE_RBTREE,
.volatile_reg = wm8960_volatile,
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
};
static void wm8960_set_pdata_from_of(struct i2c_client *i2c,
struct wm8960_data *pdata)
{
const struct device_node *np = i2c->dev.of_node;
if (of_property_read_bool(np, "wlf,capless"))
pdata->capless = true;
if (of_property_read_bool(np, "wlf,shared-lrclk"))
pdata->shared_lrclk = true;
}
static int wm8960_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct wm8960_data *pdata = dev_get_platdata(&i2c->dev);
struct wm8960_priv *wm8960;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 04:15:21 +08:00
int ret;
wm8960 = devm_kzalloc(&i2c->dev, sizeof(struct wm8960_priv),
GFP_KERNEL);
if (wm8960 == NULL)
return -ENOMEM;
wm8960->mclk = devm_clk_get(&i2c->dev, "mclk");
if (IS_ERR(wm8960->mclk)) {
if (PTR_ERR(wm8960->mclk) == -EPROBE_DEFER)
return -EPROBE_DEFER;
}
wm8960->regmap = devm_regmap_init_i2c(i2c, &wm8960_regmap);
if (IS_ERR(wm8960->regmap))
return PTR_ERR(wm8960->regmap);
if (pdata)
memcpy(&wm8960->pdata, pdata, sizeof(struct wm8960_data));
else if (i2c->dev.of_node)
wm8960_set_pdata_from_of(i2c, &wm8960->pdata);
ret = wm8960_reset(wm8960->regmap);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to issue reset\n");
return ret;
}
if (wm8960->pdata.shared_lrclk) {
ret = regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2,
0x4, 0x4);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to enable LRCM: %d\n",
ret);
return ret;
}
}
/* Latch the update bits */
regmap_update_bits(wm8960->regmap, WM8960_LINVOL, 0x100, 0x100);
regmap_update_bits(wm8960->regmap, WM8960_RINVOL, 0x100, 0x100);
regmap_update_bits(wm8960->regmap, WM8960_LADC, 0x100, 0x100);
regmap_update_bits(wm8960->regmap, WM8960_RADC, 0x100, 0x100);
regmap_update_bits(wm8960->regmap, WM8960_LDAC, 0x100, 0x100);
regmap_update_bits(wm8960->regmap, WM8960_RDAC, 0x100, 0x100);
regmap_update_bits(wm8960->regmap, WM8960_LOUT1, 0x100, 0x100);
regmap_update_bits(wm8960->regmap, WM8960_ROUT1, 0x100, 0x100);
regmap_update_bits(wm8960->regmap, WM8960_LOUT2, 0x100, 0x100);
regmap_update_bits(wm8960->regmap, WM8960_ROUT2, 0x100, 0x100);
i2c_set_clientdata(i2c, wm8960);
ret = devm_snd_soc_register_component(&i2c->dev,
&soc_component_dev_wm8960, &wm8960_dai, 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>
2010-03-18 04:15:21 +08:00
return ret;
}
static int wm8960_i2c_remove(struct i2c_client *client)
{
return 0;
}
static const struct i2c_device_id wm8960_i2c_id[] = {
{ "wm8960", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id);
static const struct of_device_id wm8960_of_match[] = {
{ .compatible = "wlf,wm8960", },
{ }
};
MODULE_DEVICE_TABLE(of, wm8960_of_match);
static struct i2c_driver wm8960_i2c_driver = {
.driver = {
.name = "wm8960",
.of_match_table = wm8960_of_match,
},
.probe = wm8960_i2c_probe,
.remove = wm8960_i2c_remove,
.id_table = wm8960_i2c_id,
};
module_i2c_driver(wm8960_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8960 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");