WSL2-Linux-Kernel/sound/soc/codecs/wm8753.c

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// SPDX-License-Identifier: GPL-2.0-or-later
/*
* wm8753.c -- WM8753 ALSA Soc Audio driver
*
* Copyright 2003-11 Wolfson Microelectronics PLC.
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* Notes:
* The WM8753 is a low power, high quality stereo codec with integrated PCM
* codec designed for portable digital telephony applications.
*
* Dual DAI:-
*
* This driver support 2 DAI PCM's. This makes the default PCM available for
* HiFi audio (e.g. MP3, ogg) playback/capture and the other PCM available for
* voice.
*
* Please note that the voice PCM can be connected directly to a Bluetooth
* codec or GSM modem and thus cannot be read or written to, although it is
* available to be configured with snd_hw_params(), etc and kcontrols in the
* normal alsa manner.
*
* Fast DAI switching:-
*
* The driver can now fast switch between the DAI configurations via a
* an alsa kcontrol. This allows the PCM to remain open.
*/
#include <linux/module.h>
#include <linux/kernel.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/of_device.h>
#include <linux/regmap.h>
#include <linux/spi/spi.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 11:04:11 +03:00
#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 "wm8753.h"
static int caps_charge = 2000;
module_param(caps_charge, int, 0);
MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
static int wm8753_hifi_write_dai_fmt(struct snd_soc_component *component,
unsigned int fmt);
static int wm8753_voice_write_dai_fmt(struct snd_soc_component *component,
unsigned int fmt);
/*
* wm8753 register cache
* We can't read the WM8753 register space when we
* are using 2 wire for device control, so we cache them instead.
*/
static const struct reg_default wm8753_reg_defaults[] = {
{ 0x00, 0x0000 },
{ 0x01, 0x0008 },
{ 0x02, 0x0000 },
{ 0x03, 0x000a },
{ 0x04, 0x000a },
{ 0x05, 0x0033 },
{ 0x06, 0x0000 },
{ 0x07, 0x0007 },
{ 0x08, 0x00ff },
{ 0x09, 0x00ff },
{ 0x0a, 0x000f },
{ 0x0b, 0x000f },
{ 0x0c, 0x007b },
{ 0x0d, 0x0000 },
{ 0x0e, 0x0032 },
{ 0x0f, 0x0000 },
{ 0x10, 0x00c3 },
{ 0x11, 0x00c3 },
{ 0x12, 0x00c0 },
{ 0x13, 0x0000 },
{ 0x14, 0x0000 },
{ 0x15, 0x0000 },
{ 0x16, 0x0000 },
{ 0x17, 0x0000 },
{ 0x18, 0x0000 },
{ 0x19, 0x0000 },
{ 0x1a, 0x0000 },
{ 0x1b, 0x0000 },
{ 0x1c, 0x0000 },
{ 0x1d, 0x0000 },
{ 0x1e, 0x0000 },
{ 0x1f, 0x0000 },
{ 0x20, 0x0055 },
{ 0x21, 0x0005 },
{ 0x22, 0x0050 },
{ 0x23, 0x0055 },
{ 0x24, 0x0050 },
{ 0x25, 0x0055 },
{ 0x26, 0x0050 },
{ 0x27, 0x0055 },
{ 0x28, 0x0079 },
{ 0x29, 0x0079 },
{ 0x2a, 0x0079 },
{ 0x2b, 0x0079 },
{ 0x2c, 0x0079 },
{ 0x2d, 0x0000 },
{ 0x2e, 0x0000 },
{ 0x2f, 0x0000 },
{ 0x30, 0x0000 },
{ 0x31, 0x0097 },
{ 0x32, 0x0097 },
{ 0x33, 0x0000 },
{ 0x34, 0x0004 },
{ 0x35, 0x0000 },
{ 0x36, 0x0083 },
{ 0x37, 0x0024 },
{ 0x38, 0x01ba },
{ 0x39, 0x0000 },
{ 0x3a, 0x0083 },
{ 0x3b, 0x0024 },
{ 0x3c, 0x01ba },
{ 0x3d, 0x0000 },
{ 0x3e, 0x0000 },
{ 0x3f, 0x0000 },
};
static bool wm8753_volatile(struct device *dev, unsigned int reg)
{
return reg == WM8753_RESET;
}
/* codec private data */
struct wm8753_priv {
struct regmap *regmap;
unsigned int sysclk;
unsigned int pcmclk;
unsigned int voice_fmt;
unsigned int hifi_fmt;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
int dai_func;
ASoC: wm8753: Integrate capacitor charging into the DAPM sequence When being powered on, either initially on probe or when resuming from suspend, the wm8971 configures the device for quick output capacitor charging. Since the charging can take a rather long time (up to multiple seconds) it is done asynchronously without blocking. A delayed work item is run once the charging is finished and the device is switched to the target bias level. This all done asynchronously to the regular DAPM sequence accessing the same data structures and registers without any looking, which can lead to race conditions. Furthermore this potentially delays the start of stream on the CODEC while the rest of the system is already up and running, meaning the first bytes of audio are lost. It also does no comply with the assumption of the DAPM core that if set_bias_level() returned successfully the device will be at the requested bias level. This patch slightly refactors things and makes sure that the caps charging is properly integrated into the DAPM sequence. When transitioning from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY the part will be put into fast charging mode and a work item will be scheduled that puts it back into standby charging once the charging period has elapsed. If a playback or capture stream is started while charging is in progress the driver will now wait in SND_SOC_BIAS_PREPARE until the charging is done. This makes sure that charging is done asynchronously in the background when the chip is idle, but at the same time makes sure that playback/capture is not started before the charging is done. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-03-30 22:04:48 +03:00
struct delayed_work charge_work;
};
#define wm8753_reset(c) snd_soc_component_write(c, WM8753_RESET, 0)
/*
* WM8753 Controls
*/
static const char *wm8753_base[] = {"Linear Control", "Adaptive Boost"};
static const char *wm8753_base_filter[] =
{"130Hz @ 48kHz", "200Hz @ 48kHz", "100Hz @ 16kHz", "400Hz @ 48kHz",
"100Hz @ 8kHz", "200Hz @ 8kHz"};
static const char *wm8753_treble[] = {"8kHz", "4kHz"};
static const char *wm8753_alc_func[] = {"Off", "Right", "Left", "Stereo"};
static const char *wm8753_ng_type[] = {"Constant PGA Gain", "Mute ADC Output"};
static const char *wm8753_3d_func[] = {"Capture", "Playback"};
static const char *wm8753_3d_uc[] = {"2.2kHz", "1.5kHz"};
static const char *wm8753_3d_lc[] = {"200Hz", "500Hz"};
static const char *wm8753_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz"};
static const char *wm8753_mono_mix[] = {"Stereo", "Left", "Right", "Mono"};
static const char *wm8753_dac_phase[] = {"Non Inverted", "Inverted"};
static const char *wm8753_line_mix[] = {"Line 1 + 2", "Line 1 - 2",
"Line 1", "Line 2"};
static const char *wm8753_mono_mux[] = {"Line Mix", "Rx Mix"};
static const char *wm8753_right_mux[] = {"Line 2", "Rx Mix"};
static const char *wm8753_left_mux[] = {"Line 1", "Rx Mix"};
static const char *wm8753_rxmsel[] = {"RXP - RXN", "RXP + RXN", "RXP", "RXN"};
static const char *wm8753_sidetone_mux[] = {"Left PGA", "Mic 1", "Mic 2",
"Right PGA"};
static const char *wm8753_mono2_src[] = {"Inverted Mono 1", "Left", "Right",
"Left + Right"};
static const char *wm8753_out3[] = {"VREF", "ROUT2", "Left + Right"};
static const char *wm8753_out4[] = {"VREF", "Capture ST", "LOUT2"};
static const char *wm8753_radcsel[] = {"PGA", "Line or RXP-RXN", "Sidetone"};
static const char *wm8753_ladcsel[] = {"PGA", "Line or RXP-RXN", "Line"};
static const char *wm8753_mono_adc[] = {"Stereo", "Analogue Mix Left",
"Analogue Mix Right", "Digital Mono Mix"};
static const char *wm8753_adc_hp[] = {"3.4Hz @ 48kHz", "82Hz @ 16k",
"82Hz @ 8kHz", "170Hz @ 8kHz"};
static const char *wm8753_adc_filter[] = {"HiFi", "Voice"};
static const char *wm8753_mic_sel[] = {"Mic 1", "Mic 2", "Mic 3"};
static const char *wm8753_dai_mode[] = {"DAI 0", "DAI 1", "DAI 2", "DAI 3"};
static const char *wm8753_dat_sel[] = {"Stereo", "Left ADC", "Right ADC",
"Channel Swap"};
static const char *wm8753_rout2_phase[] = {"Non Inverted", "Inverted"};
static const struct soc_enum wm8753_enum[] = {
SOC_ENUM_SINGLE(WM8753_BASS, 7, 2, wm8753_base),
SOC_ENUM_SINGLE(WM8753_BASS, 4, 6, wm8753_base_filter),
SOC_ENUM_SINGLE(WM8753_TREBLE, 6, 2, wm8753_treble),
SOC_ENUM_SINGLE(WM8753_ALC1, 7, 4, wm8753_alc_func),
SOC_ENUM_SINGLE(WM8753_NGATE, 1, 2, wm8753_ng_type),
SOC_ENUM_SINGLE(WM8753_3D, 7, 2, wm8753_3d_func),
SOC_ENUM_SINGLE(WM8753_3D, 6, 2, wm8753_3d_uc),
SOC_ENUM_SINGLE(WM8753_3D, 5, 2, wm8753_3d_lc),
SOC_ENUM_SINGLE(WM8753_DAC, 1, 4, wm8753_deemp),
SOC_ENUM_SINGLE(WM8753_DAC, 4, 4, wm8753_mono_mix),
SOC_ENUM_SINGLE(WM8753_DAC, 6, 2, wm8753_dac_phase),
SOC_ENUM_SINGLE(WM8753_INCTL1, 3, 4, wm8753_line_mix),
SOC_ENUM_SINGLE(WM8753_INCTL1, 2, 2, wm8753_mono_mux),
SOC_ENUM_SINGLE(WM8753_INCTL1, 1, 2, wm8753_right_mux),
SOC_ENUM_SINGLE(WM8753_INCTL1, 0, 2, wm8753_left_mux),
SOC_ENUM_SINGLE(WM8753_INCTL2, 6, 4, wm8753_rxmsel),
SOC_ENUM_SINGLE(WM8753_INCTL2, 4, 4, wm8753_sidetone_mux),
SOC_ENUM_SINGLE(WM8753_OUTCTL, 7, 4, wm8753_mono2_src),
SOC_ENUM_SINGLE(WM8753_OUTCTL, 0, 3, wm8753_out3),
SOC_ENUM_SINGLE(WM8753_ADCTL2, 7, 3, wm8753_out4),
SOC_ENUM_SINGLE(WM8753_ADCIN, 2, 3, wm8753_radcsel),
SOC_ENUM_SINGLE(WM8753_ADCIN, 0, 3, wm8753_ladcsel),
SOC_ENUM_SINGLE(WM8753_ADCIN, 4, 4, wm8753_mono_adc),
SOC_ENUM_SINGLE(WM8753_ADC, 2, 4, wm8753_adc_hp),
SOC_ENUM_SINGLE(WM8753_ADC, 4, 2, wm8753_adc_filter),
SOC_ENUM_SINGLE(WM8753_MICBIAS, 6, 3, wm8753_mic_sel),
SOC_ENUM_SINGLE(WM8753_IOCTL, 2, 4, wm8753_dai_mode),
SOC_ENUM_SINGLE(WM8753_ADC, 7, 4, wm8753_dat_sel),
SOC_ENUM_SINGLE(WM8753_OUTCTL, 2, 2, wm8753_rout2_phase),
};
static int wm8753_get_dai(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
struct wm8753_priv *wm8753 = snd_soc_component_get_drvdata(component);
ucontrol->value.enumerated.item[0] = wm8753->dai_func;
return 0;
}
static int wm8753_set_dai(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol);
struct wm8753_priv *wm8753 = snd_soc_component_get_drvdata(component);
u16 ioctl;
if (wm8753->dai_func == ucontrol->value.enumerated.item[0])
return 0;
if (snd_soc_component_active(component))
return -EBUSY;
ioctl = snd_soc_component_read(component, WM8753_IOCTL);
wm8753->dai_func = ucontrol->value.enumerated.item[0];
if (((ioctl >> 2) & 0x3) == wm8753->dai_func)
return 1;
ioctl = (ioctl & 0x1f3) | (wm8753->dai_func << 2);
snd_soc_component_write(component, WM8753_IOCTL, ioctl);
wm8753_hifi_write_dai_fmt(component, wm8753->hifi_fmt);
wm8753_voice_write_dai_fmt(component, wm8753->voice_fmt);
return 1;
}
static const DECLARE_TLV_DB_SCALE(rec_mix_tlv, -1500, 300, 0);
static const DECLARE_TLV_DB_SCALE(mic_preamp_tlv, 1200, 600, 0);
static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
static const DECLARE_TLV_DB_RANGE(out_tlv,
/* 0000000 - 0101111 = "Analogue mute" */
0, 48, TLV_DB_SCALE_ITEM(-25500, 0, 0),
48, 127, TLV_DB_SCALE_ITEM(-7300, 100, 0)
);
static const DECLARE_TLV_DB_SCALE(mix_tlv, -1500, 300, 0);
static const DECLARE_TLV_DB_SCALE(voice_mix_tlv, -1200, 300, 0);
static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0);
static const struct snd_kcontrol_new wm8753_snd_controls[] = {
SOC_SINGLE("Hi-Fi DAC Left/Right channel Swap", WM8753_HIFI, 5, 1, 0),
SOC_DOUBLE_R_TLV("PCM Volume", WM8753_LDAC, WM8753_RDAC, 0, 255, 0, dac_tlv),
SOC_DOUBLE_R_TLV("ADC Capture Volume", WM8753_LADC, WM8753_RADC, 0, 255, 0,
adc_tlv),
SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8753_LOUT1V, WM8753_ROUT1V,
0, 127, 0, out_tlv),
SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8753_LOUT2V, WM8753_ROUT2V, 0,
127, 0, out_tlv),
SOC_SINGLE_TLV("Mono Playback Volume", WM8753_MOUTV, 0, 127, 0, out_tlv),
SOC_DOUBLE_R_TLV("Bypass Playback Volume", WM8753_LOUTM1, WM8753_ROUTM1, 4, 7,
1, mix_tlv),
SOC_DOUBLE_R_TLV("Sidetone Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 4,
7, 1, mix_tlv),
SOC_DOUBLE_R_TLV("Voice Playback Volume", WM8753_LOUTM2, WM8753_ROUTM2, 0, 7,
1, voice_mix_tlv),
SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8753_LOUT1V, WM8753_ROUT1V, 7,
1, 0),
SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8753_LOUT2V, WM8753_ROUT2V, 7,
1, 0),
SOC_SINGLE_TLV("Mono Bypass Playback Volume", WM8753_MOUTM1, 4, 7, 1, mix_tlv),
SOC_SINGLE_TLV("Mono Sidetone Playback Volume", WM8753_MOUTM2, 4, 7, 1,
mix_tlv),
SOC_SINGLE_TLV("Mono Voice Playback Volume", WM8753_MOUTM2, 0, 7, 1,
voice_mix_tlv),
SOC_SINGLE("Mono Playback ZC Switch", WM8753_MOUTV, 7, 1, 0),
SOC_ENUM("Bass Boost", wm8753_enum[0]),
SOC_ENUM("Bass Filter", wm8753_enum[1]),
SOC_SINGLE("Bass Volume", WM8753_BASS, 0, 15, 1),
SOC_SINGLE("Treble Volume", WM8753_TREBLE, 0, 15, 1),
SOC_ENUM("Treble Cut-off", wm8753_enum[2]),
SOC_DOUBLE_TLV("Sidetone Capture Volume", WM8753_RECMIX1, 0, 4, 7, 1,
rec_mix_tlv),
SOC_SINGLE_TLV("Voice Sidetone Capture Volume", WM8753_RECMIX2, 0, 7, 1,
rec_mix_tlv),
SOC_DOUBLE_R_TLV("Capture Volume", WM8753_LINVOL, WM8753_RINVOL, 0, 63, 0,
pga_tlv),
SOC_DOUBLE_R("Capture ZC Switch", WM8753_LINVOL, WM8753_RINVOL, 6, 1, 0),
SOC_DOUBLE_R("Capture Switch", WM8753_LINVOL, WM8753_RINVOL, 7, 1, 1),
SOC_ENUM("Capture Filter Select", wm8753_enum[23]),
SOC_ENUM("Capture Filter Cut-off", wm8753_enum[24]),
SOC_SINGLE("Capture Filter Switch", WM8753_ADC, 0, 1, 1),
SOC_SINGLE("ALC Capture Target Volume", WM8753_ALC1, 0, 7, 0),
SOC_SINGLE("ALC Capture Max Volume", WM8753_ALC1, 4, 7, 0),
SOC_ENUM("ALC Capture Function", wm8753_enum[3]),
SOC_SINGLE("ALC Capture ZC Switch", WM8753_ALC2, 8, 1, 0),
SOC_SINGLE("ALC Capture Hold Time", WM8753_ALC2, 0, 15, 1),
SOC_SINGLE("ALC Capture Decay Time", WM8753_ALC3, 4, 15, 1),
SOC_SINGLE("ALC Capture Attack Time", WM8753_ALC3, 0, 15, 0),
SOC_SINGLE("ALC Capture NG Threshold", WM8753_NGATE, 3, 31, 0),
SOC_ENUM("ALC Capture NG Type", wm8753_enum[4]),
SOC_SINGLE("ALC Capture NG Switch", WM8753_NGATE, 0, 1, 0),
SOC_ENUM("3D Function", wm8753_enum[5]),
SOC_ENUM("3D Upper Cut-off", wm8753_enum[6]),
SOC_ENUM("3D Lower Cut-off", wm8753_enum[7]),
SOC_SINGLE("3D Volume", WM8753_3D, 1, 15, 0),
SOC_SINGLE("3D Switch", WM8753_3D, 0, 1, 0),
SOC_SINGLE("Capture 6dB Attenuate", WM8753_ADCTL1, 2, 1, 0),
SOC_SINGLE("Playback 6dB Attenuate", WM8753_ADCTL1, 1, 1, 0),
SOC_ENUM("De-emphasis", wm8753_enum[8]),
SOC_ENUM("Playback Mono Mix", wm8753_enum[9]),
SOC_ENUM("Playback Phase", wm8753_enum[10]),
SOC_SINGLE_TLV("Mic2 Capture Volume", WM8753_INCTL1, 7, 3, 0, mic_preamp_tlv),
SOC_SINGLE_TLV("Mic1 Capture Volume", WM8753_INCTL1, 5, 3, 0, mic_preamp_tlv),
SOC_ENUM_EXT("DAI Mode", wm8753_enum[26], wm8753_get_dai, wm8753_set_dai),
SOC_ENUM("ADC Data Select", wm8753_enum[27]),
SOC_ENUM("ROUT2 Phase", wm8753_enum[28]),
};
/*
* _DAPM_ Controls
*/
/* Left Mixer */
static const struct snd_kcontrol_new wm8753_left_mixer_controls[] = {
SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_LOUTM2, 8, 1, 0),
SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_LOUTM2, 7, 1, 0),
SOC_DAPM_SINGLE("Left Playback Switch", WM8753_LOUTM1, 8, 1, 0),
SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_LOUTM1, 7, 1, 0),
};
/* Right mixer */
static const struct snd_kcontrol_new wm8753_right_mixer_controls[] = {
SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_ROUTM2, 8, 1, 0),
SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_ROUTM2, 7, 1, 0),
SOC_DAPM_SINGLE("Right Playback Switch", WM8753_ROUTM1, 8, 1, 0),
SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_ROUTM1, 7, 1, 0),
};
/* Mono mixer */
static const struct snd_kcontrol_new wm8753_mono_mixer_controls[] = {
SOC_DAPM_SINGLE("Left Playback Switch", WM8753_MOUTM1, 8, 1, 0),
SOC_DAPM_SINGLE("Right Playback Switch", WM8753_MOUTM2, 8, 1, 0),
SOC_DAPM_SINGLE("Voice Playback Switch", WM8753_MOUTM2, 3, 1, 0),
SOC_DAPM_SINGLE("Sidetone Playback Switch", WM8753_MOUTM2, 7, 1, 0),
SOC_DAPM_SINGLE("Bypass Playback Switch", WM8753_MOUTM1, 7, 1, 0),
};
/* Mono 2 Mux */
static const struct snd_kcontrol_new wm8753_mono2_controls =
SOC_DAPM_ENUM("Route", wm8753_enum[17]);
/* Out 3 Mux */
static const struct snd_kcontrol_new wm8753_out3_controls =
SOC_DAPM_ENUM("Route", wm8753_enum[18]);
/* Out 4 Mux */
static const struct snd_kcontrol_new wm8753_out4_controls =
SOC_DAPM_ENUM("Route", wm8753_enum[19]);
/* ADC Mono Mix */
static const struct snd_kcontrol_new wm8753_adc_mono_controls =
SOC_DAPM_ENUM("Route", wm8753_enum[22]);
/* Record mixer */
static const struct snd_kcontrol_new wm8753_record_mixer_controls[] = {
SOC_DAPM_SINGLE("Voice Capture Switch", WM8753_RECMIX2, 3, 1, 0),
SOC_DAPM_SINGLE("Left Capture Switch", WM8753_RECMIX1, 3, 1, 0),
SOC_DAPM_SINGLE("Right Capture Switch", WM8753_RECMIX1, 7, 1, 0),
};
/* Left ADC mux */
static const struct snd_kcontrol_new wm8753_adc_left_controls =
SOC_DAPM_ENUM("Route", wm8753_enum[21]);
/* Right ADC mux */
static const struct snd_kcontrol_new wm8753_adc_right_controls =
SOC_DAPM_ENUM("Route", wm8753_enum[20]);
/* MIC mux */
static const struct snd_kcontrol_new wm8753_mic_mux_controls =
SOC_DAPM_ENUM("Route", wm8753_enum[16]);
/* ALC mixer */
static const struct snd_kcontrol_new wm8753_alc_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Capture Switch", WM8753_INCTL2, 3, 1, 0),
SOC_DAPM_SINGLE("Mic2 Capture Switch", WM8753_INCTL2, 2, 1, 0),
SOC_DAPM_SINGLE("Mic1 Capture Switch", WM8753_INCTL2, 1, 1, 0),
SOC_DAPM_SINGLE("Rx Capture Switch", WM8753_INCTL2, 0, 1, 0),
};
/* Left Line mux */
static const struct snd_kcontrol_new wm8753_line_left_controls =
SOC_DAPM_ENUM("Route", wm8753_enum[14]);
/* Right Line mux */
static const struct snd_kcontrol_new wm8753_line_right_controls =
SOC_DAPM_ENUM("Route", wm8753_enum[13]);
/* Mono Line mux */
static const struct snd_kcontrol_new wm8753_line_mono_controls =
SOC_DAPM_ENUM("Route", wm8753_enum[12]);
/* Line mux and mixer */
static const struct snd_kcontrol_new wm8753_line_mux_mix_controls =
SOC_DAPM_ENUM("Route", wm8753_enum[11]);
/* Rx mux and mixer */
static const struct snd_kcontrol_new wm8753_rx_mux_mix_controls =
SOC_DAPM_ENUM("Route", wm8753_enum[15]);
/* Mic Selector Mux */
static const struct snd_kcontrol_new wm8753_mic_sel_mux_controls =
SOC_DAPM_ENUM("Route", wm8753_enum[25]);
static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
SND_SOC_DAPM_MICBIAS("Mic Bias", WM8753_PWR1, 5, 0),
SND_SOC_DAPM_MIXER("Left Mixer", WM8753_PWR4, 0, 0,
&wm8753_left_mixer_controls[0], ARRAY_SIZE(wm8753_left_mixer_controls)),
SND_SOC_DAPM_PGA("Left Out 1", WM8753_PWR3, 8, 0, NULL, 0),
SND_SOC_DAPM_PGA("Left Out 2", WM8753_PWR3, 6, 0, NULL, 0),
SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", WM8753_PWR1, 3, 0),
SND_SOC_DAPM_OUTPUT("LOUT1"),
SND_SOC_DAPM_OUTPUT("LOUT2"),
SND_SOC_DAPM_MIXER("Right Mixer", WM8753_PWR4, 1, 0,
&wm8753_right_mixer_controls[0], ARRAY_SIZE(wm8753_right_mixer_controls)),
SND_SOC_DAPM_PGA("Right Out 1", WM8753_PWR3, 7, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Out 2", WM8753_PWR3, 5, 0, NULL, 0),
SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", WM8753_PWR1, 2, 0),
SND_SOC_DAPM_OUTPUT("ROUT1"),
SND_SOC_DAPM_OUTPUT("ROUT2"),
SND_SOC_DAPM_MIXER("Mono Mixer", WM8753_PWR4, 2, 0,
&wm8753_mono_mixer_controls[0], ARRAY_SIZE(wm8753_mono_mixer_controls)),
SND_SOC_DAPM_PGA("Mono Out 1", WM8753_PWR3, 2, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mono Out 2", WM8753_PWR3, 1, 0, NULL, 0),
SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", WM8753_PWR1, 4, 0),
SND_SOC_DAPM_OUTPUT("MONO1"),
SND_SOC_DAPM_MUX("Mono 2 Mux", SND_SOC_NOPM, 0, 0, &wm8753_mono2_controls),
SND_SOC_DAPM_OUTPUT("MONO2"),
SND_SOC_DAPM_MIXER("Out3 Left + Right", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out3_controls),
SND_SOC_DAPM_PGA("Out 3", WM8753_PWR3, 4, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("OUT3"),
SND_SOC_DAPM_MUX("Out4 Mux", SND_SOC_NOPM, 0, 0, &wm8753_out4_controls),
SND_SOC_DAPM_PGA("Out 4", WM8753_PWR3, 3, 0, NULL, 0),
SND_SOC_DAPM_OUTPUT("OUT4"),
SND_SOC_DAPM_MIXER("Playback Mixer", WM8753_PWR4, 3, 0,
&wm8753_record_mixer_controls[0],
ARRAY_SIZE(wm8753_record_mixer_controls)),
SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8753_PWR2, 3, 0),
SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8753_PWR2, 2, 0),
SND_SOC_DAPM_MUX("Capture Left Mixer", SND_SOC_NOPM, 0, 0,
&wm8753_adc_mono_controls),
SND_SOC_DAPM_MUX("Capture Right Mixer", SND_SOC_NOPM, 0, 0,
&wm8753_adc_mono_controls),
SND_SOC_DAPM_MUX("Capture Left Mux", SND_SOC_NOPM, 0, 0,
&wm8753_adc_left_controls),
SND_SOC_DAPM_MUX("Capture Right Mux", SND_SOC_NOPM, 0, 0,
&wm8753_adc_right_controls),
SND_SOC_DAPM_MUX("Mic Sidetone Mux", SND_SOC_NOPM, 0, 0,
&wm8753_mic_mux_controls),
SND_SOC_DAPM_PGA("Left Capture Volume", WM8753_PWR2, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Capture Volume", WM8753_PWR2, 4, 0, NULL, 0),
SND_SOC_DAPM_MIXER("ALC Mixer", WM8753_PWR2, 6, 0,
&wm8753_alc_mixer_controls[0], ARRAY_SIZE(wm8753_alc_mixer_controls)),
SND_SOC_DAPM_MUX("Line Left Mux", SND_SOC_NOPM, 0, 0,
&wm8753_line_left_controls),
SND_SOC_DAPM_MUX("Line Right Mux", SND_SOC_NOPM, 0, 0,
&wm8753_line_right_controls),
SND_SOC_DAPM_MUX("Line Mono Mux", SND_SOC_NOPM, 0, 0,
&wm8753_line_mono_controls),
SND_SOC_DAPM_MUX("Line Mixer", WM8753_PWR2, 0, 0,
&wm8753_line_mux_mix_controls),
SND_SOC_DAPM_MUX("Rx Mixer", WM8753_PWR2, 1, 0,
&wm8753_rx_mux_mix_controls),
SND_SOC_DAPM_PGA("Mic 1 Volume", WM8753_PWR2, 8, 0, NULL, 0),
SND_SOC_DAPM_PGA("Mic 2 Volume", WM8753_PWR2, 7, 0, NULL, 0),
SND_SOC_DAPM_MUX("Mic Selection Mux", SND_SOC_NOPM, 0, 0,
&wm8753_mic_sel_mux_controls),
SND_SOC_DAPM_INPUT("LINE1"),
SND_SOC_DAPM_INPUT("LINE2"),
SND_SOC_DAPM_INPUT("RXP"),
SND_SOC_DAPM_INPUT("RXN"),
SND_SOC_DAPM_INPUT("ACIN"),
SND_SOC_DAPM_OUTPUT("ACOP"),
SND_SOC_DAPM_INPUT("MIC1N"),
SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2N"),
SND_SOC_DAPM_INPUT("MIC2"),
SND_SOC_DAPM_VMID("VREF"),
};
static const struct snd_soc_dapm_route wm8753_dapm_routes[] = {
/* left mixer */
{"Left Mixer", "Left Playback Switch", "Left DAC"},
{"Left Mixer", "Voice Playback Switch", "Voice DAC"},
{"Left Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
{"Left Mixer", "Bypass Playback Switch", "Line Left Mux"},
/* right mixer */
{"Right Mixer", "Right Playback Switch", "Right DAC"},
{"Right Mixer", "Voice Playback Switch", "Voice DAC"},
{"Right Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
{"Right Mixer", "Bypass Playback Switch", "Line Right Mux"},
/* mono mixer */
{"Mono Mixer", "Voice Playback Switch", "Voice DAC"},
{"Mono Mixer", "Left Playback Switch", "Left DAC"},
{"Mono Mixer", "Right Playback Switch", "Right DAC"},
{"Mono Mixer", "Sidetone Playback Switch", "Mic Sidetone Mux"},
{"Mono Mixer", "Bypass Playback Switch", "Line Mono Mux"},
/* left out */
{"Left Out 1", NULL, "Left Mixer"},
{"Left Out 2", NULL, "Left Mixer"},
{"LOUT1", NULL, "Left Out 1"},
{"LOUT2", NULL, "Left Out 2"},
/* right out */
{"Right Out 1", NULL, "Right Mixer"},
{"Right Out 2", NULL, "Right Mixer"},
{"ROUT1", NULL, "Right Out 1"},
{"ROUT2", NULL, "Right Out 2"},
/* mono 1 out */
{"Mono Out 1", NULL, "Mono Mixer"},
{"MONO1", NULL, "Mono Out 1"},
/* mono 2 out */
{"Mono 2 Mux", "Left + Right", "Out3 Left + Right"},
{"Mono 2 Mux", "Inverted Mono 1", "MONO1"},
{"Mono 2 Mux", "Left", "Left Mixer"},
{"Mono 2 Mux", "Right", "Right Mixer"},
{"Mono Out 2", NULL, "Mono 2 Mux"},
{"MONO2", NULL, "Mono Out 2"},
/* out 3 */
{"Out3 Left + Right", NULL, "Left Mixer"},
{"Out3 Left + Right", NULL, "Right Mixer"},
{"Out3 Mux", "VREF", "VREF"},
{"Out3 Mux", "Left + Right", "Out3 Left + Right"},
{"Out3 Mux", "ROUT2", "ROUT2"},
{"Out 3", NULL, "Out3 Mux"},
{"OUT3", NULL, "Out 3"},
/* out 4 */
{"Out4 Mux", "VREF", "VREF"},
{"Out4 Mux", "Capture ST", "Playback Mixer"},
{"Out4 Mux", "LOUT2", "LOUT2"},
{"Out 4", NULL, "Out4 Mux"},
{"OUT4", NULL, "Out 4"},
/* record mixer */
{"Playback Mixer", "Left Capture Switch", "Left Mixer"},
{"Playback Mixer", "Voice Capture Switch", "Mono Mixer"},
{"Playback Mixer", "Right Capture Switch", "Right Mixer"},
/* Mic/SideTone Mux */
{"Mic Sidetone Mux", "Left PGA", "Left Capture Volume"},
{"Mic Sidetone Mux", "Right PGA", "Right Capture Volume"},
{"Mic Sidetone Mux", "Mic 1", "Mic 1 Volume"},
{"Mic Sidetone Mux", "Mic 2", "Mic 2 Volume"},
/* Capture Left Mux */
{"Capture Left Mux", "PGA", "Left Capture Volume"},
{"Capture Left Mux", "Line or RXP-RXN", "Line Left Mux"},
{"Capture Left Mux", "Line", "LINE1"},
/* Capture Right Mux */
{"Capture Right Mux", "PGA", "Right Capture Volume"},
{"Capture Right Mux", "Line or RXP-RXN", "Line Right Mux"},
{"Capture Right Mux", "Sidetone", "Playback Mixer"},
/* Mono Capture mixer-mux */
{"Capture Right Mixer", "Stereo", "Capture Right Mux"},
{"Capture Left Mixer", "Stereo", "Capture Left Mux"},
{"Capture Left Mixer", "Analogue Mix Left", "Capture Left Mux"},
{"Capture Left Mixer", "Analogue Mix Left", "Capture Right Mux"},
{"Capture Right Mixer", "Analogue Mix Right", "Capture Left Mux"},
{"Capture Right Mixer", "Analogue Mix Right", "Capture Right Mux"},
{"Capture Left Mixer", "Digital Mono Mix", "Capture Left Mux"},
{"Capture Left Mixer", "Digital Mono Mix", "Capture Right Mux"},
{"Capture Right Mixer", "Digital Mono Mix", "Capture Left Mux"},
{"Capture Right Mixer", "Digital Mono Mix", "Capture Right Mux"},
/* ADC */
{"Left ADC", NULL, "Capture Left Mixer"},
{"Right ADC", NULL, "Capture Right Mixer"},
/* Left Capture Volume */
{"Left Capture Volume", NULL, "ACIN"},
/* Right Capture Volume */
{"Right Capture Volume", NULL, "Mic 2 Volume"},
/* ALC Mixer */
{"ALC Mixer", "Line Capture Switch", "Line Mixer"},
{"ALC Mixer", "Mic2 Capture Switch", "Mic 2 Volume"},
{"ALC Mixer", "Mic1 Capture Switch", "Mic 1 Volume"},
{"ALC Mixer", "Rx Capture Switch", "Rx Mixer"},
/* Line Left Mux */
{"Line Left Mux", "Line 1", "LINE1"},
{"Line Left Mux", "Rx Mix", "Rx Mixer"},
/* Line Right Mux */
{"Line Right Mux", "Line 2", "LINE2"},
{"Line Right Mux", "Rx Mix", "Rx Mixer"},
/* Line Mono Mux */
{"Line Mono Mux", "Line Mix", "Line Mixer"},
{"Line Mono Mux", "Rx Mix", "Rx Mixer"},
/* Line Mixer/Mux */
{"Line Mixer", "Line 1 + 2", "LINE1"},
{"Line Mixer", "Line 1 - 2", "LINE1"},
{"Line Mixer", "Line 1 + 2", "LINE2"},
{"Line Mixer", "Line 1 - 2", "LINE2"},
{"Line Mixer", "Line 1", "LINE1"},
{"Line Mixer", "Line 2", "LINE2"},
/* Rx Mixer/Mux */
{"Rx Mixer", "RXP - RXN", "RXP"},
{"Rx Mixer", "RXP + RXN", "RXP"},
{"Rx Mixer", "RXP - RXN", "RXN"},
{"Rx Mixer", "RXP + RXN", "RXN"},
{"Rx Mixer", "RXP", "RXP"},
{"Rx Mixer", "RXN", "RXN"},
/* Mic 1 Volume */
{"Mic 1 Volume", NULL, "MIC1N"},
{"Mic 1 Volume", NULL, "Mic Selection Mux"},
/* Mic 2 Volume */
{"Mic 2 Volume", NULL, "MIC2N"},
{"Mic 2 Volume", NULL, "MIC2"},
/* Mic Selector Mux */
{"Mic Selection Mux", "Mic 1", "MIC1"},
{"Mic Selection Mux", "Mic 2", "MIC2N"},
{"Mic Selection Mux", "Mic 3", "MIC2"},
/* ACOP */
{"ACOP", NULL, "ALC Mixer"},
};
/* PLL divisors */
struct _pll_div {
u32 div2:1;
u32 n:4;
u32 k:24;
};
/* The size in bits of the pll divide multiplied by 10
* to allow rounding later */
#define FIXED_PLL_SIZE ((1 << 22) * 10)
static void pll_factors(struct _pll_div *pll_div, unsigned int target,
unsigned int source)
{
u64 Kpart;
unsigned int K, Ndiv, Nmod;
Ndiv = target / source;
if (Ndiv < 6) {
source >>= 1;
pll_div->div2 = 1;
Ndiv = target / source;
} else
pll_div->div2 = 0;
if ((Ndiv < 6) || (Ndiv > 12))
printk(KERN_WARNING
"wm8753: unsupported N = %u\n", Ndiv);
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;
}
static int wm8753_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
int source, unsigned int freq_in, unsigned int freq_out)
{
u16 reg, enable;
int offset;
struct snd_soc_component *component = codec_dai->component;
if (pll_id < WM8753_PLL1 || pll_id > WM8753_PLL2)
return -ENODEV;
if (pll_id == WM8753_PLL1) {
offset = 0;
enable = 0x10;
reg = snd_soc_component_read(component, WM8753_CLOCK) & 0xffef;
} else {
offset = 4;
enable = 0x8;
reg = snd_soc_component_read(component, WM8753_CLOCK) & 0xfff7;
}
if (!freq_in || !freq_out) {
/* disable PLL */
snd_soc_component_write(component, WM8753_PLL1CTL1 + offset, 0x0026);
snd_soc_component_write(component, WM8753_CLOCK, reg);
return 0;
} else {
u16 value = 0;
struct _pll_div pll_div;
pll_factors(&pll_div, freq_out * 8, freq_in);
/* set up N and K PLL divisor ratios */
/* bits 8:5 = PLL_N, bits 3:0 = PLL_K[21:18] */
value = (pll_div.n << 5) + ((pll_div.k & 0x3c0000) >> 18);
snd_soc_component_write(component, WM8753_PLL1CTL2 + offset, value);
/* bits 8:0 = PLL_K[17:9] */
value = (pll_div.k & 0x03fe00) >> 9;
snd_soc_component_write(component, WM8753_PLL1CTL3 + offset, value);
/* bits 8:0 = PLL_K[8:0] */
value = pll_div.k & 0x0001ff;
snd_soc_component_write(component, WM8753_PLL1CTL4 + offset, value);
/* set PLL as input and enable */
snd_soc_component_write(component, WM8753_PLL1CTL1 + offset, 0x0027 |
(pll_div.div2 << 3));
snd_soc_component_write(component, WM8753_CLOCK, reg | enable);
}
return 0;
}
struct _coeff_div {
u32 mclk;
u32 rate;
u8 sr:5;
u8 usb:1;
};
/* codec hifi mclk (after PLL) clock divider coefficients */
static const struct _coeff_div coeff_div[] = {
/* 8k */
{12288000, 8000, 0x6, 0x0},
{11289600, 8000, 0x16, 0x0},
{18432000, 8000, 0x7, 0x0},
{16934400, 8000, 0x17, 0x0},
{12000000, 8000, 0x6, 0x1},
/* 11.025k */
{11289600, 11025, 0x18, 0x0},
{16934400, 11025, 0x19, 0x0},
{12000000, 11025, 0x19, 0x1},
/* 16k */
{12288000, 16000, 0xa, 0x0},
{18432000, 16000, 0xb, 0x0},
{12000000, 16000, 0xa, 0x1},
/* 22.05k */
{11289600, 22050, 0x1a, 0x0},
{16934400, 22050, 0x1b, 0x0},
{12000000, 22050, 0x1b, 0x1},
/* 32k */
{12288000, 32000, 0xc, 0x0},
{18432000, 32000, 0xd, 0x0},
{12000000, 32000, 0xa, 0x1},
/* 44.1k */
{11289600, 44100, 0x10, 0x0},
{16934400, 44100, 0x11, 0x0},
{12000000, 44100, 0x11, 0x1},
/* 48k */
{12288000, 48000, 0x0, 0x0},
{18432000, 48000, 0x1, 0x0},
{12000000, 48000, 0x0, 0x1},
/* 88.2k */
{11289600, 88200, 0x1e, 0x0},
{16934400, 88200, 0x1f, 0x0},
{12000000, 88200, 0x1f, 0x1},
/* 96k */
{12288000, 96000, 0xe, 0x0},
{18432000, 96000, 0xf, 0x0},
{12000000, 96000, 0xe, 0x1},
};
static int get_coeff(int mclk, int rate)
{
int i;
for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk)
return i;
}
return -EINVAL;
}
/*
* Clock after PLL and dividers
*/
static int wm8753_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 wm8753_priv *wm8753 = snd_soc_component_get_drvdata(component);
switch (freq) {
case 11289600:
case 12000000:
case 12288000:
case 16934400:
case 18432000:
if (clk_id == WM8753_MCLK) {
wm8753->sysclk = freq;
return 0;
} else if (clk_id == WM8753_PCMCLK) {
wm8753->pcmclk = freq;
return 0;
}
break;
}
return -EINVAL;
}
/*
* Set's ADC and Voice DAC format.
*/
static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_component *component,
unsigned int fmt)
{
u16 voice = snd_soc_component_read(component, WM8753_PCM) & 0x01ec;
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
voice |= 0x0002;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
voice |= 0x0001;
break;
case SND_SOC_DAIFMT_DSP_A:
voice |= 0x0003;
break;
case SND_SOC_DAIFMT_DSP_B:
voice |= 0x0013;
break;
default:
return -EINVAL;
}
snd_soc_component_write(component, WM8753_PCM, voice);
return 0;
}
/*
* Set PCM DAI bit size and sample rate.
*/
static int wm8753_pcm_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 wm8753_priv *wm8753 = snd_soc_component_get_drvdata(component);
u16 voice = snd_soc_component_read(component, WM8753_PCM) & 0x01f3;
u16 srate = snd_soc_component_read(component, WM8753_SRATE1) & 0x017f;
/* bit size */
switch (params_width(params)) {
case 16:
break;
case 20:
voice |= 0x0004;
break;
case 24:
voice |= 0x0008;
break;
case 32:
voice |= 0x000c;
break;
}
/* sample rate */
if (params_rate(params) * 384 == wm8753->pcmclk)
srate |= 0x80;
snd_soc_component_write(component, WM8753_SRATE1, srate);
snd_soc_component_write(component, WM8753_PCM, voice);
return 0;
}
/*
* Set's PCM dai fmt and BCLK.
*/
static int wm8753_pcm_set_dai_fmt(struct snd_soc_component *component,
unsigned int fmt)
{
u16 voice, ioctl;
voice = snd_soc_component_read(component, WM8753_PCM) & 0x011f;
ioctl = snd_soc_component_read(component, WM8753_IOCTL) & 0x015d;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
break;
case SND_SOC_DAIFMT_CBM_CFM:
ioctl |= 0x2;
fallthrough;
case SND_SOC_DAIFMT_CBM_CFS:
voice |= 0x0040;
break;
default:
return -EINVAL;
}
/* clock inversion */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_A:
case SND_SOC_DAIFMT_DSP_B:
/* frame inversion not valid for DSP modes */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_IB_NF:
voice |= 0x0080;
break;
default:
return -EINVAL;
}
break;
case SND_SOC_DAIFMT_I2S:
case SND_SOC_DAIFMT_RIGHT_J:
case SND_SOC_DAIFMT_LEFT_J:
voice &= ~0x0010;
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_IB_IF:
voice |= 0x0090;
break;
case SND_SOC_DAIFMT_IB_NF:
voice |= 0x0080;
break;
case SND_SOC_DAIFMT_NB_IF:
voice |= 0x0010;
break;
default:
return -EINVAL;
}
break;
default:
return -EINVAL;
}
snd_soc_component_write(component, WM8753_PCM, voice);
snd_soc_component_write(component, WM8753_IOCTL, ioctl);
return 0;
}
static int wm8753_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 WM8753_PCMDIV:
reg = snd_soc_component_read(component, WM8753_CLOCK) & 0x003f;
snd_soc_component_write(component, WM8753_CLOCK, reg | div);
break;
case WM8753_BCLKDIV:
reg = snd_soc_component_read(component, WM8753_SRATE2) & 0x01c7;
snd_soc_component_write(component, WM8753_SRATE2, reg | div);
break;
case WM8753_VXCLKDIV:
reg = snd_soc_component_read(component, WM8753_SRATE2) & 0x003f;
snd_soc_component_write(component, WM8753_SRATE2, reg | div);
break;
default:
return -EINVAL;
}
return 0;
}
/*
* Set's HiFi DAC format.
*/
static int wm8753_hdac_set_dai_fmt(struct snd_soc_component *component,
unsigned int fmt)
{
u16 hifi = snd_soc_component_read(component, WM8753_HIFI) & 0x01e0;
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
hifi |= 0x0002;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
hifi |= 0x0001;
break;
case SND_SOC_DAIFMT_DSP_A:
hifi |= 0x0003;
break;
case SND_SOC_DAIFMT_DSP_B:
hifi |= 0x0013;
break;
default:
return -EINVAL;
}
snd_soc_component_write(component, WM8753_HIFI, hifi);
return 0;
}
/*
* Set's I2S DAI format.
*/
static int wm8753_i2s_set_dai_fmt(struct snd_soc_component *component,
unsigned int fmt)
{
u16 ioctl, hifi;
hifi = snd_soc_component_read(component, WM8753_HIFI) & 0x013f;
ioctl = snd_soc_component_read(component, WM8753_IOCTL) & 0x00ae;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBS_CFS:
break;
case SND_SOC_DAIFMT_CBM_CFM:
ioctl |= 0x1;
fallthrough;
case SND_SOC_DAIFMT_CBM_CFS:
hifi |= 0x0040;
break;
default:
return -EINVAL;
}
/* clock inversion */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_DSP_A:
case SND_SOC_DAIFMT_DSP_B:
/* frame inversion not valid for DSP modes */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_IB_NF:
hifi |= 0x0080;
break;
default:
return -EINVAL;
}
break;
case SND_SOC_DAIFMT_I2S:
case SND_SOC_DAIFMT_RIGHT_J:
case SND_SOC_DAIFMT_LEFT_J:
hifi &= ~0x0010;
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_IB_IF:
hifi |= 0x0090;
break;
case SND_SOC_DAIFMT_IB_NF:
hifi |= 0x0080;
break;
case SND_SOC_DAIFMT_NB_IF:
hifi |= 0x0010;
break;
default:
return -EINVAL;
}
break;
default:
return -EINVAL;
}
snd_soc_component_write(component, WM8753_HIFI, hifi);
snd_soc_component_write(component, WM8753_IOCTL, ioctl);
return 0;
}
/*
* Set PCM DAI bit size and sample rate.
*/
static int wm8753_i2s_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 wm8753_priv *wm8753 = snd_soc_component_get_drvdata(component);
u16 srate = snd_soc_component_read(component, WM8753_SRATE1) & 0x01c0;
u16 hifi = snd_soc_component_read(component, WM8753_HIFI) & 0x01f3;
int coeff;
/* is digital filter coefficient valid ? */
coeff = get_coeff(wm8753->sysclk, params_rate(params));
if (coeff < 0) {
printk(KERN_ERR "wm8753 invalid MCLK or rate\n");
return coeff;
}
snd_soc_component_write(component, WM8753_SRATE1, srate | (coeff_div[coeff].sr << 1) |
coeff_div[coeff].usb);
/* bit size */
switch (params_width(params)) {
case 16:
break;
case 20:
hifi |= 0x0004;
break;
case 24:
hifi |= 0x0008;
break;
case 32:
hifi |= 0x000c;
break;
}
snd_soc_component_write(component, WM8753_HIFI, hifi);
return 0;
}
static int wm8753_mode1v_set_dai_fmt(struct snd_soc_component *component,
unsigned int fmt)
{
u16 clock;
/* set clk source as pcmclk */
clock = snd_soc_component_read(component, WM8753_CLOCK) & 0xfffb;
snd_soc_component_write(component, WM8753_CLOCK, clock);
return wm8753_vdac_adc_set_dai_fmt(component, fmt);
}
static int wm8753_mode1h_set_dai_fmt(struct snd_soc_component *component,
unsigned int fmt)
{
return wm8753_hdac_set_dai_fmt(component, fmt);
}
static int wm8753_mode2_set_dai_fmt(struct snd_soc_component *component,
unsigned int fmt)
{
u16 clock;
/* set clk source as pcmclk */
clock = snd_soc_component_read(component, WM8753_CLOCK) & 0xfffb;
snd_soc_component_write(component, WM8753_CLOCK, clock);
return wm8753_vdac_adc_set_dai_fmt(component, fmt);
}
static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_component *component,
unsigned int fmt)
{
u16 clock;
/* set clk source as mclk */
clock = snd_soc_component_read(component, WM8753_CLOCK) & 0xfffb;
snd_soc_component_write(component, WM8753_CLOCK, clock | 0x4);
if (wm8753_hdac_set_dai_fmt(component, fmt) < 0)
return -EINVAL;
return wm8753_vdac_adc_set_dai_fmt(component, fmt);
}
static int wm8753_hifi_write_dai_fmt(struct snd_soc_component *component,
unsigned int fmt)
{
struct wm8753_priv *wm8753 = snd_soc_component_get_drvdata(component);
int ret = 0;
switch (wm8753->dai_func) {
case 0:
ret = wm8753_mode1h_set_dai_fmt(component, fmt);
break;
case 1:
ret = wm8753_mode2_set_dai_fmt(component, fmt);
break;
case 2:
case 3:
ret = wm8753_mode3_4_set_dai_fmt(component, fmt);
break;
default:
break;
}
if (ret)
return ret;
return wm8753_i2s_set_dai_fmt(component, fmt);
}
static int wm8753_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_component *component = codec_dai->component;
struct wm8753_priv *wm8753 = snd_soc_component_get_drvdata(component);
wm8753->hifi_fmt = fmt;
return wm8753_hifi_write_dai_fmt(component, fmt);
};
static int wm8753_voice_write_dai_fmt(struct snd_soc_component *component,
unsigned int fmt)
{
struct wm8753_priv *wm8753 = snd_soc_component_get_drvdata(component);
int ret = 0;
if (wm8753->dai_func != 0)
return 0;
ret = wm8753_mode1v_set_dai_fmt(component, fmt);
if (ret)
return ret;
ret = wm8753_pcm_set_dai_fmt(component, fmt);
if (ret)
return ret;
return 0;
};
static int wm8753_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_component *component = codec_dai->component;
struct wm8753_priv *wm8753 = snd_soc_component_get_drvdata(component);
wm8753->voice_fmt = fmt;
return wm8753_voice_write_dai_fmt(component, fmt);
};
static int wm8753_mute(struct snd_soc_dai *dai, int mute, int direction)
{
struct snd_soc_component *component = dai->component;
u16 mute_reg = snd_soc_component_read(component, WM8753_DAC) & 0xfff7;
struct wm8753_priv *wm8753 = snd_soc_component_get_drvdata(component);
/* the digital mute covers the HiFi and Voice DAC's on the WM8753.
* make sure we check if they are not both active when we mute */
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
if (mute && wm8753->dai_func == 1) {
if (!snd_soc_component_active(component))
snd_soc_component_write(component, WM8753_DAC, mute_reg | 0x8);
} else {
if (mute)
snd_soc_component_write(component, WM8753_DAC, mute_reg | 0x8);
else
snd_soc_component_write(component, WM8753_DAC, mute_reg);
}
return 0;
}
ASoC: wm8753: Integrate capacitor charging into the DAPM sequence When being powered on, either initially on probe or when resuming from suspend, the wm8971 configures the device for quick output capacitor charging. Since the charging can take a rather long time (up to multiple seconds) it is done asynchronously without blocking. A delayed work item is run once the charging is finished and the device is switched to the target bias level. This all done asynchronously to the regular DAPM sequence accessing the same data structures and registers without any looking, which can lead to race conditions. Furthermore this potentially delays the start of stream on the CODEC while the rest of the system is already up and running, meaning the first bytes of audio are lost. It also does no comply with the assumption of the DAPM core that if set_bias_level() returned successfully the device will be at the requested bias level. This patch slightly refactors things and makes sure that the caps charging is properly integrated into the DAPM sequence. When transitioning from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY the part will be put into fast charging mode and a work item will be scheduled that puts it back into standby charging once the charging period has elapsed. If a playback or capture stream is started while charging is in progress the driver will now wait in SND_SOC_BIAS_PREPARE until the charging is done. This makes sure that charging is done asynchronously in the background when the chip is idle, but at the same time makes sure that playback/capture is not started before the charging is done. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-03-30 22:04:48 +03:00
static void wm8753_charge_work(struct work_struct *work)
{
struct wm8753_priv *wm8753 =
container_of(work, struct wm8753_priv, charge_work.work);
/* Set to 500k */
regmap_update_bits(wm8753->regmap, WM8753_PWR1, 0x0180, 0x0100);
}
static int wm8753_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
struct wm8753_priv *wm8753 = snd_soc_component_get_drvdata(component);
u16 pwr_reg = snd_soc_component_read(component, WM8753_PWR1) & 0xfe3e;
switch (level) {
case SND_SOC_BIAS_ON:
/* set vmid to 50k and unmute dac */
snd_soc_component_write(component, WM8753_PWR1, pwr_reg | 0x00c0);
break;
case SND_SOC_BIAS_PREPARE:
ASoC: wm8753: Integrate capacitor charging into the DAPM sequence When being powered on, either initially on probe or when resuming from suspend, the wm8971 configures the device for quick output capacitor charging. Since the charging can take a rather long time (up to multiple seconds) it is done asynchronously without blocking. A delayed work item is run once the charging is finished and the device is switched to the target bias level. This all done asynchronously to the regular DAPM sequence accessing the same data structures and registers without any looking, which can lead to race conditions. Furthermore this potentially delays the start of stream on the CODEC while the rest of the system is already up and running, meaning the first bytes of audio are lost. It also does no comply with the assumption of the DAPM core that if set_bias_level() returned successfully the device will be at the requested bias level. This patch slightly refactors things and makes sure that the caps charging is properly integrated into the DAPM sequence. When transitioning from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY the part will be put into fast charging mode and a work item will be scheduled that puts it back into standby charging once the charging period has elapsed. If a playback or capture stream is started while charging is in progress the driver will now wait in SND_SOC_BIAS_PREPARE until the charging is done. This makes sure that charging is done asynchronously in the background when the chip is idle, but at the same time makes sure that playback/capture is not started before the charging is done. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-03-30 22:04:48 +03:00
/* Wait until fully charged */
flush_delayed_work(&wm8753->charge_work);
break;
case SND_SOC_BIAS_STANDBY:
if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_OFF) {
ASoC: wm8753: Integrate capacitor charging into the DAPM sequence When being powered on, either initially on probe or when resuming from suspend, the wm8971 configures the device for quick output capacitor charging. Since the charging can take a rather long time (up to multiple seconds) it is done asynchronously without blocking. A delayed work item is run once the charging is finished and the device is switched to the target bias level. This all done asynchronously to the regular DAPM sequence accessing the same data structures and registers without any looking, which can lead to race conditions. Furthermore this potentially delays the start of stream on the CODEC while the rest of the system is already up and running, meaning the first bytes of audio are lost. It also does no comply with the assumption of the DAPM core that if set_bias_level() returned successfully the device will be at the requested bias level. This patch slightly refactors things and makes sure that the caps charging is properly integrated into the DAPM sequence. When transitioning from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY the part will be put into fast charging mode and a work item will be scheduled that puts it back into standby charging once the charging period has elapsed. If a playback or capture stream is started while charging is in progress the driver will now wait in SND_SOC_BIAS_PREPARE until the charging is done. This makes sure that charging is done asynchronously in the background when the chip is idle, but at the same time makes sure that playback/capture is not started before the charging is done. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-03-30 22:04:48 +03:00
/* set vmid to 5k for quick power up */
snd_soc_component_write(component, WM8753_PWR1, pwr_reg | 0x01c1);
ASoC: wm8753: Integrate capacitor charging into the DAPM sequence When being powered on, either initially on probe or when resuming from suspend, the wm8971 configures the device for quick output capacitor charging. Since the charging can take a rather long time (up to multiple seconds) it is done asynchronously without blocking. A delayed work item is run once the charging is finished and the device is switched to the target bias level. This all done asynchronously to the regular DAPM sequence accessing the same data structures and registers without any looking, which can lead to race conditions. Furthermore this potentially delays the start of stream on the CODEC while the rest of the system is already up and running, meaning the first bytes of audio are lost. It also does no comply with the assumption of the DAPM core that if set_bias_level() returned successfully the device will be at the requested bias level. This patch slightly refactors things and makes sure that the caps charging is properly integrated into the DAPM sequence. When transitioning from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY the part will be put into fast charging mode and a work item will be scheduled that puts it back into standby charging once the charging period has elapsed. If a playback or capture stream is started while charging is in progress the driver will now wait in SND_SOC_BIAS_PREPARE until the charging is done. This makes sure that charging is done asynchronously in the background when the chip is idle, but at the same time makes sure that playback/capture is not started before the charging is done. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-03-30 22:04:48 +03:00
schedule_delayed_work(&wm8753->charge_work,
msecs_to_jiffies(caps_charge));
} else {
/* mute dac and set vmid to 500k, enable VREF */
snd_soc_component_write(component, WM8753_PWR1, pwr_reg | 0x0141);
ASoC: wm8753: Integrate capacitor charging into the DAPM sequence When being powered on, either initially on probe or when resuming from suspend, the wm8971 configures the device for quick output capacitor charging. Since the charging can take a rather long time (up to multiple seconds) it is done asynchronously without blocking. A delayed work item is run once the charging is finished and the device is switched to the target bias level. This all done asynchronously to the regular DAPM sequence accessing the same data structures and registers without any looking, which can lead to race conditions. Furthermore this potentially delays the start of stream on the CODEC while the rest of the system is already up and running, meaning the first bytes of audio are lost. It also does no comply with the assumption of the DAPM core that if set_bias_level() returned successfully the device will be at the requested bias level. This patch slightly refactors things and makes sure that the caps charging is properly integrated into the DAPM sequence. When transitioning from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY the part will be put into fast charging mode and a work item will be scheduled that puts it back into standby charging once the charging period has elapsed. If a playback or capture stream is started while charging is in progress the driver will now wait in SND_SOC_BIAS_PREPARE until the charging is done. This makes sure that charging is done asynchronously in the background when the chip is idle, but at the same time makes sure that playback/capture is not started before the charging is done. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-03-30 22:04:48 +03:00
}
break;
case SND_SOC_BIAS_OFF:
ASoC: wm8753: Integrate capacitor charging into the DAPM sequence When being powered on, either initially on probe or when resuming from suspend, the wm8971 configures the device for quick output capacitor charging. Since the charging can take a rather long time (up to multiple seconds) it is done asynchronously without blocking. A delayed work item is run once the charging is finished and the device is switched to the target bias level. This all done asynchronously to the regular DAPM sequence accessing the same data structures and registers without any looking, which can lead to race conditions. Furthermore this potentially delays the start of stream on the CODEC while the rest of the system is already up and running, meaning the first bytes of audio are lost. It also does no comply with the assumption of the DAPM core that if set_bias_level() returned successfully the device will be at the requested bias level. This patch slightly refactors things and makes sure that the caps charging is properly integrated into the DAPM sequence. When transitioning from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY the part will be put into fast charging mode and a work item will be scheduled that puts it back into standby charging once the charging period has elapsed. If a playback or capture stream is started while charging is in progress the driver will now wait in SND_SOC_BIAS_PREPARE until the charging is done. This makes sure that charging is done asynchronously in the background when the chip is idle, but at the same time makes sure that playback/capture is not started before the charging is done. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-03-30 22:04:48 +03:00
cancel_delayed_work_sync(&wm8753->charge_work);
snd_soc_component_write(component, WM8753_PWR1, 0x0001);
break;
}
return 0;
}
#define WM8753_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
#define WM8753_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
SNDRV_PCM_FMTBIT_S24_LE)
/*
* The WM8753 supports up to 4 different and mutually exclusive DAI
* configurations. This gives 2 PCM's available for use, hifi and voice.
* NOTE: The Voice PCM cannot play or capture audio to the CPU as it's DAI
* is connected between the wm8753 and a BT codec or GSM modem.
*
* 1. Voice over PCM DAI - HIFI DAC over HIFI DAI
* 2. Voice over HIFI DAI - HIFI disabled
* 3. Voice disabled - HIFI over HIFI
* 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
*/
static const struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = {
.hw_params = wm8753_i2s_hw_params,
.mute_stream = wm8753_mute,
.set_fmt = wm8753_hifi_set_dai_fmt,
.set_clkdiv = wm8753_set_dai_clkdiv,
.set_pll = wm8753_set_dai_pll,
.set_sysclk = wm8753_set_dai_sysclk,
.no_capture_mute = 1,
};
static const struct snd_soc_dai_ops wm8753_dai_ops_voice_mode = {
.hw_params = wm8753_pcm_hw_params,
.mute_stream = wm8753_mute,
.set_fmt = wm8753_voice_set_dai_fmt,
.set_clkdiv = wm8753_set_dai_clkdiv,
.set_pll = wm8753_set_dai_pll,
.set_sysclk = wm8753_set_dai_sysclk,
.no_capture_mute = 1,
};
static struct snd_soc_dai_driver wm8753_dai[] = {
/* DAI HiFi mode 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 23:15:21 +03:00
{ .name = "wm8753-hifi",
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
.channels_max = 2,
.rates = WM8753_RATES,
.formats = WM8753_FORMATS
},
.capture = { /* dummy for fast DAI switching */
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8753_RATES,
.formats = WM8753_FORMATS
},
.ops = &wm8753_dai_ops_hifi_mode,
},
/* DAI Voice mode 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 23:15:21 +03:00
{ .name = "wm8753-voice",
.playback = {
.stream_name = "Voice Playback",
.channels_min = 1,
.channels_max = 1,
.rates = WM8753_RATES,
.formats = WM8753_FORMATS,
},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8753_RATES,
.formats = WM8753_FORMATS,
},
.ops = &wm8753_dai_ops_voice_mode,
},
};
static int wm8753_resume(struct snd_soc_component *component)
{
struct wm8753_priv *wm8753 = snd_soc_component_get_drvdata(component);
regcache_sync(wm8753->regmap);
return 0;
}
static int wm8753_probe(struct snd_soc_component *component)
{
struct wm8753_priv *wm8753 = snd_soc_component_get_drvdata(component);
int ret;
ASoC: wm8753: Integrate capacitor charging into the DAPM sequence When being powered on, either initially on probe or when resuming from suspend, the wm8971 configures the device for quick output capacitor charging. Since the charging can take a rather long time (up to multiple seconds) it is done asynchronously without blocking. A delayed work item is run once the charging is finished and the device is switched to the target bias level. This all done asynchronously to the regular DAPM sequence accessing the same data structures and registers without any looking, which can lead to race conditions. Furthermore this potentially delays the start of stream on the CODEC while the rest of the system is already up and running, meaning the first bytes of audio are lost. It also does no comply with the assumption of the DAPM core that if set_bias_level() returned successfully the device will be at the requested bias level. This patch slightly refactors things and makes sure that the caps charging is properly integrated into the DAPM sequence. When transitioning from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY the part will be put into fast charging mode and a work item will be scheduled that puts it back into standby charging once the charging period has elapsed. If a playback or capture stream is started while charging is in progress the driver will now wait in SND_SOC_BIAS_PREPARE until the charging is done. This makes sure that charging is done asynchronously in the background when the chip is idle, but at the same time makes sure that playback/capture is not started before the charging is done. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie@kernel.org>
2015-03-30 22:04:48 +03:00
INIT_DELAYED_WORK(&wm8753->charge_work, wm8753_charge_work);
ret = wm8753_reset(component);
if (ret < 0) {
dev_err(component->dev, "Failed to issue reset: %d\n", ret);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
return ret;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
wm8753->dai_func = 0;
/* set the update bits */
snd_soc_component_update_bits(component, WM8753_LDAC, 0x0100, 0x0100);
snd_soc_component_update_bits(component, WM8753_RDAC, 0x0100, 0x0100);
snd_soc_component_update_bits(component, WM8753_LADC, 0x0100, 0x0100);
snd_soc_component_update_bits(component, WM8753_RADC, 0x0100, 0x0100);
snd_soc_component_update_bits(component, WM8753_LOUT1V, 0x0100, 0x0100);
snd_soc_component_update_bits(component, WM8753_ROUT1V, 0x0100, 0x0100);
snd_soc_component_update_bits(component, WM8753_LOUT2V, 0x0100, 0x0100);
snd_soc_component_update_bits(component, WM8753_ROUT2V, 0x0100, 0x0100);
snd_soc_component_update_bits(component, WM8753_LINVOL, 0x0100, 0x0100);
snd_soc_component_update_bits(component, WM8753_RINVOL, 0x0100, 0x0100);
return 0;
}
static const struct snd_soc_component_driver soc_component_dev_wm8753 = {
.probe = wm8753_probe,
.resume = wm8753_resume,
.set_bias_level = wm8753_set_bias_level,
.controls = wm8753_snd_controls,
.num_controls = ARRAY_SIZE(wm8753_snd_controls),
.dapm_widgets = wm8753_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm8753_dapm_widgets),
.dapm_routes = wm8753_dapm_routes,
.num_dapm_routes = ARRAY_SIZE(wm8753_dapm_routes),
.suspend_bias_off = 1,
.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 23:15:21 +03:00
};
static const struct of_device_id wm8753_of_match[] = {
{ .compatible = "wlf,wm8753", },
{ }
};
MODULE_DEVICE_TABLE(of, wm8753_of_match);
static const struct regmap_config wm8753_regmap = {
.reg_bits = 7,
.val_bits = 9,
.max_register = WM8753_ADCTL2,
.volatile_reg = wm8753_volatile,
.cache_type = REGCACHE_RBTREE,
.reg_defaults = wm8753_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(wm8753_reg_defaults),
};
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
#if defined(CONFIG_SPI_MASTER)
static int wm8753_spi_probe(struct spi_device *spi)
{
struct wm8753_priv *wm8753;
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 23:15:21 +03:00
int ret;
wm8753 = devm_kzalloc(&spi->dev, sizeof(struct wm8753_priv),
GFP_KERNEL);
if (wm8753 == NULL)
return -ENOMEM;
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 23:15:21 +03:00
spi_set_drvdata(spi, wm8753);
wm8753->regmap = devm_regmap_init_spi(spi, &wm8753_regmap);
if (IS_ERR(wm8753->regmap)) {
ret = PTR_ERR(wm8753->regmap);
dev_err(&spi->dev, "Failed to allocate register map: %d\n",
ret);
return ret;
}
ret = devm_snd_soc_register_component(&spi->dev, &soc_component_dev_wm8753,
wm8753_dai, ARRAY_SIZE(wm8753_dai));
if (ret != 0)
dev_err(&spi->dev, "Failed to register CODEC: %d\n", ret);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
return ret;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
static struct spi_driver wm8753_spi_driver = {
.driver = {
.name = "wm8753",
.of_match_table = wm8753_of_match,
},
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 23:15:21 +03:00
.probe = wm8753_spi_probe,
};
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 23:15:21 +03:00
#endif /* CONFIG_SPI_MASTER */
#if IS_ENABLED(CONFIG_I2C)
static int wm8753_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct wm8753_priv *wm8753;
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 23:15:21 +03:00
int ret;
wm8753 = devm_kzalloc(&i2c->dev, sizeof(struct wm8753_priv),
GFP_KERNEL);
if (wm8753 == NULL)
return -ENOMEM;
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 23:15:21 +03:00
i2c_set_clientdata(i2c, wm8753);
wm8753->regmap = devm_regmap_init_i2c(i2c, &wm8753_regmap);
if (IS_ERR(wm8753->regmap)) {
ret = PTR_ERR(wm8753->regmap);
dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
ret);
return ret;
}
ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_wm8753,
wm8753_dai, ARRAY_SIZE(wm8753_dai));
if (ret != 0)
dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
return ret;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
static const struct i2c_device_id wm8753_i2c_id[] = {
{ "wm8753", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id);
static struct i2c_driver wm8753_i2c_driver = {
.driver = {
.name = "wm8753",
.of_match_table = wm8753_of_match,
},
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 23:15:21 +03:00
.probe = wm8753_i2c_probe,
.id_table = wm8753_i2c_id,
};
#endif
static int __init wm8753_modinit(void)
{
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 23:15:21 +03:00
int ret = 0;
#if IS_ENABLED(CONFIG_I2C)
ret = i2c_add_driver(&wm8753_i2c_driver);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
if (ret != 0) {
printk(KERN_ERR "Failed to register wm8753 I2C driver: %d\n",
ret);
}
#endif
#if defined(CONFIG_SPI_MASTER)
ret = spi_register_driver(&wm8753_spi_driver);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-17 23:15:21 +03:00
if (ret != 0) {
printk(KERN_ERR "Failed to register wm8753 SPI driver: %d\n",
ret);
}
#endif
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 23:15:21 +03:00
return ret;
}
module_init(wm8753_modinit);
static void __exit wm8753_exit(void)
{
#if IS_ENABLED(CONFIG_I2C)
i2c_del_driver(&wm8753_i2c_driver);
#endif
#if defined(CONFIG_SPI_MASTER)
spi_unregister_driver(&wm8753_spi_driver);
#endif
}
module_exit(wm8753_exit);
MODULE_DESCRIPTION("ASoC WM8753 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");