linux/sound/soc/soc-dapm.c

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[ALSA] ASoC: dynamic audio power management (DAPM) This patch adds Dynamic Audio Power Management (DAPM) to ASoC. Dynamic Audio Power Management (DAPM) is designed to allow portable and handheld Linux devices to use the minimum amount of power within the audio subsystem at all times. It is independent of other kernel PM and as such, can easily co-exist with the other PM systems. DAPM is also completely transparent to all user space applications as all power switching is done within the ASoC core. No code changes or recompiling are required for user space applications. DAPM makes power switching decisions based upon any audio stream (capture/playback) activity and audio mixer settings within the device. DAPM spans the whole machine. It covers power control within the entire audio subsystem, this includes internal codec power blocks and machine level power systems. There are 4 power domains within DAPM:- 1. Codec domain - VREF, VMID (core codec and audio power) Usually controlled at codec probe/remove and suspend/resume, although can be set at stream time if power is not needed for sidetone, etc. 2. Platform/Machine domain - physically connected inputs and outputs Is platform/machine and user action specific, is configured by the machine driver and responds to asynchronous events e.g when HP are inserted 3. Path domain - audio subsystem signal paths Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. 4. Stream domain - DAC's and ADC's. Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. All DAPM power switching decisions are made automatically by consulting an audio routing map of the whole machine. This map is specific to each machine and consists of the interconnections between every audio component (including internal codec components). Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-07 00:32:18 +08:00
/*
* soc-dapm.c -- ALSA SoC Dynamic Audio Power Management
*
* Copyright 2005 Wolfson Microelectronics PLC.
* Author: Liam Girdwood
* liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
* Revision history
* 12th Aug 2005 Initial version.
* 25th Oct 2005 Implemented path power domain.
* 18th Dec 2005 Implemented machine and stream level power domain.
*
* Features:
* o Changes power status of internal codec blocks depending on the
* dynamic configuration of codec internal audio paths and active
* DAC's/ADC's.
* o Platform power domain - can support external components i.e. amps and
* mic/meadphone insertion events.
* o Automatic Mic Bias support
* o Jack insertion power event initiation - e.g. hp insertion will enable
* sinks, dacs, etc
* o Delayed powerdown of audio susbsytem to reduce pops between a quick
* device reopen.
*
* Todo:
* o DAPM power change sequencing - allow for configurable per
* codec sequences.
* o Support for analogue bias optimisation.
* o Support for reduced codec oversampling rates.
* o Support for reduced codec bias currents.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/bitops.h>
#include <linux/platform_device.h>
#include <linux/jiffies.h>
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc-dapm.h>
#include <sound/initval.h>
/* debug */
#define DAPM_DEBUG 0
#if DAPM_DEBUG
#define dump_dapm(codec, action) dbg_dump_dapm(codec, action)
#define dbg(format, arg...) printk(format, ## arg)
#else
#define dump_dapm(codec, action)
#define dbg(format, arg...)
#endif
#define POP_DEBUG 0
#if POP_DEBUG
#define POP_TIME 500 /* 500 msecs - change if pop debug is too fast */
#define pop_wait(time) schedule_timeout_interruptible(msecs_to_jiffies(time))
#define pop_dbg(format, arg...) printk(format, ## arg); pop_wait(POP_TIME)
#else
#define pop_dbg(format, arg...)
#define pop_wait(time)
#endif
/* dapm power sequences - make this per codec in the future */
static int dapm_up_seq[] = {
snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic,
snd_soc_dapm_mux, snd_soc_dapm_dac, snd_soc_dapm_mixer, snd_soc_dapm_pga,
snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post
};
static int dapm_down_seq[] = {
snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
snd_soc_dapm_pga, snd_soc_dapm_mixer, snd_soc_dapm_dac, snd_soc_dapm_mic,
snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_post
};
static int dapm_status = 1;
module_param(dapm_status, int, 0);
MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries");
/* create a new dapm widget */
static inline struct snd_soc_dapm_widget *dapm_cnew_widget(
[ALSA] ASoC: dynamic audio power management (DAPM) This patch adds Dynamic Audio Power Management (DAPM) to ASoC. Dynamic Audio Power Management (DAPM) is designed to allow portable and handheld Linux devices to use the minimum amount of power within the audio subsystem at all times. It is independent of other kernel PM and as such, can easily co-exist with the other PM systems. DAPM is also completely transparent to all user space applications as all power switching is done within the ASoC core. No code changes or recompiling are required for user space applications. DAPM makes power switching decisions based upon any audio stream (capture/playback) activity and audio mixer settings within the device. DAPM spans the whole machine. It covers power control within the entire audio subsystem, this includes internal codec power blocks and machine level power systems. There are 4 power domains within DAPM:- 1. Codec domain - VREF, VMID (core codec and audio power) Usually controlled at codec probe/remove and suspend/resume, although can be set at stream time if power is not needed for sidetone, etc. 2. Platform/Machine domain - physically connected inputs and outputs Is platform/machine and user action specific, is configured by the machine driver and responds to asynchronous events e.g when HP are inserted 3. Path domain - audio subsystem signal paths Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. 4. Stream domain - DAC's and ADC's. Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. All DAPM power switching decisions are made automatically by consulting an audio routing map of the whole machine. This map is specific to each machine and consists of the interconnections between every audio component (including internal codec components). Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-07 00:32:18 +08:00
const struct snd_soc_dapm_widget *_widget)
{
return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL);
[ALSA] ASoC: dynamic audio power management (DAPM) This patch adds Dynamic Audio Power Management (DAPM) to ASoC. Dynamic Audio Power Management (DAPM) is designed to allow portable and handheld Linux devices to use the minimum amount of power within the audio subsystem at all times. It is independent of other kernel PM and as such, can easily co-exist with the other PM systems. DAPM is also completely transparent to all user space applications as all power switching is done within the ASoC core. No code changes or recompiling are required for user space applications. DAPM makes power switching decisions based upon any audio stream (capture/playback) activity and audio mixer settings within the device. DAPM spans the whole machine. It covers power control within the entire audio subsystem, this includes internal codec power blocks and machine level power systems. There are 4 power domains within DAPM:- 1. Codec domain - VREF, VMID (core codec and audio power) Usually controlled at codec probe/remove and suspend/resume, although can be set at stream time if power is not needed for sidetone, etc. 2. Platform/Machine domain - physically connected inputs and outputs Is platform/machine and user action specific, is configured by the machine driver and responds to asynchronous events e.g when HP are inserted 3. Path domain - audio subsystem signal paths Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. 4. Stream domain - DAC's and ADC's. Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. All DAPM power switching decisions are made automatically by consulting an audio routing map of the whole machine. This map is specific to each machine and consists of the interconnections between every audio component (including internal codec components). Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-07 00:32:18 +08:00
}
/* set up initial codec paths */
static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
struct snd_soc_dapm_path *p, int i)
{
switch (w->id) {
case snd_soc_dapm_switch:
case snd_soc_dapm_mixer: {
int val;
int reg = w->kcontrols[i].private_value & 0xff;
int shift = (w->kcontrols[i].private_value >> 8) & 0x0f;
int mask = (w->kcontrols[i].private_value >> 16) & 0xff;
int invert = (w->kcontrols[i].private_value >> 24) & 0x01;
val = snd_soc_read(w->codec, reg);
val = (val >> shift) & mask;
if ((invert && !val) || (!invert && val))
p->connect = 1;
else
p->connect = 0;
}
break;
case snd_soc_dapm_mux: {
struct soc_enum *e = (struct soc_enum *)w->kcontrols[i].private_value;
int val, item, bitmask;
for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
;
val = snd_soc_read(w->codec, e->reg);
item = (val >> e->shift_l) & (bitmask - 1);
p->connect = 0;
for (i = 0; i < e->mask; i++) {
if (!(strcmp(p->name, e->texts[i])) && item == i)
p->connect = 1;
}
}
break;
/* does not effect routing - always connected */
case snd_soc_dapm_pga:
case snd_soc_dapm_output:
case snd_soc_dapm_adc:
case snd_soc_dapm_input:
case snd_soc_dapm_dac:
case snd_soc_dapm_micbias:
case snd_soc_dapm_vmid:
p->connect = 1;
break;
/* does effect routing - dynamically connected */
case snd_soc_dapm_hp:
case snd_soc_dapm_mic:
case snd_soc_dapm_spk:
case snd_soc_dapm_line:
case snd_soc_dapm_pre:
case snd_soc_dapm_post:
p->connect = 0;
break;
}
}
/* connect mux widget to it's interconnecting audio paths */
static int dapm_connect_mux(struct snd_soc_codec *codec,
struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
struct snd_soc_dapm_path *path, const char *control_name,
const struct snd_kcontrol_new *kcontrol)
{
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
int i;
for (i = 0; i < e->mask; i++) {
if (!(strcmp(control_name, e->texts[i]))) {
list_add(&path->list, &codec->dapm_paths);
list_add(&path->list_sink, &dest->sources);
list_add(&path->list_source, &src->sinks);
path->name = (char*)e->texts[i];
dapm_set_path_status(dest, path, 0);
return 0;
}
}
return -ENODEV;
}
/* connect mixer widget to it's interconnecting audio paths */
static int dapm_connect_mixer(struct snd_soc_codec *codec,
struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest,
struct snd_soc_dapm_path *path, const char *control_name)
{
int i;
/* search for mixer kcontrol */
for (i = 0; i < dest->num_kcontrols; i++) {
if (!strcmp(control_name, dest->kcontrols[i].name)) {
list_add(&path->list, &codec->dapm_paths);
list_add(&path->list_sink, &dest->sources);
list_add(&path->list_source, &src->sinks);
path->name = dest->kcontrols[i].name;
dapm_set_path_status(dest, path, i);
return 0;
}
}
return -ENODEV;
}
/* update dapm codec register bits */
static int dapm_update_bits(struct snd_soc_dapm_widget *widget)
{
int change, power;
unsigned short old, new;
struct snd_soc_codec *codec = widget->codec;
/* check for valid widgets */
if (widget->reg < 0 || widget->id == snd_soc_dapm_input ||
widget->id == snd_soc_dapm_output ||
widget->id == snd_soc_dapm_hp ||
widget->id == snd_soc_dapm_mic ||
widget->id == snd_soc_dapm_line ||
widget->id == snd_soc_dapm_spk)
return 0;
power = widget->power;
if (widget->invert)
power = (power ? 0:1);
old = snd_soc_read(codec, widget->reg);
new = (old & ~(0x1 << widget->shift)) | (power << widget->shift);
change = old != new;
if (change) {
pop_dbg("pop test %s : %s in %d ms\n", widget->name,
widget->power ? "on" : "off", POP_TIME);
snd_soc_write(codec, widget->reg, new);
pop_wait(POP_TIME);
}
dbg("reg old %x new %x change %d\n", old, new, change);
return change;
}
/* ramps the volume up or down to minimise pops before or after a
* DAPM power event */
static int dapm_set_pga(struct snd_soc_dapm_widget *widget, int power)
{
const struct snd_kcontrol_new *k = widget->kcontrols;
if (widget->muted && !power)
return 0;
if (!widget->muted && power)
return 0;
if (widget->num_kcontrols && k) {
int reg = k->private_value & 0xff;
int shift = (k->private_value >> 8) & 0x0f;
int mask = (k->private_value >> 16) & 0xff;
int invert = (k->private_value >> 24) & 0x01;
if (power) {
int i;
/* power up has happended, increase volume to last level */
if (invert) {
for (i = mask; i > widget->saved_value; i--)
snd_soc_update_bits(widget->codec, reg, mask, i);
} else {
for (i = 0; i < widget->saved_value; i++)
snd_soc_update_bits(widget->codec, reg, mask, i);
}
widget->muted = 0;
} else {
/* power down is about to occur, decrease volume to mute */
int val = snd_soc_read(widget->codec, reg);
int i = widget->saved_value = (val >> shift) & mask;
if (invert) {
for (; i < mask; i++)
snd_soc_update_bits(widget->codec, reg, mask, i);
} else {
for (; i > 0; i--)
snd_soc_update_bits(widget->codec, reg, mask, i);
}
widget->muted = 1;
}
}
return 0;
}
/* create new dapm mixer control */
static int dapm_new_mixer(struct snd_soc_codec *codec,
struct snd_soc_dapm_widget *w)
{
int i, ret = 0;
char name[32];
struct snd_soc_dapm_path *path;
/* add kcontrol */
for (i = 0; i < w->num_kcontrols; i++) {
/* match name */
list_for_each_entry(path, &w->sources, list_sink) {
/* mixer/mux paths name must match control name */
if (path->name != (char*)w->kcontrols[i].name)
continue;
/* add dapm control with long name */
snprintf(name, 32, "%s %s", w->name, w->kcontrols[i].name);
path->long_name = kstrdup (name, GFP_KERNEL);
if (path->long_name == NULL)
return -ENOMEM;
path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
path->long_name);
ret = snd_ctl_add(codec->card, path->kcontrol);
if (ret < 0) {
printk(KERN_ERR "asoc: failed to add dapm kcontrol %s\n",
path->long_name);
kfree(path->long_name);
path->long_name = NULL;
return ret;
}
}
}
return ret;
}
/* create new dapm mux control */
static int dapm_new_mux(struct snd_soc_codec *codec,
struct snd_soc_dapm_widget *w)
{
struct snd_soc_dapm_path *path = NULL;
struct snd_kcontrol *kcontrol;
int ret = 0;
if (!w->num_kcontrols) {
printk(KERN_ERR "asoc: mux %s has no controls\n", w->name);
return -EINVAL;
}
kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
ret = snd_ctl_add(codec->card, kcontrol);
if (ret < 0)
goto err;
list_for_each_entry(path, &w->sources, list_sink)
path->kcontrol = kcontrol;
return ret;
err:
printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name);
return ret;
}
/* create new dapm volume control */
static int dapm_new_pga(struct snd_soc_codec *codec,
struct snd_soc_dapm_widget *w)
{
struct snd_kcontrol *kcontrol;
int ret = 0;
if (!w->num_kcontrols)
return -EINVAL;
kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
ret = snd_ctl_add(codec->card, kcontrol);
if (ret < 0) {
printk(KERN_ERR "asoc: failed to add kcontrol %s\n", w->name);
return ret;
}
return ret;
}
/* reset 'walked' bit for each dapm path */
static inline void dapm_clear_walk(struct snd_soc_codec *codec)
{
struct snd_soc_dapm_path *p;
list_for_each_entry(p, &codec->dapm_paths, list)
p->walked = 0;
}
/*
* Recursively check for a completed path to an active or physically connected
* output widget. Returns number of complete paths.
*/
static int is_connected_output_ep(struct snd_soc_dapm_widget *widget)
{
struct snd_soc_dapm_path *path;
int con = 0;
if (widget->id == snd_soc_dapm_adc && widget->active)
return 1;
if (widget->connected) {
/* connected pin ? */
if (widget->id == snd_soc_dapm_output && !widget->ext)
return 1;
/* connected jack or spk ? */
if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk ||
widget->id == snd_soc_dapm_line)
return 1;
}
list_for_each_entry(path, &widget->sinks, list_source) {
if (path->walked)
continue;
if (path->sink && path->connect) {
path->walked = 1;
con += is_connected_output_ep(path->sink);
}
}
return con;
}
/*
* Recursively check for a completed path to an active or physically connected
* input widget. Returns number of complete paths.
*/
static int is_connected_input_ep(struct snd_soc_dapm_widget *widget)
{
struct snd_soc_dapm_path *path;
int con = 0;
/* active stream ? */
if (widget->id == snd_soc_dapm_dac && widget->active)
return 1;
if (widget->connected) {
/* connected pin ? */
if (widget->id == snd_soc_dapm_input && !widget->ext)
return 1;
/* connected VMID/Bias for lower pops */
if (widget->id == snd_soc_dapm_vmid)
return 1;
/* connected jack ? */
if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line)
return 1;
}
list_for_each_entry(path, &widget->sources, list_sink) {
if (path->walked)
continue;
if (path->source && path->connect) {
path->walked = 1;
con += is_connected_input_ep(path->source);
}
}
return con;
}
/*
* Scan each dapm widget for complete audio path.
* A complete path is a route that has valid endpoints i.e.:-
*
* o DAC to output pin.
* o Input Pin to ADC.
* o Input pin to Output pin (bypass, sidetone)
* o DAC to ADC (loopback).
*/
static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
[ALSA] ASoC: dynamic audio power management (DAPM) This patch adds Dynamic Audio Power Management (DAPM) to ASoC. Dynamic Audio Power Management (DAPM) is designed to allow portable and handheld Linux devices to use the minimum amount of power within the audio subsystem at all times. It is independent of other kernel PM and as such, can easily co-exist with the other PM systems. DAPM is also completely transparent to all user space applications as all power switching is done within the ASoC core. No code changes or recompiling are required for user space applications. DAPM makes power switching decisions based upon any audio stream (capture/playback) activity and audio mixer settings within the device. DAPM spans the whole machine. It covers power control within the entire audio subsystem, this includes internal codec power blocks and machine level power systems. There are 4 power domains within DAPM:- 1. Codec domain - VREF, VMID (core codec and audio power) Usually controlled at codec probe/remove and suspend/resume, although can be set at stream time if power is not needed for sidetone, etc. 2. Platform/Machine domain - physically connected inputs and outputs Is platform/machine and user action specific, is configured by the machine driver and responds to asynchronous events e.g when HP are inserted 3. Path domain - audio subsystem signal paths Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. 4. Stream domain - DAC's and ADC's. Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. All DAPM power switching decisions are made automatically by consulting an audio routing map of the whole machine. This map is specific to each machine and consists of the interconnections between every audio component (including internal codec components). Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-07 00:32:18 +08:00
{
struct snd_soc_dapm_widget *w;
int in, out, i, c = 1, *seq = NULL, ret = 0, power_change, power;
/* do we have a sequenced stream event */
if (event == SND_SOC_DAPM_STREAM_START) {
c = ARRAY_SIZE(dapm_up_seq);
seq = dapm_up_seq;
} else if (event == SND_SOC_DAPM_STREAM_STOP) {
c = ARRAY_SIZE(dapm_down_seq);
seq = dapm_down_seq;
}
for(i = 0; i < c; i++) {
list_for_each_entry(w, &codec->dapm_widgets, list) {
/* is widget in stream order */
if (seq && seq[i] && w->id != seq[i])
continue;
/* vmid - no action */
if (w->id == snd_soc_dapm_vmid)
continue;
/* active ADC */
if (w->id == snd_soc_dapm_adc && w->active) {
in = is_connected_input_ep(w);
dapm_clear_walk(w->codec);
w->power = (in != 0) ? 1 : 0;
dapm_update_bits(w);
continue;
}
/* active DAC */
if (w->id == snd_soc_dapm_dac && w->active) {
out = is_connected_output_ep(w);
dapm_clear_walk(w->codec);
w->power = (out != 0) ? 1 : 0;
dapm_update_bits(w);
continue;
}
/* programmable gain/attenuation */
if (w->id == snd_soc_dapm_pga) {
int on;
in = is_connected_input_ep(w);
dapm_clear_walk(w->codec);
out = is_connected_output_ep(w);
dapm_clear_walk(w->codec);
w->power = on = (out != 0 && in != 0) ? 1 : 0;
if (!on)
dapm_set_pga(w, on); /* lower volume to reduce pops */
dapm_update_bits(w);
if (on)
dapm_set_pga(w, on); /* restore volume from zero */
continue;
}
/* pre and post event widgets */
if (w->id == snd_soc_dapm_pre) {
if (!w->event)
continue;
if (event == SND_SOC_DAPM_STREAM_START) {
ret = w->event(w, SND_SOC_DAPM_PRE_PMU);
if (ret < 0)
return ret;
} else if (event == SND_SOC_DAPM_STREAM_STOP) {
ret = w->event(w, SND_SOC_DAPM_PRE_PMD);
if (ret < 0)
return ret;
}
continue;
}
if (w->id == snd_soc_dapm_post) {
if (!w->event)
continue;
if (event == SND_SOC_DAPM_STREAM_START) {
ret = w->event(w, SND_SOC_DAPM_POST_PMU);
if (ret < 0)
return ret;
} else if (event == SND_SOC_DAPM_STREAM_STOP) {
ret = w->event(w, SND_SOC_DAPM_POST_PMD);
if (ret < 0)
return ret;
}
continue;
}
/* all other widgets */
in = is_connected_input_ep(w);
dapm_clear_walk(w->codec);
out = is_connected_output_ep(w);
dapm_clear_walk(w->codec);
power = (out != 0 && in != 0) ? 1 : 0;
power_change = (w->power == power) ? 0: 1;
w->power = power;
/* call any power change event handlers */
if (power_change) {
if (w->event) {
dbg("power %s event for %s flags %x\n",
w->power ? "on" : "off", w->name, w->event_flags);
if (power) {
/* power up event */
if (w->event_flags & SND_SOC_DAPM_PRE_PMU) {
ret = w->event(w, SND_SOC_DAPM_PRE_PMU);
if (ret < 0)
return ret;
}
dapm_update_bits(w);
if (w->event_flags & SND_SOC_DAPM_POST_PMU){
ret = w->event(w, SND_SOC_DAPM_POST_PMU);
if (ret < 0)
return ret;
}
} else {
/* power down event */
if (w->event_flags & SND_SOC_DAPM_PRE_PMD) {
ret = w->event(w, SND_SOC_DAPM_PRE_PMD);
if (ret < 0)
return ret;
}
dapm_update_bits(w);
if (w->event_flags & SND_SOC_DAPM_POST_PMD) {
ret = w->event(w, SND_SOC_DAPM_POST_PMD);
if (ret < 0)
return ret;
}
}
} else
/* no event handler */
dapm_update_bits(w);
}
}
}
return ret;
}
#if DAPM_DEBUG
static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action)
{
struct snd_soc_dapm_widget *w;
struct snd_soc_dapm_path *p = NULL;
int in, out;
printk("DAPM %s %s\n", codec->name, action);
list_for_each_entry(w, &codec->dapm_widgets, list) {
/* only display widgets that effect routing */
switch (w->id) {
case snd_soc_dapm_pre:
case snd_soc_dapm_post:
case snd_soc_dapm_vmid:
continue;
case snd_soc_dapm_mux:
case snd_soc_dapm_output:
case snd_soc_dapm_input:
case snd_soc_dapm_switch:
case snd_soc_dapm_hp:
case snd_soc_dapm_mic:
case snd_soc_dapm_spk:
case snd_soc_dapm_line:
case snd_soc_dapm_micbias:
case snd_soc_dapm_dac:
case snd_soc_dapm_adc:
case snd_soc_dapm_pga:
case snd_soc_dapm_mixer:
if (w->name) {
in = is_connected_input_ep(w);
dapm_clear_walk(w->codec);
out = is_connected_output_ep(w);
dapm_clear_walk(w->codec);
printk("%s: %s in %d out %d\n", w->name,
w->power ? "On":"Off",in, out);
list_for_each_entry(p, &w->sources, list_sink) {
if (p->connect)
printk(" in %s %s\n", p->name ? p->name : "static",
p->source->name);
}
list_for_each_entry(p, &w->sinks, list_source) {
if (p->connect)
printk(" out %s %s\n", p->name ? p->name : "static",
p->sink->name);
}
}
break;
}
}
}
#endif
/* test and update the power status of a mux widget */
static int dapm_mux_update_power(struct snd_soc_dapm_widget *widget,
struct snd_kcontrol *kcontrol, int mask,
int val, struct soc_enum* e)
[ALSA] ASoC: dynamic audio power management (DAPM) This patch adds Dynamic Audio Power Management (DAPM) to ASoC. Dynamic Audio Power Management (DAPM) is designed to allow portable and handheld Linux devices to use the minimum amount of power within the audio subsystem at all times. It is independent of other kernel PM and as such, can easily co-exist with the other PM systems. DAPM is also completely transparent to all user space applications as all power switching is done within the ASoC core. No code changes or recompiling are required for user space applications. DAPM makes power switching decisions based upon any audio stream (capture/playback) activity and audio mixer settings within the device. DAPM spans the whole machine. It covers power control within the entire audio subsystem, this includes internal codec power blocks and machine level power systems. There are 4 power domains within DAPM:- 1. Codec domain - VREF, VMID (core codec and audio power) Usually controlled at codec probe/remove and suspend/resume, although can be set at stream time if power is not needed for sidetone, etc. 2. Platform/Machine domain - physically connected inputs and outputs Is platform/machine and user action specific, is configured by the machine driver and responds to asynchronous events e.g when HP are inserted 3. Path domain - audio subsystem signal paths Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. 4. Stream domain - DAC's and ADC's. Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. All DAPM power switching decisions are made automatically by consulting an audio routing map of the whole machine. This map is specific to each machine and consists of the interconnections between every audio component (including internal codec components). Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-07 00:32:18 +08:00
{
struct snd_soc_dapm_path *path;
int found = 0;
if (widget->id != snd_soc_dapm_mux)
return -ENODEV;
if (!snd_soc_test_bits(widget->codec, e->reg, mask, val))
return 0;
/* find dapm widget path assoc with kcontrol */
list_for_each_entry(path, &widget->codec->dapm_paths, list) {
if (path->kcontrol != kcontrol)
continue;
if (!path->name || ! e->texts[val])
continue;
found = 1;
/* we now need to match the string in the enum to the path */
if (!(strcmp(path->name, e->texts[val])))
path->connect = 1; /* new connection */
else
path->connect = 0; /* old connection must be powered down */
}
if (found)
dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
return 0;
}
/* test and update the power status of a mixer widget */
static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
struct snd_kcontrol *kcontrol, int reg,
int val_mask, int val, int invert)
[ALSA] ASoC: dynamic audio power management (DAPM) This patch adds Dynamic Audio Power Management (DAPM) to ASoC. Dynamic Audio Power Management (DAPM) is designed to allow portable and handheld Linux devices to use the minimum amount of power within the audio subsystem at all times. It is independent of other kernel PM and as such, can easily co-exist with the other PM systems. DAPM is also completely transparent to all user space applications as all power switching is done within the ASoC core. No code changes or recompiling are required for user space applications. DAPM makes power switching decisions based upon any audio stream (capture/playback) activity and audio mixer settings within the device. DAPM spans the whole machine. It covers power control within the entire audio subsystem, this includes internal codec power blocks and machine level power systems. There are 4 power domains within DAPM:- 1. Codec domain - VREF, VMID (core codec and audio power) Usually controlled at codec probe/remove and suspend/resume, although can be set at stream time if power is not needed for sidetone, etc. 2. Platform/Machine domain - physically connected inputs and outputs Is platform/machine and user action specific, is configured by the machine driver and responds to asynchronous events e.g when HP are inserted 3. Path domain - audio subsystem signal paths Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. 4. Stream domain - DAC's and ADC's. Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. All DAPM power switching decisions are made automatically by consulting an audio routing map of the whole machine. This map is specific to each machine and consists of the interconnections between every audio component (including internal codec components). Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-07 00:32:18 +08:00
{
struct snd_soc_dapm_path *path;
int found = 0;
if (widget->id != snd_soc_dapm_mixer)
return -ENODEV;
if (!snd_soc_test_bits(widget->codec, reg, val_mask, val))
return 0;
/* find dapm widget path assoc with kcontrol */
list_for_each_entry(path, &widget->codec->dapm_paths, list) {
if (path->kcontrol != kcontrol)
continue;
/* found, now check type */
found = 1;
if (val)
/* new connection */
path->connect = invert ? 0:1;
else
/* old connection must be powered down */
path->connect = invert ? 1:0;
break;
}
if (found)
dapm_power_widgets(widget->codec, SND_SOC_DAPM_STREAM_NOP);
return 0;
}
/* show dapm widget status in sys fs */
static ssize_t dapm_widget_show(struct device *dev,
struct device_attribute *attr, char *buf)
{
struct snd_soc_device *devdata = dev_get_drvdata(dev);
struct snd_soc_codec *codec = devdata->codec;
struct snd_soc_dapm_widget *w;
int count = 0;
char *state = "not set";
list_for_each_entry(w, &codec->dapm_widgets, list) {
/* only display widgets that burnm power */
switch (w->id) {
case snd_soc_dapm_hp:
case snd_soc_dapm_mic:
case snd_soc_dapm_spk:
case snd_soc_dapm_line:
case snd_soc_dapm_micbias:
case snd_soc_dapm_dac:
case snd_soc_dapm_adc:
case snd_soc_dapm_pga:
case snd_soc_dapm_mixer:
if (w->name)
count += sprintf(buf + count, "%s: %s\n",
w->name, w->power ? "On":"Off");
break;
default:
break;
}
}
switch(codec->dapm_state){
case SNDRV_CTL_POWER_D0:
state = "D0";
break;
case SNDRV_CTL_POWER_D1:
state = "D1";
break;
case SNDRV_CTL_POWER_D2:
state = "D2";
break;
case SNDRV_CTL_POWER_D3hot:
state = "D3hot";
break;
case SNDRV_CTL_POWER_D3cold:
state = "D3cold";
break;
}
count += sprintf(buf + count, "PM State: %s\n", state);
return count;
}
static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL);
int snd_soc_dapm_sys_add(struct device *dev)
{
int ret = 0;
if (dapm_status)
ret = device_create_file(dev, &dev_attr_dapm_widget);
return ret;
}
static void snd_soc_dapm_sys_remove(struct device *dev)
{
if (dapm_status)
device_remove_file(dev, &dev_attr_dapm_widget);
}
/* free all dapm widgets and resources */
static void dapm_free_widgets(struct snd_soc_codec *codec)
[ALSA] ASoC: dynamic audio power management (DAPM) This patch adds Dynamic Audio Power Management (DAPM) to ASoC. Dynamic Audio Power Management (DAPM) is designed to allow portable and handheld Linux devices to use the minimum amount of power within the audio subsystem at all times. It is independent of other kernel PM and as such, can easily co-exist with the other PM systems. DAPM is also completely transparent to all user space applications as all power switching is done within the ASoC core. No code changes or recompiling are required for user space applications. DAPM makes power switching decisions based upon any audio stream (capture/playback) activity and audio mixer settings within the device. DAPM spans the whole machine. It covers power control within the entire audio subsystem, this includes internal codec power blocks and machine level power systems. There are 4 power domains within DAPM:- 1. Codec domain - VREF, VMID (core codec and audio power) Usually controlled at codec probe/remove and suspend/resume, although can be set at stream time if power is not needed for sidetone, etc. 2. Platform/Machine domain - physically connected inputs and outputs Is platform/machine and user action specific, is configured by the machine driver and responds to asynchronous events e.g when HP are inserted 3. Path domain - audio subsystem signal paths Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. 4. Stream domain - DAC's and ADC's. Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. All DAPM power switching decisions are made automatically by consulting an audio routing map of the whole machine. This map is specific to each machine and consists of the interconnections between every audio component (including internal codec components). Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-07 00:32:18 +08:00
{
struct snd_soc_dapm_widget *w, *next_w;
struct snd_soc_dapm_path *p, *next_p;
list_for_each_entry_safe(w, next_w, &codec->dapm_widgets, list) {
list_del(&w->list);
kfree(w);
}
list_for_each_entry_safe(p, next_p, &codec->dapm_paths, list) {
list_del(&p->list);
kfree(p->long_name);
kfree(p);
}
}
/**
* snd_soc_dapm_sync_endpoints - scan and power dapm paths
* @codec: audio codec
*
* Walks all dapm audio paths and powers widgets according to their
* stream or path usage.
*
* Returns 0 for success.
*/
int snd_soc_dapm_sync_endpoints(struct snd_soc_codec *codec)
{
return dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_endpoints);
/**
* snd_soc_dapm_connect_input - connect dapm widgets
* @codec: audio codec
* @sink: name of target widget
* @control: mixer control name
* @source: name of source name
*
* Connects 2 dapm widgets together via a named audio path. The sink is
* the widget receiving the audio signal, whilst the source is the sender
* of the audio signal.
*
* Returns 0 for success else error.
*/
int snd_soc_dapm_connect_input(struct snd_soc_codec *codec, const char *sink,
const char * control, const char *source)
{
struct snd_soc_dapm_path *path;
struct snd_soc_dapm_widget *wsource = NULL, *wsink = NULL, *w;
int ret = 0;
/* find src and dest widgets */
list_for_each_entry(w, &codec->dapm_widgets, list) {
if (!wsink && !(strcmp(w->name, sink))) {
wsink = w;
continue;
}
if (!wsource && !(strcmp(w->name, source))) {
wsource = w;
}
}
if (wsource == NULL || wsink == NULL)
return -ENODEV;
path = kzalloc(sizeof(struct snd_soc_dapm_path), GFP_KERNEL);
if (!path)
return -ENOMEM;
path->source = wsource;
path->sink = wsink;
INIT_LIST_HEAD(&path->list);
INIT_LIST_HEAD(&path->list_source);
INIT_LIST_HEAD(&path->list_sink);
/* check for external widgets */
if (wsink->id == snd_soc_dapm_input) {
if (wsource->id == snd_soc_dapm_micbias ||
wsource->id == snd_soc_dapm_mic ||
wsink->id == snd_soc_dapm_line)
wsink->ext = 1;
}
if (wsource->id == snd_soc_dapm_output) {
if (wsink->id == snd_soc_dapm_spk ||
wsink->id == snd_soc_dapm_hp ||
wsink->id == snd_soc_dapm_line)
wsource->ext = 1;
}
/* connect static paths */
if (control == NULL) {
list_add(&path->list, &codec->dapm_paths);
list_add(&path->list_sink, &wsink->sources);
list_add(&path->list_source, &wsource->sinks);
path->connect = 1;
return 0;
}
/* connect dynamic paths */
switch(wsink->id) {
case snd_soc_dapm_adc:
case snd_soc_dapm_dac:
case snd_soc_dapm_pga:
case snd_soc_dapm_input:
case snd_soc_dapm_output:
case snd_soc_dapm_micbias:
case snd_soc_dapm_vmid:
case snd_soc_dapm_pre:
case snd_soc_dapm_post:
list_add(&path->list, &codec->dapm_paths);
list_add(&path->list_sink, &wsink->sources);
list_add(&path->list_source, &wsource->sinks);
path->connect = 1;
return 0;
case snd_soc_dapm_mux:
ret = dapm_connect_mux(codec, wsource, wsink, path, control,
&wsink->kcontrols[0]);
if (ret != 0)
goto err;
break;
case snd_soc_dapm_switch:
case snd_soc_dapm_mixer:
ret = dapm_connect_mixer(codec, wsource, wsink, path, control);
if (ret != 0)
goto err;
break;
case snd_soc_dapm_hp:
case snd_soc_dapm_mic:
case snd_soc_dapm_line:
case snd_soc_dapm_spk:
list_add(&path->list, &codec->dapm_paths);
list_add(&path->list_sink, &wsink->sources);
list_add(&path->list_source, &wsource->sinks);
path->connect = 0;
return 0;
}
return 0;
err:
printk(KERN_WARNING "asoc: no dapm match for %s --> %s --> %s\n", source,
control, sink);
kfree(path);
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_connect_input);
/**
* snd_soc_dapm_new_widgets - add new dapm widgets
* @codec: audio codec
*
* Checks the codec for any new dapm widgets and creates them if found.
*
* Returns 0 for success.
*/
int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
{
struct snd_soc_dapm_widget *w;
mutex_lock(&codec->mutex);
list_for_each_entry(w, &codec->dapm_widgets, list)
{
if (w->new)
continue;
switch(w->id) {
case snd_soc_dapm_switch:
case snd_soc_dapm_mixer:
dapm_new_mixer(codec, w);
break;
case snd_soc_dapm_mux:
dapm_new_mux(codec, w);
break;
case snd_soc_dapm_adc:
case snd_soc_dapm_dac:
case snd_soc_dapm_pga:
dapm_new_pga(codec, w);
break;
case snd_soc_dapm_input:
case snd_soc_dapm_output:
case snd_soc_dapm_micbias:
case snd_soc_dapm_spk:
case snd_soc_dapm_hp:
case snd_soc_dapm_mic:
case snd_soc_dapm_line:
case snd_soc_dapm_vmid:
case snd_soc_dapm_pre:
case snd_soc_dapm_post:
break;
}
w->new = 1;
}
dapm_power_widgets(codec, SND_SOC_DAPM_STREAM_NOP);
mutex_unlock(&codec->mutex);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_new_widgets);
/**
* snd_soc_dapm_get_volsw - dapm mixer get callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to get the value of a dapm mixer control.
*
* Returns 0 for success.
*/
int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
int reg = kcontrol->private_value & 0xff;
int shift = (kcontrol->private_value >> 8) & 0x0f;
int rshift = (kcontrol->private_value >> 12) & 0x0f;
int mask = (kcontrol->private_value >> 16) & 0xff;
int invert = (kcontrol->private_value >> 24) & 0x01;
/* return the saved value if we are powered down */
if (widget->id == snd_soc_dapm_pga && !widget->power) {
ucontrol->value.integer.value[0] = widget->saved_value;
return 0;
}
ucontrol->value.integer.value[0] =
(snd_soc_read(widget->codec, reg) >> shift) & mask;
if (shift != rshift)
ucontrol->value.integer.value[1] =
(snd_soc_read(widget->codec, reg) >> rshift) & mask;
if (invert) {
ucontrol->value.integer.value[0] =
mask - ucontrol->value.integer.value[0];
if (shift != rshift)
ucontrol->value.integer.value[1] =
mask - ucontrol->value.integer.value[1];
}
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_get_volsw);
/**
* snd_soc_dapm_put_volsw - dapm mixer set callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to set the value of a dapm mixer control.
*
* Returns 0 for success.
*/
int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
int reg = kcontrol->private_value & 0xff;
int shift = (kcontrol->private_value >> 8) & 0x0f;
int rshift = (kcontrol->private_value >> 12) & 0x0f;
int mask = (kcontrol->private_value >> 16) & 0xff;
int invert = (kcontrol->private_value >> 24) & 0x01;
unsigned short val, val2, val_mask;
int ret;
val = (ucontrol->value.integer.value[0] & mask);
if (invert)
val = mask - val;
val_mask = mask << shift;
val = val << shift;
if (shift != rshift) {
val2 = (ucontrol->value.integer.value[1] & mask);
if (invert)
val2 = mask - val2;
val_mask |= mask << rshift;
val |= val2 << rshift;
}
mutex_lock(&widget->codec->mutex);
widget->value = val;
/* save volume value if the widget is powered down */
if (widget->id == snd_soc_dapm_pga && !widget->power) {
widget->saved_value = val;
mutex_unlock(&widget->codec->mutex);
return 1;
}
dapm_mixer_update_power(widget, kcontrol, reg, val_mask, val, invert);
if (widget->event) {
if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
ret = widget->event(widget, SND_SOC_DAPM_PRE_REG);
if (ret < 0)
goto out;
}
ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
if (widget->event_flags & SND_SOC_DAPM_POST_REG)
ret = widget->event(widget, SND_SOC_DAPM_POST_REG);
} else
ret = snd_soc_update_bits(widget->codec, reg, val_mask, val);
out:
mutex_unlock(&widget->codec->mutex);
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw);
/**
* snd_soc_dapm_get_enum_double - dapm enumerated double mixer get callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to get the value of a dapm enumerated double mixer control.
*
* Returns 0 for success.
*/
int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
unsigned short val, bitmask;
for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
;
val = snd_soc_read(widget->codec, e->reg);
ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & (bitmask - 1);
if (e->shift_l != e->shift_r)
ucontrol->value.enumerated.item[1] =
(val >> e->shift_r) & (bitmask - 1);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double);
/**
* snd_soc_dapm_put_enum_double - dapm enumerated double mixer set callback
* @kcontrol: mixer control
* @uinfo: control element information
*
* Callback to set the value of a dapm enumerated double mixer control.
*
* Returns 0 for success.
*/
int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
unsigned short val, mux;
unsigned short mask, bitmask;
int ret = 0;
for (bitmask = 1; bitmask < e->mask; bitmask <<= 1)
;
if (ucontrol->value.enumerated.item[0] > e->mask - 1)
return -EINVAL;
mux = ucontrol->value.enumerated.item[0];
val = mux << e->shift_l;
mask = (bitmask - 1) << e->shift_l;
if (e->shift_l != e->shift_r) {
if (ucontrol->value.enumerated.item[1] > e->mask - 1)
return -EINVAL;
val |= ucontrol->value.enumerated.item[1] << e->shift_r;
mask |= (bitmask - 1) << e->shift_r;
}
mutex_lock(&widget->codec->mutex);
widget->value = val;
dapm_mux_update_power(widget, kcontrol, mask, mux, e);
if (widget->event) {
if (widget->event_flags & SND_SOC_DAPM_PRE_REG) {
ret = widget->event(widget, SND_SOC_DAPM_PRE_REG);
if (ret < 0)
goto out;
}
ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
if (widget->event_flags & SND_SOC_DAPM_POST_REG)
ret = widget->event(widget, SND_SOC_DAPM_POST_REG);
} else
ret = snd_soc_update_bits(widget->codec, e->reg, mask, val);
out:
mutex_unlock(&widget->codec->mutex);
return ret;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double);
/**
* snd_soc_dapm_new_control - create new dapm control
* @codec: audio codec
* @widget: widget template
*
* Creates a new dapm control based upon the template.
*
* Returns 0 for success else error.
*/
int snd_soc_dapm_new_control(struct snd_soc_codec *codec,
const struct snd_soc_dapm_widget *widget)
{
struct snd_soc_dapm_widget *w;
if ((w = dapm_cnew_widget(widget)) == NULL)
return -ENOMEM;
w->codec = codec;
INIT_LIST_HEAD(&w->sources);
INIT_LIST_HEAD(&w->sinks);
INIT_LIST_HEAD(&w->list);
list_add(&w->list, &codec->dapm_widgets);
/* machine layer set ups unconnected pins and insertions */
w->connected = 1;
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_new_control);
/**
* snd_soc_dapm_stream_event - send a stream event to the dapm core
* @codec: audio codec
* @stream: stream name
* @event: stream event
*
* Sends a stream event to the dapm core. The core then makes any
* necessary widget power changes.
*
* Returns 0 for success else error.
*/
int snd_soc_dapm_stream_event(struct snd_soc_codec *codec,
char *stream, int event)
{
struct snd_soc_dapm_widget *w;
if (stream == NULL)
return 0;
[ALSA] ASoC: dynamic audio power management (DAPM) This patch adds Dynamic Audio Power Management (DAPM) to ASoC. Dynamic Audio Power Management (DAPM) is designed to allow portable and handheld Linux devices to use the minimum amount of power within the audio subsystem at all times. It is independent of other kernel PM and as such, can easily co-exist with the other PM systems. DAPM is also completely transparent to all user space applications as all power switching is done within the ASoC core. No code changes or recompiling are required for user space applications. DAPM makes power switching decisions based upon any audio stream (capture/playback) activity and audio mixer settings within the device. DAPM spans the whole machine. It covers power control within the entire audio subsystem, this includes internal codec power blocks and machine level power systems. There are 4 power domains within DAPM:- 1. Codec domain - VREF, VMID (core codec and audio power) Usually controlled at codec probe/remove and suspend/resume, although can be set at stream time if power is not needed for sidetone, etc. 2. Platform/Machine domain - physically connected inputs and outputs Is platform/machine and user action specific, is configured by the machine driver and responds to asynchronous events e.g when HP are inserted 3. Path domain - audio subsystem signal paths Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. 4. Stream domain - DAC's and ADC's. Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. All DAPM power switching decisions are made automatically by consulting an audio routing map of the whole machine. This map is specific to each machine and consists of the interconnections between every audio component (including internal codec components). Signed-off-by: Richard Purdie <rpurdie@rpsys.net> Signed-off-by: Liam Girdwood <liam.girdwood@wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de> Signed-off-by: Jaroslav Kysela <perex@suse.cz>
2006-10-07 00:32:18 +08:00
mutex_lock(&codec->mutex);
list_for_each_entry(w, &codec->dapm_widgets, list)
{
if (!w->sname)
continue;
dbg("widget %s\n %s stream %s event %d\n", w->name, w->sname,
stream, event);
if (strstr(w->sname, stream)) {
switch(event) {
case SND_SOC_DAPM_STREAM_START:
w->active = 1;
break;
case SND_SOC_DAPM_STREAM_STOP:
w->active = 0;
break;
case SND_SOC_DAPM_STREAM_SUSPEND:
if (w->active)
w->suspend = 1;
w->active = 0;
break;
case SND_SOC_DAPM_STREAM_RESUME:
if (w->suspend) {
w->active = 1;
w->suspend = 0;
}
break;
case SND_SOC_DAPM_STREAM_PAUSE_PUSH:
break;
case SND_SOC_DAPM_STREAM_PAUSE_RELEASE:
break;
}
}
}
mutex_unlock(&codec->mutex);
dapm_power_widgets(codec, event);
dump_dapm(codec, __FUNCTION__);
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
/**
* snd_soc_dapm_set_endpoint - set audio endpoint status
* @codec: audio codec
* @endpoint: audio signal endpoint (or start point)
* @status: point status
*
* Set audio endpoint status - connected or disconnected.
*
* Returns 0 for success else error.
*/
int snd_soc_dapm_set_endpoint(struct snd_soc_codec *codec,
char *endpoint, int status)
{
struct snd_soc_dapm_widget *w;
list_for_each_entry(w, &codec->dapm_widgets, list) {
if (!strcmp(w->name, endpoint)) {
w->connected = status;
}
}
return 0;
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_set_endpoint);
/**
* snd_soc_dapm_free - free dapm resources
* @socdev: SoC device
*
* Free all dapm widgets and resources.
*/
void snd_soc_dapm_free(struct snd_soc_device *socdev)
{
struct snd_soc_codec *codec = socdev->codec;
snd_soc_dapm_sys_remove(socdev->dev);
dapm_free_widgets(codec);
}
EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
/* Module information */
MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC");
MODULE_LICENSE("GPL");