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Commit 3578be57 authored by Eduardo Valentin's avatar Eduardo Valentin Committed by Tony Lindgren
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Code clean-up for sound/arm/omap/omap-alsa-tsc2101-mixer.c 

Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.
Signed-off-by: default avatarEduardo Valentin <eduardo.valentin@indt.org.br>
Signed-off-by: default avatarTony Lindgren <tony@atomide.com>
parent 1a4ff9ad
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Showing with 342 additions and 282 deletions
sound/arm/omap/omap-alsa-tsc2101-mixer.c
View file @ 3578be57
/*
* sound/arm/omap/omap-alsa-tsc2101-mixer.c
*
*
* Alsa Driver for TSC2101 codec for OMAP platform boards.
*
* Copyright (C) 2005 Mika Laitio <lamikr@cc.jyu.fi> and
* Copyright (C) 2005 Mika Laitio <lamikr@cc.jyu.fi> and
* Everett Coleman II <gcc80x86@fuzzyneural.net>
*
* Board initialization code is based on the code in TSC2101 OSS driver.
* Copyright (C) 2004 Texas Instruments, Inc.
* Written by Nishanth Menon and Sriram Kannan
*
*
* 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
......@@ -33,13 +33,13 @@
* History:
*
* 2006-03-01 Mika Laitio - Mixer for the tsc2101 driver used in omap boards.
* Can switch between headset and loudspeaker playback,
* Can switch between headset and loudspeaker playback,
* mute and unmute dgc, set dgc volume. Record source switch,
* keyclick, buzzer and headset volume and handset volume control
* keyclick, buzzer and headset volume and handset volume control
* are still missing.
*
*
*/
#include "omap-alsa-tsc2101.h"
#include "omap-alsa-tsc2101-mixer.h"
......@@ -48,8 +48,15 @@
#include <sound/initval.h>
#include <sound/control.h>
//#define M_DPRINTK(ARGS...) printk(KERN_INFO "<%s>: ",__FUNCTION__);printk(ARGS)
#ifdef DEBUG
#define M_DPRINTK(ARGS...) \
do { \
printk(KERN_INFO "<%s>: ", __func__); \
printk(ARGS); \
} while (0)
#else
#define M_DPRINTK(ARGS...) /* nop */
#endif
#define CHECK_BIT(INDX, ARG) (((ARG) & TSC2101_BIT(INDX)) >> INDX)
#define IS_UNMUTED(INDX, ARG) (((CHECK_BIT(INDX, ARG)) == 0))
......@@ -64,7 +71,7 @@
static int current_playback_target = PLAYBACK_TARGET_LOUDSPEAKER;
static int current_rec_src = REC_SRC_SINGLE_ENDED_MICIN_HED;
/*
/*
* Simplified write for the tsc2101 audio registers.
*/
inline void omap_tsc2101_audio_write(u8 address, u16 data)
......@@ -73,7 +80,7 @@ inline void omap_tsc2101_audio_write(u8 address, u16 data)
address, data);
}
/*
/*
* Simplified read for the tsc2101 audio registers.
*/
inline u16 omap_tsc2101_audio_read(u8 address)
......@@ -88,8 +95,9 @@ inline u16 omap_tsc2101_audio_read(u8 address)
static void set_record_source(int val)
{
u16 data;
/* Mute Analog Sidetone
/*
* Mute Analog Sidetone
* Analog sidetone gain db?
* Input selected by MICSEL connected to ADC
*/
......@@ -98,77 +106,84 @@ static void set_record_source(int val)
data |= MPC_MICSEL(val);
data |= MPC_MICADC;
omap_tsc2101_audio_write(TSC2101_MIXER_PGA_CTRL, data);
current_rec_src = val;
}
/*
* Converts the Alsa mixer volume (0 - 100) to real
* Converts the Alsa mixer volume (0 - 100) to real
* Digital Gain Control (DGC) value that can be written
* or read from the TSC2101 registry.
*
*
* Note that the number "OUTPUT_VOLUME_MAX" is smaller than OUTPUT_VOLUME_MIN
* because DGC works as a volume decreaser. (The more bigger value is put
* to DGC, the more the volume of controlled channel is decreased)
*
* In addition the TCS2101 chip would allow the maximum volume reduction be 63.5 DB
*
* In addition the TCS2101 chip would allow the maximum
* volume reduction be 63.5 DB
* but according to some tests user can not hear anything with this chip
* when the volume is set to be less than 25 db.
* Therefore this function will return a value that means 38.5 db (63.5 db - 25 db)
* Therefore this function will return a value
* that means 38.5 db (63.5 db - 25 db)
* reduction in the channel volume, when mixer is set to 0.
* For mixer value 100, this will return a value that means 0 db volume reduction.
* For mixer value 100, this will return a value that means
* 0 db volume reduction.
* ([mute_left_bit]0000000[mute_right_bit]0000000)
*/
*/
int get_mixer_volume_as_dac_gain_control_volume(int vol)
{
u16 retVal;
/* Convert 0 -> 100 volume to 0x7F(min) -> y(max) volume range */
retVal = ((vol * OUTPUT_VOLUME_RANGE) / 100) + OUTPUT_VOLUME_MAX;
retVal = ((vol * OUTPUT_VOLUME_RANGE) / 100) + OUTPUT_VOLUME_MAX;
/* invert the value for getting the proper range 0 min and 100 max */
retVal = OUTPUT_VOLUME_MIN - retVal;
retVal = OUTPUT_VOLUME_MIN - retVal;
return retVal;
}
/*
* Converts the Alsa mixer volume (0 - 100) to TSC2101
* Converts the Alsa mixer volume (0 - 100) to TSC2101
* Digital Gain Control (DGC) volume. Alsa mixer volume 0
* is converted to value meaning the volume reduction of -38.5 db
* and Alsa mixer volume 100 is converted to value meaning the
* reduction of 0 db.
*/
int set_mixer_volume_as_dac_gain_control_volume(int mixerVolL, int mixerVolR)
int set_mixer_volume_as_dac_gain_control_volume(int mixerVolL, int mixerVolR)
{
u16 val;
int retVal;
int volL;
int volR;
if ((mixerVolL < 0) ||
if ((mixerVolL < 0) ||
(mixerVolL > 100) ||
(mixerVolR < 0) ||
(mixerVolR > 100)) {
printk(KERN_ERR "Trying a bad mixer volume as dac gain control volume value, left (%d), right (%d)!\n", mixerVolL, mixerVolR);
printk(KERN_ERR "Trying a bad mixer volume as dac gain control"
" volume value, left (%d), right (%d)!\n", mixerVolL,
mixerVolR);
return -EPERM;
}
M_DPRINTK("mixer volume left = %d, right = %d\n", mixerVolL, mixerVolR);
M_DPRINTK("mixer volume left = %d, right = %d\n", mixerVolL, mixerVolR);
volL = get_mixer_volume_as_dac_gain_control_volume(mixerVolL);
volR = get_mixer_volume_as_dac_gain_control_volume(mixerVolR);
val = omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
/* keep the old mute bit settings */
val &= ~(DGC_DALVL(OUTPUT_VOLUME_MIN) | DGC_DARVL(OUTPUT_VOLUME_MIN));
val &= ~(DGC_DALVL(OUTPUT_VOLUME_MIN) |
DGC_DARVL(OUTPUT_VOLUME_MIN));
val |= DGC_DALVL(volL) | DGC_DARVL(volR);
retVal = 2;
if (retVal) {
if (retVal)
omap_tsc2101_audio_write(TSC2101_DAC_GAIN_CTRL, val);
}
M_DPRINTK("to registry: left = %d, right = %d, total = %d\n", DGC_DALVL_EXTRACT(val), DGC_DARVL_EXTRACT(val), val);
M_DPRINTK("to registry: left = %d, right = %d, total = %d\n",
DGC_DALVL_EXTRACT(val), DGC_DARVL_EXTRACT(val), val);
return retVal;
}
/**
/*
* If unmuteLeft/unmuteRight == 0 --> mute
* If unmuteLeft/unmuteRight == 1 --> unmute
*/
......@@ -179,15 +194,16 @@ int dac_gain_control_unmute(int unmuteLeft, int unmuteRight)
count = 0;
val = omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
/* in alsa mixer 1 --> on, 0 == off. In tsc2101 registry 1 --> off, 0 --> on
* so if values are same, it's time to change the registry value.
/*
* in alsa mixer 1 --> on, 0 == off. In tsc2101 registry 1 --> off,
* 0 --> on so if values are same, it's time to change the registry
* value.
*/
if (unmuteLeft != IS_UNMUTED(15, val)) {
if (unmuteLeft == 0) {
/* mute --> turn bit on */
val = val | DGC_DALMU;
}
else {
} else {
/* unmute --> turn bit off */
val = val & ~DGC_DALMU;
}
......@@ -197,69 +213,71 @@ int dac_gain_control_unmute(int unmuteLeft, int unmuteRight)
if (unmuteRight == 0) {
/* mute --> turn bit on */
val = val | DGC_DARMU;
}
else {
} else {
/* unmute --> turn bit off */
val = val & ~DGC_DARMU;
}
}
count++;
} /* R */
if (count) {
omap_tsc2101_audio_write(TSC2101_DAC_GAIN_CTRL, val);
M_DPRINTK("changed value, is_unmuted left = %d, right = %d\n",
M_DPRINTK("changed value, is_unmuted left = %d, right = %d\n",
IS_UNMUTED(15, val),
IS_UNMUTED(7, val));
}
return count;
return count;
}
/**
/*
* unmute: 0 --> mute, 1 --> unmute
* page2RegIndx: Registry index in tsc2101 page2.
* muteBitIndx: Index number for the bit in registry that indicates whether muted or unmuted.
* muteBitIndx: Index number for the bit in registry that indicates whether
* muted or unmuted.
*/
int adc_pga_unmute_control(int unmute, int page2regIndx, int muteBitIndx)
{
int count;
u16 val;
count = 0;
val = omap_tsc2101_audio_read(page2regIndx);
/* in alsa mixer 1 --> on, 0 == off. In tsc2101 registry 1 --> off, 0 --> on
* so if the values are same, it's time to change the registry value...
/*
* in alsa mixer 1 --> on, 0 == off. In tsc2101 registry 1 --> off,
* 0 --> on so if the values are same, it's time to change the
* registry value...
*/
if (unmute != IS_UNMUTED(muteBitIndx, val)) {
if (unmute == 0) {
/* mute --> turn bit on */
val = val | TSC2101_BIT(muteBitIndx);
}
else {
} else {
/* unmute --> turn bit off */
val = val & ~TSC2101_BIT(muteBitIndx);
}
M_DPRINTK("changed value, is_unmuted = %d\n", IS_UNMUTED(muteBitIndx, val));
M_DPRINTK("changed value, is_unmuted = %d\n",
IS_UNMUTED(muteBitIndx, val));
count++;
}
if (count) {
if (count)
omap_tsc2101_audio_write(page2regIndx, val);
}
return count;
}
/*
* Converts the DGC registry value read from the TSC2101 registry to
* Converts the DGC registry value read from the TSC2101 registry to
* Alsa mixer volume format (0 - 100).
*/
int get_dac_gain_control_volume_as_mixer_volume(u16 vol)
int get_dac_gain_control_volume_as_mixer_volume(u16 vol)
{
u16 retVal;
u16 retVal;
retVal = OUTPUT_VOLUME_MIN - vol;
retVal = ((retVal - OUTPUT_VOLUME_MAX) * 100) / OUTPUT_VOLUME_RANGE;
/* fix scaling error */
if ((retVal > 0) && (retVal < 100)) {
if ((retVal > 0) && (retVal < 100))
retVal++;
}
return retVal;
}
......@@ -267,10 +285,10 @@ int get_dac_gain_control_volume_as_mixer_volume(u16 vol)
* Converts the headset gain control volume (0 - 63.5 db)
* to Alsa mixer volume (0 - 100)
*/
int get_headset_gain_control_volume_as_mixer_volume(u16 registerVal)
int get_headset_gain_control_volume_as_mixer_volume(u16 registerVal)
{
u16 retVal;
retVal = ((registerVal * 100) / INPUT_VOLUME_RANGE);
return retVal;
}
......@@ -279,71 +297,78 @@ int get_headset_gain_control_volume_as_mixer_volume(u16 registerVal)
* Converts the handset gain control volume (0 - 63.5 db)
* to Alsa mixer volume (0 - 100)
*/
int get_handset_gain_control_volume_as_mixer_volume(u16 registerVal)
int get_handset_gain_control_volume_as_mixer_volume(u16 registerVal)
{
return get_headset_gain_control_volume_as_mixer_volume(registerVal);
}
/*
* Converts the Alsa mixer volume (0 - 100) to
* Converts the Alsa mixer volume (0 - 100) to
* headset gain control volume (0 - 63.5 db)
*/
int get_mixer_volume_as_headset_gain_control_volume(u16 mixerVal)
int get_mixer_volume_as_headset_gain_control_volume(u16 mixerVal)
{
u16 retVal;
retVal = ((mixerVal * INPUT_VOLUME_RANGE) / 100) + INPUT_VOLUME_MIN;
retVal = ((mixerVal * INPUT_VOLUME_RANGE) / 100) + INPUT_VOLUME_MIN;
return retVal;
}
/*
* Writes Alsa mixer volume (0 - 100) to TSC2101 headset volume registry in
* a TSC2101 format. (0 - 63.5 db)
* In TSC2101 OSS driver this functionality was controlled with "SET_LINE" parameter.
* In TSC2101 OSS driver this functionality was controlled with "SET_LINE"
* parameter.
*/
int set_mixer_volume_as_headset_gain_control_volume(int mixerVol)
int set_mixer_volume_as_headset_gain_control_volume(int mixerVol)
{
int volume;
int retVal;
u16 val;
if (mixerVol < 0 || mixerVol > 100) {
M_DPRINTK("Trying a bad headset mixer volume value(%d)!\n", mixerVol);
M_DPRINTK("Trying a bad headset mixer volume value(%d)!\n",
mixerVol);
return -EPERM;
}
M_DPRINTK("mixer volume = %d\n", mixerVol);
/* Convert 0 -> 100 volume to 0x0(min) -> 0x7D(max) volume range */
/* NOTE: 0 is minimum volume and not mute */
volume = get_mixer_volume_as_headset_gain_control_volume(mixerVol);
/*
* Convert 0 -> 100 volume to 0x0(min) -> 0x7D(max) volume range
* NOTE: 0 is minimum volume and not mute
*/
volume = get_mixer_volume_as_headset_gain_control_volume(mixerVol);
val = omap_tsc2101_audio_read(TSC2101_HEADSET_GAIN_CTRL);
/* preserve the old mute settings */
val &= ~(HGC_ADPGA_HED(INPUT_VOLUME_MAX));
val |= HGC_ADPGA_HED(volume);
omap_tsc2101_audio_write(TSC2101_HEADSET_GAIN_CTRL, val);
omap_tsc2101_audio_write(TSC2101_HEADSET_GAIN_CTRL, val);
retVal = 1;
M_DPRINTK("to registry = %d\n", val);
M_DPRINTK("to registry = %d\n", val);
return retVal;
}
/*
* Writes Alsa mixer volume (0 - 100) to TSC2101 handset volume registry in
* a TSC2101 format. (0 - 63.5 db)
* In TSC2101 OSS driver this functionality was controlled with "SET_MIC" parameter.
* In TSC2101 OSS driver this functionality was controlled with
* "SET_MIC" parameter.
*/
int set_mixer_volume_as_handset_gain_control_volume(int mixerVol)
int set_mixer_volume_as_handset_gain_control_volume(int mixerVol)
{
int volume;
int retVal;
u16 val;
u16 val;
if (mixerVol < 0 || mixerVol > 100) {
M_DPRINTK("Trying a bad mic mixer volume value(%d)!\n", mixerVol);
M_DPRINTK("Trying a bad mic mixer volume value(%d)!\n",
mixerVol);
return -EPERM;
}
M_DPRINTK("mixer volume = %d\n", mixerVol);
/* Convert 0 -> 100 volume to 0x0(min) -> 0x7D(max) volume range
* NOTE: 0 is minimum volume and not mute
/*
* Convert 0 -> 100 volume to 0x0(min) -> 0x7D(max) volume range
* NOTE: 0 is minimum volume and not mute
*/
volume = get_mixer_volume_as_headset_gain_control_volume(mixerVol);
val = omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
......@@ -352,8 +377,8 @@ int set_mixer_volume_as_handset_gain_control_volume(int mixerVol)
val |= HNGC_ADPGA_HND(volume);
omap_tsc2101_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);
retVal = 1;
M_DPRINTK("to registry = %d\n", val);
M_DPRINTK("to registry = %d\n", val);
return retVal;
}
......@@ -361,27 +386,31 @@ void set_loudspeaker_to_playback_target(void)
{
/* power down SPK1, SPK2 and loudspeaker */
omap_tsc2101_audio_write(TSC2101_CODEC_POWER_CTRL,
CPC_SP1PWDN | CPC_SP2PWDN | CPC_LDAPWDF);
/* ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled
CPC_SP1PWDN | CPC_SP2PWDN | CPC_LDAPWDF);
/*
* ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled
* 1dB AGC hysteresis
* MICes bias 2V
*/
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_4, AC4_MB_HED(0));
/* DAC left and right routed to SPK1/SPK2
/*
* DAC left and right routed to SPK1/SPK2
* SPK1/SPK2 unmuted
* Keyclicks routed to SPK1/SPK2 */
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_5,
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_5,
AC5_DIFFIN |
AC5_DAC2SPK1(3) | AC5_AST2SPK1 | AC5_KCL2SPK1 |
AC5_DAC2SPK2(3) | AC5_AST2SPK2 | AC5_KCL2SPK2);
/* routing selected to SPK1 goes also to OUT8P/OUT8N. (loudspeaker)
/*
* routing selected to SPK1 goes also to OUT8P/OUT8N. (loudspeaker)
* analog sidetone routed to loudspeaker
* buzzer pga routed to loudspeaker
* keyclick routing to loudspeaker
* cellphone input routed to loudspeaker
* mic selection (control register 04h/page2) routed to cell phone output (CP_OUT)
* mic selection (control register 04h/page2) routed to cell phone
* output (CP_OUT)
* routing selected for SPK1 goes also to cellphone output (CP_OUT)
* OUT8P/OUT8N (loudspeakers) unmuted (0 = unmuted)
* Cellphone output is not muted (0 = unmuted)
......@@ -399,19 +428,23 @@ void set_headphone_to_playback_target(void)
/* power down SPK1, SPK2 and loudspeaker */
omap_tsc2101_audio_write(TSC2101_CODEC_POWER_CTRL,
CPC_SP1PWDN | CPC_SP2PWDN | CPC_LDAPWDF);
/* ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled */
/* 1dB AGC hysteresis */
/* MICes bias 2V */
/*
* ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled
 * 1dB AGC hysteresis
* MICes bias 2V
*/
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_4, AC4_MB_HED(0));
/* DAC left and right routed to SPK1/SPK2
/*
* DAC left and right routed to SPK1/SPK2
* SPK1/SPK2 unmuted
* Keyclicks routed to SPK1/SPK2 */
* Keyclicks routed to SPK1/SPK2
*/
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_5,
AC5_DAC2SPK1(3) | AC5_AST2SPK1 | AC5_KCL2SPK1 |
AC5_DAC2SPK2(3) | AC5_AST2SPK2 | AC5_KCL2SPK2 |
AC5_HDSCPTC);
/* OUT8P/OUT8N muted, CPOUT muted */
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_6,
AC6_MUTLSPK | AC6_MUTSPK2 | AC6_LDSCPTC |
......@@ -421,45 +454,47 @@ void set_headphone_to_playback_target(void)
void set_telephone_to_playback_target(void)
{
/*
/*
* 0110 1101 0101 1100
* power down MICBIAS_HED, Analog sidetone, SPK2, DAC,
* power down MICBIAS_HED, Analog sidetone, SPK2, DAC,
* Driver virtual ground, loudspeaker. Values D2-d5 are flags.
*/
*/
omap_tsc2101_audio_write(TSC2101_CODEC_POWER_CTRL,
CPC_MBIAS_HED | CPC_ASTPWD | CPC_SP2PWDN | CPC_DAPWDN |
CPC_VGPWDN | CPC_LSPWDN);
/*
/*
* 0010 1010 0100 0000
* ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled
* 1dB AGC hysteresis
* MICes bias 2V
*/
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_4,
AC4_MB_HND | AC4_MB_HED(0) | AC4_AGCHYS(1) |
AC4_MB_HND | AC4_MB_HED(0) | AC4_AGCHYS(1) |
AC4_BISTPD | AC4_ASSTPD | AC4_DASTPD);
printk("set_telephone_to_playback_target(), TSC2101_AUDIO_CTRL_4 = %d\n", omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_4));
/*
printk(KERN_INFO "set_telephone_to_playback_target(), "
"TSC2101_AUDIO_CTRL_4 = %d\n",
omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_4));
/*
* 1110 0010 0000 0010
* DAC left and right routed to SPK1/SPK2
* SPK1/SPK2 unmuted
* keyclicks routed to SPK1/SPK2
*/
*/
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_5,
AC5_DIFFIN | AC5_DAC2SPK1(3) |
AC5_CPI2SPK1 | AC5_MUTSPK2);
AC5_DIFFIN | AC5_DAC2SPK1(3) |
AC5_CPI2SPK1 | AC5_MUTSPK2);
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_6,
AC6_MIC2CPO | AC6_MUTLSPK |
AC6_MIC2CPO | AC6_MUTLSPK |
AC6_LDSCPTC | AC6_VGNDSCPTC | AC6_CAPINTF);
current_playback_target = PLAYBACK_TARGET_CELLPHONE;
}
/*
* 1100 0101 1101 0000
*
*
* #define MPC_ASTMU TSC2101_BIT(15)
* #define MPC_ASTG(ARG) (((ARG) & 0x7F) << 8)
* #define MPC_MICSEL(ARG) (((ARG) & 0x07) << 5)
......@@ -470,14 +505,14 @@ void set_telephone_to_playback_target(void)
static void set_telephone_to_record_source(void)
{
u16 val;
/*
* D0 = 0:
/*
* D0 = 0:
* --> AGC is off for handset input.
* --> ADC PGA is controlled by the ADMUT_HDN + ADPGA_HND
* (D15, D14-D8)
* D4 - D1 = 0000
* --> AGC time constant for handset input,
* D4 - D1 = 0000
* --> AGC time constant for handset input,
* attack time = 8 mc, decay time = 100 ms
* D7 - D5 = 000
* --> AGC Target gain for handset input = -5.5 db
......@@ -486,33 +521,36 @@ static void set_telephone_to_record_source(void)
* D15 = 0
* --> Handset input ON (unmuted)
*/
val = 0x3c00; // 0011 1100 0000 0000 = 60 = 30
val = 0x3c00; /* 0011 1100 0000 0000 = 60 = 30 */
omap_tsc2101_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);
/*
* D0 = 0
* --> AGC is off for headset/Aux input
* --> ADC headset/Aux PGA is contoller by ADMUT_HED + ADPGA_HED
* --> ADC headset/Aux PGA is contoller by
* ADMUT_HED + ADPGA_HED
* (D15, D14-D8)
* D4 - D1 = 0000
* D4 - D1 = 0000
* --> Agc constant for headset/Aux input,
* attack time = 8 mc, decay time = 100 ms
* attack time = 8 mc, decay time = 100 ms
* D7 - D5 = 000
* --> AGC target gain for headset input = -5.5 db
* D14 - D8 = 000 0000
* --> Adc headset/AUX pga settings = 0 db
* D15 = 1
* --> Headset/AUX input muted
*
*
* Mute headset aux input
*/
val = 0x8000; // 1000 0000 0000 0000
val = 0x8000; /* 1000 0000 0000 0000 */
omap_tsc2101_audio_write(TSC2101_HEADSET_GAIN_CTRL, val);
set_record_source(REC_SRC_MICIN_HND_AND_AUX1);
// hacks start
/* D0 = flag, Headset/Aux or handset PGA flag
* --> & with 1 (= 1 -->gain applied == pga register settings)
/*
* hacks start
* D0 = flag, Headset/Aux or handset PGA flag
* --> & with 1 (= 1 -->gain applied == pga
* register settings)
* D1 = 0, DAC channel PGA soft stepping control
* --> 0.5 db change every WCLK
* D2 = flag, DAC right channel PGA flag
......@@ -521,8 +559,8 @@ static void set_telephone_to_record_source(void)
* -- > & with 1
* D7 - D4 = 0001, keyclick length
* --> 4 periods key clicks
* D10 - D8 = 100, keyclick frequenzy
* --> 1 kHz,
* D10 - D8 = 100, keyclick frequency
* --> 1 kHz,
* D11 = 0, Headset/Aux or handset soft stepping control
* --> 0,5 db change every WCLK or ADWS
* D14 -D12 = 100, Keyclick applitude control
......@@ -531,7 +569,7 @@ static void set_telephone_to_record_source(void)
*/
val = omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_2);
val = val & 0x441d;
val = val | 0x4410; // D14, D10, D4 bits == 1
val = val | 0x4410; /* D14, D10, D4 bits == 1 */
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_2, val);
/*
......@@ -543,24 +581,28 @@ static void set_telephone_to_record_source(void)
* --> MICBIAS_HND = 2.0 v
* D8 - D7 = 00
* --> MICBIAS_HED = 3.3 v
* D10 - D9 = 01,
* D10 - D9 = 01,
* --> Mic AGC hysteric selection = 2 db
* D11 = 1,
* D11 = 1,
* --> Disable buzzer PGA soft stepping
* D12 = 0,
* --> Enable CELL phone PGA soft stepping control
* D13 = 1
* --> Disable analog sidetone soft stepping control
* --> Disable analog sidetone soft
* stepping control
* D14 = 0
* --> Enable DAC PGA soft stepping control
* D15 = 0,
* --> Enable headset/Aux or Handset soft stepping control
* --> Enable headset/Aux or Handset soft
* stepping control
*/
val = omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_4);
val = val & 0x2a42; // 0010 1010 0100 0010
val = val | 0x2a40; // bits D13, D11, D9, D6 == 1
val = val & 0x2a42; /* 0010 1010 0100 0010 */
val = val | 0x2a40; /* bits D13, D11, D9, D6 == 1 */
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_4, val);
printk("set_telephone_to_record_source(), TSC2101_AUDIO_CTRL_4 = %d\n", omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_4));
printk(KERN_INFO "set_telephone_to_record_source(), "
"TSC2101_AUDIO_CTRL_4 = %d\n",
omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_4));
/*
* D0 = 0
* --> reserved, write always = 0
......@@ -579,10 +621,12 @@ static void set_telephone_to_record_source(void)
*/
val = omap_tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL);
val = val & 0x5dfe;
val = val | 0x5dfe; // bits, D14, D12, D11, D10, D8, D6, D5,D4,D3,D2
/* bits, D14, D12, D11, D10, D8, D6, D5,D4,D3,D2 */
val = val | 0x5dfe;
omap_tsc2101_audio_write(TSC2101_BUZZER_GAIN_CTRL, val);
/* D6 - D0 = 000 1001
/*
* D6 - D0 = 000 1001
* --> -4.5 db for DAC right channel volume control
* D7 = 1
* --> DAC right channel muted
......@@ -591,12 +635,13 @@ static void set_telephone_to_record_source(void)
* D15 = 1
* --> DAC left channel muted
*/
//val = omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
/* val = omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL); */
val = 0x8989;
omap_tsc2101_audio_write(TSC2101_DAC_GAIN_CTRL, val);
/* 0000 0000 0100 0000
*
omap_tsc2101_audio_write(TSC2101_DAC_GAIN_CTRL, val);
/*
* 0000 0000 0100 0000
*
* D1 - D0 = 0
* --> GPIO 1 pin output is three stated
* D2 = 0
......@@ -610,18 +655,18 @@ static void set_telephone_to_record_source(void)
* --> 8 ms clitch detection
* D8 = reserved, write only 0
* D10 -D9 = 00
* --> 16 ms de bouncing programmatitily
* --> 16 ms de-bouncing
* for glitch detection during headset detection
* D11 = flag for button press
* D12 = flag for headset detection
* D14-D13 = 00
* --> type of headset detected = 00 == no stereo headset deected
* --> type of headset detected = 00 == no stereo
* headset deected
* D15 = 0
* --> Disable headset detection
*
* */
*/
val = 0x40;
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_7, val);
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_7, val);
}
/*
......@@ -637,17 +682,17 @@ u16 get_headset_detected(void)
u16 curDetected;
u16 curType;
u16 curVal;
curType = 0; /* not detected */
curVal = omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_7);
curDetected = curVal & AC7_HDDETFL;
if (curDetected) {
printk("headset detected, checking type from %d \n", curVal);
printk(KERN_INFO "headset detected, checking type from %d \n",
curVal);
curType = ((curVal & 0x6000) >> 13);
printk("headset type detected = %d \n", curType);
}
else {
printk("headset not detected\n");
printk(KERN_INFO "headset type detected = %d \n", curType);
} else {
printk(KERN_INFO "headset not detected\n");
}
return curType;
}
......@@ -657,40 +702,46 @@ void init_playback_targets(void)
u16 val;
set_loudspeaker_to_playback_target();
/* Left line input volume control
/*
* Left line input volume control
* = SET_LINE in the OSS driver
*/
set_mixer_volume_as_headset_gain_control_volume(DEFAULT_INPUT_VOLUME);
/* Set headset to be controllable by handset mixer
/*
* Set headset to be controllable by handset mixer
* AGC enable for handset input
* Handset input not muted
*/
val = omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
val = val | HNGC_AGCEN_HND;
val = val | HNGC_AGCEN_HND;
val = val & ~HNGC_ADMUT_HND;
omap_tsc2101_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);
/* mic input volume control
* SET_MIC in the OSS driver
omap_tsc2101_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);
/*
* mic input volume control
* SET_MIC in the OSS driver
*/
set_mixer_volume_as_handset_gain_control_volume(DEFAULT_INPUT_VOLUME);
/* Left/Right headphone channel volume control
/*
* Left/Right headphone channel volume control
* Zero-cross detect on
*/
set_mixer_volume_as_dac_gain_control_volume(DEFAULT_OUTPUT_VOLUME, DEFAULT_OUTPUT_VOLUME);
set_mixer_volume_as_dac_gain_control_volume(DEFAULT_OUTPUT_VOLUME,
DEFAULT_OUTPUT_VOLUME);
/* unmute */
dac_gain_control_unmute(1, 1);
}
/*
* Initializes tsc2101 recourd source (to line) and playback target (to loudspeaker)
* Initializes tsc2101 recourd source (to line) and playback target
* (to loudspeaker)
*/
void snd_omap_init_mixer(void)
{
{
FN_IN;
/* Headset/Hook switch detect enabled */
omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_7, AC7_DETECT);
......@@ -706,17 +757,17 @@ static int __pcm_playback_target_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[PLAYBACK_TARGET_COUNT] = {
"Loudspeaker", "Headphone", "Cellphone"
"Loudspeaker", "Headphone", "Cellphone"
};
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = PLAYBACK_TARGET_COUNT;
if (uinfo->value.enumerated.item > PLAYBACK_TARGET_COUNT - 1) {
uinfo->value.enumerated.item = PLAYBACK_TARGET_COUNT - 1;
}
if (uinfo->value.enumerated.item > PLAYBACK_TARGET_COUNT - 1)
uinfo->value.enumerated.item = PLAYBACK_TARGET_COUNT - 1;
strcpy(uinfo->value.enumerated.name,
texts[uinfo->value.enumerated.item]);
texts[uinfo->value.enumerated.item]);
return 0;
}
......@@ -732,28 +783,26 @@ static int __pcm_playback_target_put(struct snd_kcontrol *kcontrol,
{
int retVal;
int curVal;
retVal = 0;
curVal = ucontrol->value.integer.value[0];
if ((curVal >= 0) &&
(curVal < PLAYBACK_TARGET_COUNT) &&
(curVal != current_playback_target)) {
(curVal != current_playback_target)) {
if (curVal == PLAYBACK_TARGET_LOUDSPEAKER) {
set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HED);
set_loudspeaker_to_playback_target();
}
else if (curVal == PLAYBACK_TARGET_HEADPHONE) {
} else if (curVal == PLAYBACK_TARGET_HEADPHONE) {
set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HND);
set_headphone_to_playback_target();
}
else if (curVal == PLAYBACK_TARGET_CELLPHONE) {
} else if (curVal == PLAYBACK_TARGET_CELLPHONE) {
set_telephone_to_record_source();
set_telephone_to_playback_target();
}
retVal = 1;
}
return retVal;
}
}
static int __pcm_playback_volume_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
......@@ -766,16 +815,16 @@ static int __pcm_playback_volume_info(struct snd_kcontrol *kcontrol,
}
/*
* Alsa mixer interface function for getting the volume read from the DGC in a
* Alsa mixer interface function for getting the volume read from the DGC in a
* 0 -100 alsa mixer format.
*/
static int __pcm_playback_volume_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
u16 volL;
u16 volR;
u16 volR;
u16 val;
val = omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
M_DPRINTK("registry value = %d!\n", val);
volL = DGC_DALVL_EXTRACT(val);
......@@ -786,19 +835,22 @@ static int __pcm_playback_volume_get(struct snd_kcontrol *kcontrol,
volL = get_dac_gain_control_volume_as_mixer_volume(volL);
volR = get_dac_gain_control_volume_as_mixer_volume(volR);
ucontrol->value.integer.value[0] = volL; /* L */
ucontrol->value.integer.value[1] = volR; /* R */
M_DPRINTK("mixer volume left = %ld, right = %ld\n", ucontrol->value.integer.value[0], ucontrol->value.integer.value[1]);
M_DPRINTK("mixer volume left = %ld, right = %ld\n",
ucontrol->value.integer.value[0],
ucontrol->value.integer.value[1]);
return 0;
}
static int __pcm_playback_volume_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return set_mixer_volume_as_dac_gain_control_volume(ucontrol->value.integer.value[0],
ucontrol->value.integer.value[1]);
return set_mixer_volume_as_dac_gain_control_volume(
ucontrol->value.integer.value[0],
ucontrol->value.integer.value[1]);
}
static int __pcm_playback_switch_info(struct snd_kcontrol *kcontrol,
......@@ -811,7 +863,7 @@ static int __pcm_playback_switch_info(struct snd_kcontrol *kcontrol,
return 0;
}
/*
/*
* When DGC_DALMU (bit 15) is 1, the left channel is muted.
* When DGC_DALMU is 0, left channel is not muted.
* Same logic apply also for the right channel.
......@@ -820,16 +872,16 @@ static int __pcm_playback_switch_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
u16 val = omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
ucontrol->value.integer.value[0] = IS_UNMUTED(15, val); // left
ucontrol->value.integer.value[1] = IS_UNMUTED(7, val); // right
ucontrol->value.integer.value[0] = IS_UNMUTED(15, val); /* left */
ucontrol->value.integer.value[1] = IS_UNMUTED(7, val); /* right */
return 0;
}
static int __pcm_playback_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return dac_gain_control_unmute(ucontrol->value.integer.value[0],
return dac_gain_control_unmute(ucontrol->value.integer.value[0],
ucontrol->value.integer.value[1]);
}
......@@ -848,7 +900,7 @@ static int __headset_playback_volume_get(struct snd_kcontrol *kcontrol,
{
u16 val;
u16 vol;
val = omap_tsc2101_audio_read(TSC2101_HEADSET_GAIN_CTRL);
M_DPRINTK("registry value = %d\n", val);
vol = HGC_ADPGA_HED_EXTRACT(val);
......@@ -856,15 +908,17 @@ static int __headset_playback_volume_get(struct snd_kcontrol *kcontrol,
vol = get_headset_gain_control_volume_as_mixer_volume(vol);
ucontrol->value.integer.value[0] = vol;
M_DPRINTK("mixer volume returned = %ld\n", ucontrol->value.integer.value[0]);
M_DPRINTK("mixer volume returned = %ld\n",
ucontrol->value.integer.value[0]);
return 0;
}
static int __headset_playback_volume_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return set_mixer_volume_as_headset_gain_control_volume(ucontrol->value.integer.value[0]);
return set_mixer_volume_as_headset_gain_control_volume(
ucontrol->value.integer.value[0]);
}
static int __headset_playback_switch_info(struct snd_kcontrol *kcontrol,
......@@ -877,7 +931,8 @@ static int __headset_playback_switch_info(struct snd_kcontrol *kcontrol,
return 0;
}
/* When HGC_ADMUT_HED (bit 15) is 1, the headset is muted.
/*
* When HGC_ADMUT_HED (bit 15) is 1, the headset is muted.
* When HGC_ADMUT_HED is 0, headset is not muted.
*/
static int __headset_playback_switch_get(struct snd_kcontrol *kcontrol,
......@@ -891,7 +946,7 @@ static int __headset_playback_switch_get(struct snd_kcontrol *kcontrol,
static int __headset_playback_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
// mute/unmute headset
/* mute/unmute headset */
return adc_pga_unmute_control(ucontrol->value.integer.value[0],
TSC2101_HEADSET_GAIN_CTRL,
15);
......@@ -912,22 +967,24 @@ static int __handset_playback_volume_get(struct snd_kcontrol *kcontrol,
{
u16 val;
u16 vol;
val = omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
M_DPRINTK("registry value = %d\n", val);
vol = HNGC_ADPGA_HND_EXTRACT(val);
vol = vol & ~HNGC_ADMUT_HND;
vol = get_handset_gain_control_volume_as_mixer_volume(vol);
ucontrol->value.integer.value[0] = vol;
M_DPRINTK("mixer volume returned = %ld\n", ucontrol->value.integer.value[0]);
M_DPRINTK("mixer volume returned = %ld\n",
ucontrol->value.integer.value[0]);
return 0;
}
static int __handset_playback_volume_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
return set_mixer_volume_as_handset_gain_control_volume(ucontrol->value.integer.value[0]);
return set_mixer_volume_as_handset_gain_control_volume(
ucontrol->value.integer.value[0]);
}
static int __handset_playback_switch_info(struct snd_kcontrol *kcontrol,
......@@ -940,7 +997,8 @@ static int __handset_playback_switch_info(struct snd_kcontrol *kcontrol,
return 0;
}
/* When HNGC_ADMUT_HND (bit 15) is 1, the handset is muted.
/*
* When HNGC_ADMUT_HND (bit 15) is 1, the handset is muted.
* When HNGC_ADMUT_HND is 0, handset is not muted.
*/
static int __handset_playback_switch_get(struct snd_kcontrol *kcontrol,
......@@ -954,7 +1012,7 @@ static int __handset_playback_switch_get(struct snd_kcontrol *kcontrol,
static int __handset_playback_switch_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
// handset mute/unmute
/* handset mute/unmute */
return adc_pga_unmute_control(ucontrol->value.integer.value[0],
TSC2101_HANDSET_GAIN_CTRL,
15);
......@@ -970,7 +1028,8 @@ static int __cellphone_input_switch_info(struct snd_kcontrol *kcontrol,
return 0;
}
/* When BGC_MUT_CP (bit 15) = 1, power down cellphone input pga.
/*
* When BGC_MUT_CP (bit 15) = 1, power down cellphone input pga.
* When BGC_MUT_CP = 0, power up cellphone input pga.
*/
static int __cellphone_input_switch_get(struct snd_kcontrol *kcontrol,
......@@ -986,7 +1045,7 @@ static int __cellphone_input_switch_put(struct snd_kcontrol *kcontrol,
{
return adc_pga_unmute_control(ucontrol->value.integer.value[0],
TSC2101_BUZZER_GAIN_CTRL,
15);
15);
}
static int __buzzer_input_switch_info(struct snd_kcontrol *kcontrol,
......@@ -999,7 +1058,8 @@ static int __buzzer_input_switch_info(struct snd_kcontrol *kcontrol,
return 0;
}
/* When BGC_MUT_BU (bit 6) = 1, power down cellphone input pga.
/*
* When BGC_MUT_BU (bit 6) = 1, power down cellphone input pga.
* When BGC_MUT_BU = 0, power up cellphone input pga.
*/
static int __buzzer_input_switch_get(struct snd_kcontrol *kcontrol,
......@@ -1015,82 +1075,82 @@ static int __buzzer_input_switch_put(struct snd_kcontrol *kcontrol,
{
return adc_pga_unmute_control(ucontrol->value.integer.value[0],
TSC2101_BUZZER_GAIN_CTRL,
6);
6);
}
static struct snd_kcontrol_new tsc2101_control[] __devinitdata = {
{
.name = "Target Playback Route",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __pcm_playback_target_info,
.get = __pcm_playback_target_get,
.put = __pcm_playback_target_put,
.name = "Target Playback Route",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __pcm_playback_target_info,
.get = __pcm_playback_target_get,
.put = __pcm_playback_target_put,
}, {
.name = "Master Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __pcm_playback_volume_info,
.get = __pcm_playback_volume_get,
.put = __pcm_playback_volume_put,
.name = "Master Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __pcm_playback_volume_info,
.get = __pcm_playback_volume_get,
.put = __pcm_playback_volume_put,
}, {
.name = "Master Playback Switch",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __pcm_playback_switch_info,
.get = __pcm_playback_switch_get,
.put = __pcm_playback_switch_put,
.name = "Master Playback Switch",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __pcm_playback_switch_info,
.get = __pcm_playback_switch_get,
.put = __pcm_playback_switch_put,
}, {
.name = "Headset Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __headset_playback_volume_info,
.get = __headset_playback_volume_get,
.put = __headset_playback_volume_put,
.name = "Headset Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __headset_playback_volume_info,
.get = __headset_playback_volume_get,
.put = __headset_playback_volume_put,
}, {
.name = "Headset Playback Switch",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __headset_playback_switch_info,
.get = __headset_playback_switch_get,
.put = __headset_playback_switch_put,
.name = "Headset Playback Switch",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __headset_playback_switch_info,
.get = __headset_playback_switch_get,
.put = __headset_playback_switch_put,
}, {
.name = "Handset Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __handset_playback_volume_info,
.get = __handset_playback_volume_get,
.put = __handset_playback_volume_put,
.name = "Handset Playback Volume",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __handset_playback_volume_info,
.get = __handset_playback_volume_get,
.put = __handset_playback_volume_put,
}, {
.name = "Handset Playback Switch",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __handset_playback_switch_info,
.get = __handset_playback_switch_get,
.put = __handset_playback_switch_put,
.name = "Handset Playback Switch",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __handset_playback_switch_info,
.get = __handset_playback_switch_get,
.put = __handset_playback_switch_put,
}, {
.name = "Cellphone Input Switch",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __cellphone_input_switch_info,
.get = __cellphone_input_switch_get,
.put = __cellphone_input_switch_put,
.name = "Cellphone Input Switch",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __cellphone_input_switch_info,
.get = __cellphone_input_switch_get,
.put = __cellphone_input_switch_put,
}, {
.name = "Buzzer Input Switch",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __buzzer_input_switch_info,
.get = __buzzer_input_switch_get,
.put = __buzzer_input_switch_put,
.name = "Buzzer Input Switch",
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.index = 0,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = __buzzer_input_switch_info,
.get = __buzzer_input_switch_get,
.put = __buzzer_input_switch_put,
}
};
......@@ -1106,20 +1166,20 @@ void snd_omap_resume_mixer(void)
}
#endif
int snd_omap_mixer(struct snd_card_omap_codec *tsc2101)
int snd_omap_mixer(struct snd_card_omap_codec *tsc2101)
{
int i=0;
int err=0;
int i = 0;
int err = 0;
if (!tsc2101) {
if (!tsc2101)
return -EINVAL;
}
for (i=0; i < ARRAY_SIZE(tsc2101_control); i++) {
if ((err = snd_ctl_add(tsc2101->card,
snd_ctl_new1(&tsc2101_control[i],
tsc2101->card))) < 0) {
for (i = 0; i < ARRAY_SIZE(tsc2101_control); i++) {
err = snd_ctl_add(tsc2101->card,
snd_ctl_new1(&tsc2101_control[i],
tsc2101->card));
if (err < 0)
return err;
}
}
return 0;
}
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