From: Eduardo Valentin <[EMAIL PROTECTED]>
Removed lots of whitespaces and a few errors and
warnings reported by checkpatch.pl.
Signed-off-by: Eduardo Valentin <[EMAIL PROTECTED]>
---
sound/arm/omap/omap-alsa-tsc2101-mixer.c | 624 ++++++++++++++++--------------
1 files changed, 342 insertions(+), 282 deletions(-)
diff --git a/sound/arm/omap/omap-alsa-tsc2101-mixer.c
b/sound/arm/omap/omap-alsa-tsc2101-mixer.c
index 09257d7..d443342 100644
--- a/sound/arm/omap/omap-alsa-tsc2101-mixer.c
+++ b/sound/arm/omap/omap-alsa-tsc2101-mixer.c
@@ -1,15 +1,15 @@
/*
* sound/arm/omap/omap-alsa-tsc2101-mixer.c
- *
+ *
* Alsa Driver for TSC2101 codec for OMAP platform boards.
*
- * Copyright (C) 2005 Mika Laitio <[EMAIL PROTECTED]> and
+ * Copyright (C) 2005 Mika Laitio <[EMAIL PROTECTED]> and
* Everett Coleman II <[EMAIL PROTECTED]>
*
* 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;
}
--
1.5.5-rc3.GIT
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