]> pilppa.org Git - linux-2.6-omap-h63xx.git/blobdiff - sound/arm/omap/omap-alsa-tsc2101-mixer.c
Merge current mainline tree into linux-omap tree
[linux-2.6-omap-h63xx.git] / sound / arm / omap / omap-alsa-tsc2101-mixer.c
index 9dfaea8d7bda6c7ae3f7d4f7c68bb5163c0f9295..d443342260a042e77fce40975d25e8e1c7639e1b 100644 (file)
@@ -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 <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
  * 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"
 
+#include <linux/spi/tsc2101.h>
 #include <linux/types.h>
 #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))
 
 #define DGC_DALVL_EXTRACT(ARG) ((ARG & 0x7f00) >> 8)
 #define DGC_DARVL_EXTRACT(ARG) ((ARG & 0x007f))
-#define GET_DGC_DALMU_BIT_VALUE(ARG)  (((ARG) & TSC2101_BIT(15)) >> 15)
-#define GET_DGC_DARMU_BIT_VALUE(ARG)  (((ARG) & TSC2101_BIT(7)) >> 7)
-#define IS_DGC_DALMU_UNMUTED(ARG)  (((GET_DGC_DALMU_BIT_VALUE(ARG)) == 0))
-#define IS_DGC_DARMU_UNMUTED(ARG) (((GET_DGC_DARMU_BIT_VALUE(ARG)) == 0))
 
 #define HGC_ADPGA_HED_EXTRACT(ARG) ((ARG & 0x7f00) >> 8)
-#define GET_DGC_HGCMU_BIT_VALUE(ARG) (((ARG) & TSC2101_BIT(15)) >> 15)
-#define IS_DGC_HGCMU_UNMUTED(ARG) (((GET_DGC_HGCMU_BIT_VALUE(ARG)) == 0))
-
 #define HNGC_ADPGA_HND_EXTRACT(ARG) ((ARG & 0x7f00) >> 8)
-#define GET_DGC_HNGCMU_BIT_VALUE(ARG) (((ARG) & TSC2101_BIT(15)) >> 15)
-#define IS_DGC_HNGCMU_UNMUTED(ARG) (((GET_DGC_HNGCMU_BIT_VALUE(ARG)) == 0))
+#define BGC_ADPGA_BGC_EXTRACT(ARG) ((ARG & 0x7f00) >> 8)
 
 static int current_playback_target     = PLAYBACK_TARGET_LOUDSPEAKER;
 static int current_rec_src             = REC_SRC_SINGLE_ENDED_MICIN_HED;
 
 /*
- * Used for switching between TSC2101 recourd sources.
- * Logic is adjusted from the TSC2101 OSS code.
+ * Simplified write for the tsc2101 audio registers.
+ */
+inline void omap_tsc2101_audio_write(u8 address, u16 data)
+{
+       tsc2101_write_sync(mcbsp_dev.tsc2101_dev, PAGE2_AUDIO_CODEC_REGISTERS,
+                               address, data);
+}
+
+/*
+ * Simplified read for the tsc2101 audio registers.
  */
-static int set_record_source(int val)
+inline u16 omap_tsc2101_audio_read(u8 address)
+{
+       return (tsc2101_read_sync(mcbsp_dev.tsc2101_dev,
+                                       PAGE2_AUDIO_CODEC_REGISTERS, address));
+}
+
+/*
+ * For selecting tsc2101 recourd source.
+ */
+static void set_record_source(int val)
 {
        u16     data;
-       int     maskedVal;
-       
-       FN_IN;
-       maskedVal       = 0xe0 & val;   
 
-       data    = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                               TSC2101_MIXER_PGA_CTRL);
+       /*
+        * Mute Analog Sidetone
+        * Analog sidetone gain db?
+        * Input selected by MICSEL connected to ADC
+        */
+       data    = MPC_ASTMU | MPC_ASTG(0x45);
        data    &= ~MPC_MICSEL(7); /* clear all MICSEL bits */
-       data    |= maskedVal;
-       omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
-                               TSC2101_MIXER_PGA_CTRL,
-                               data);
-       current_rec_src = val;
+       data    |= MPC_MICSEL(val);
+       data    |= MPC_MICADC;
+       omap_tsc2101_audio_write(TSC2101_MIXER_PGA_CTRL, data);
 
-       FN_OUT(0);
-       return 0;
+       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_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_DAC_GAIN_CTRL);
+
+       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) {
-               omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                               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);
+       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);
        return retVal;
 }
 
-int dac_gain_control_unmute_control(int muteLeft, int muteRight)
+/*
+ * If unmuteLeft/unmuteRight == 0  --> mute
+ * If unmuteLeft/unmuteRight == 1 --> unmute
+ */
+int dac_gain_control_unmute(int unmuteLeft, int unmuteRight)
 {
        u16 val;
        int count;
 
        count   = 0;
-       val     = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 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.
+       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.
         */
-       if (muteLeft == GET_DGC_DALMU_BIT_VALUE(val)) {
-               if (muteLeft == 0) {
+       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;
                }
                count++;
        } /* L */
-       if (muteRight == GET_DGC_DARMU_BIT_VALUE(val)) {
-               if (muteRight == 0) {
+       if (unmuteRight != IS_UNMUTED(7, val)) {
+               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_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_DAC_GAIN_CTRL, val);
-               M_DPRINTK("changed value, is_unmuted left = %d, right = %d\n", 
-                       IS_DGC_DALMU_UNMUTED(val),
-                       IS_DGC_DARMU_UNMUTED(val));
+               omap_tsc2101_audio_write(TSC2101_DAC_GAIN_CTRL, val);
+               M_DPRINTK("changed value, is_unmuted left = %d, right = %d\n",
+                       IS_UNMUTED(15, val),
+                       IS_UNMUTED(7, val));
        }
-       return count;   
+       return count;
 }
 
 /*
- * Converts the DGC registry value read from the TSC2101 registry to 
+ * 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.
+ */
+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...
+        */
+       if (unmute != IS_UNMUTED(muteBitIndx, val)) {
+               if (unmute == 0) {
+                       /* mute --> turn bit on */
+                       val     = val | TSC2101_BIT(muteBitIndx);
+               } else {
+                       /* unmute --> turn bit off */
+                       val     = val & ~TSC2101_BIT(muteBitIndx);
+               }
+               M_DPRINTK("changed value, is_unmuted = %d\n",
+                               IS_UNMUTED(muteBitIndx, val));
+               count++;
+       }
+       if (count)
+               omap_tsc2101_audio_write(page2regIndx, val);
+
+       return count;
+}
+
+/*
+ * 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;
 }
 
@@ -223,10 +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;
 }
@@ -235,189 +297,378 @@ 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);    
-       val     = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                               TSC2101_HEADSET_GAIN_CTRL);
+       /*
+        * 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_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                       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_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_HANDSET_GAIN_CTRL);
+       val     = omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
        /* preserve the old mute settigns */
        val     &= ~(HNGC_ADPGA_HND(INPUT_VOLUME_MAX));
        val     |= HNGC_ADPGA_HND(volume);
-       omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                       TSC2101_HANDSET_GAIN_CTRL,
-                       val);
+       omap_tsc2101_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);
        retVal  = 1;
-       
-       M_DPRINTK("to registry = %d\n", val);   
-       return retVal;
-}
 
-void init_record_sources(void)
-{
-       /* Mute Analog Sidetone
-        * analog sidetone gain db?
-        * Cell Phone In not connected to ADC
-        * Input selected by MICSEL connected to ADC
-        */
-       omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                         TSC2101_MIXER_PGA_CTRL,
-                         MPC_ASTMU | MPC_ASTG(0x40) | ~MPC_CPADC | MPC_MICADC);
-       /* Set record source, Select MIC_INHED input for headset */
-       set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HED);      
+       M_DPRINTK("to registry = %d\n", val);
+       return retVal;
 }
 
 void set_loudspeaker_to_playback_target(void)
 {
-       u16     val;
-
-       /* power down sp1, sp2 and loudspeaker */
-       omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
-                       TSC2101_CODEC_POWER_CTRL,
-                       CPC_SP1PWDN | CPC_SP2PWDN | CPC_LDAPWDF);       
-       /* ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled
+       /* 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
         */
-       omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                       TSC2101_AUDIO_CTRL_4, 
-                       AC4_MB_HED(0));
+       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
-        */
-       val     = AC5_DIFFIN |
+        * Keyclicks routed to SPK1/SPK2 */
+       omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_5,
+                       AC5_DIFFIN |
                        AC5_DAC2SPK1(3) | AC5_AST2SPK1 | AC5_KCL2SPK1 |
-                       AC5_DAC2SPK2(3) | AC5_AST2SPK2 | AC5_KCL2SPK2 |
-                       AC5_HDSCPTC;
-       val     = val & ~AC5_HDSCPTC;
-       omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
-                       TSC2101_AUDIO_CTRL_5,
-                       val);
-       
-       /* powerdown spk1/out32n and spk2 */
-       val     = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
-                               TSC2101_POWERDOWN_STS);
-       val     = val & ~(~PS_SPK1FL | ~PS_HNDFL | PS_LSPKFL);
-       omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
-                       TSC2101_POWERDOWN_STS,
-                       val);
-
-       /* routing selected to SPK1 goes to OUT8P/OUT84 alsa. (loudspeaker)
+                       AC5_DAC2SPK2(3) | AC5_AST2SPK2 | AC5_KCL2SPK2);
+
+       /*
+        * 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)
         * Enable loudspeaker short protection control (0 = enable protection)
         * VGND short protection control (0 = enable protection)
         */
-       omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
-                       TSC2101_AUDIO_CTRL_6,
+       omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_6,
                        AC6_SPL2LSK | AC6_AST2LSK | AC6_BUZ2LSK | AC6_KCL2LSK |
-                       AC6_CPI2LSK | AC6_MIC2CPO | AC6_SPL2CPO |
-                       ~AC6_MUTLSPK | ~AC6_MUTSPK2 | ~AC6_LDSCPTC | ~AC6_VGNDSCPTC);
+                       AC6_CPI2LSK | AC6_MIC2CPO | AC6_SPL2CPO);
        current_playback_target = PLAYBACK_TARGET_LOUDSPEAKER;
 }
 
 void set_headphone_to_playback_target(void)
 {
-       /* power down sp1, sp2 and loudspeaker */
-       omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
-                       TSC2101_CODEC_POWER_CTRL,
+       /* 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 */
-       omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
-                               TSC2101_AUDIO_CTRL_4, 
-                               AC4_MB_HED(0));
-
-       /* DAC left and right routed to SPK2 */
-       /* SPK1/2 unmuted */
-       omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
-                       TSC2101_AUDIO_CTRL_5,
+       /*
+        * ADC, DAC, Analog Sidetone, cellphone, buzzer softstepping enabled
+       Â * 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
+        * SPK1/SPK2 unmuted
+        * 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/N muted, CPOUT muted */
-       omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
-                       TSC2101_AUDIO_CTRL_6,
+       /* OUT8P/OUT8N muted, CPOUT muted */
+       omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_6,
                        AC6_MUTLSPK | AC6_MUTSPK2 | AC6_LDSCPTC |
                        AC6_VGNDSCPTC);
        current_playback_target = PLAYBACK_TARGET_HEADPHONE;
 }
 
+void set_telephone_to_playback_target(void)
+{
+       /*
+        * 0110 1101 0101 1100
+        * 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_BISTPD | AC4_ASSTPD | AC4_DASTPD);
+       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);
+
+       omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_6,
+                       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)
+ * #define MPC_MICADC          TSC2101_BIT(4)
+ * #define MPC_CPADC           TSC2101_BIT(3)
+ * #define MPC_ASTGF           (0x01)
+ */
+static void set_telephone_to_record_source(void)
+{
+       u16     val;
+
+       /*
+        * 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,
+        *              attack time = 8 mc, decay time = 100 ms
+        * D7 - D5  = 000
+        *              --> AGC Target gain for handset input = -5.5 db
+        * D14 - D8 = 011 1100
+        *              --> ADC handset PGA settings = 60 = 30 db
+        * D15          = 0
+        *              --> Handset input ON (unmuted)
+        */
+       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
+        *          (D15, D14-D8)
+        * D4 - D1      = 0000
+        *              --> Agc constant for headset/Aux input,
+        *              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 */
+       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)
+        * D1           = 0, DAC channel PGA soft stepping control
+        *              --> 0.5 db change every WCLK
+        * D2           = flag, DAC right channel PGA flag
+        *              --> & with 1
+        * D3           = flag, DAC left channel PGA flag
+        *              -- > & with 1
+        * D7 - D4      = 0001, keyclick length
+        *              --> 4 periods key clicks
+        * 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
+        *              --> Medium amplitude
+        * D15          = 0, keyclick disabled
+        */
+       val     = omap_tsc2101_audio_read(TSC2101_AUDIO_CTRL_2);
+       val     = val & 0x441d;
+       val     = val | 0x4410; /* D14, D10, D4 bits == 1 */
+       omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_2, val);
+
+       /*
+        * D0           = 0     (reserved, write always 0)
+        * D1           = flag,
+        *                      --> & with 1
+        * D2 - D5      = 0000 (reserved, write always 0000)
+        * D6           = 1
+        *                      --> MICBIAS_HND = 2.0 v
+        * D8 - D7      = 00
+        *                      --> MICBIAS_HED = 3.3 v
+        * D10 - D9     = 01,
+        *                      --> Mic AGC hysteric selection = 2 db
+        * D11          = 1,
+        *                      --> Disable buzzer PGA soft stepping
+        * D12          = 0,
+        *                      --> Enable CELL phone PGA soft stepping control
+        * D13          = 1
+        *                      --> Disable analog sidetone soft
+        *                      stepping control
+        * D14          = 0
+        *                      --> Enable DAC PGA soft stepping control
+        * D15          = 0,
+        *                      --> 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 */
+       omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_4, val);
+       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
+        * D1           = flag, read only
+        *              --> & with 1
+        * D5 - D2      = 1111, Buzzer input PGA settings
+        *              --> 0 db
+        * D6           = 1,
+        *              --> power down buzzer input pga
+        * D7           = flag, read only
+        *              --> & with 1
+        * D14 - D8     = 101 1101
+        *              --> 12 DB
+        * D15          = 0
+        *              --> power up cell phone input PGA
+        */
+       val     = omap_tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL);
+       val     = val & 0x5dfe;
+       /* 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
+        *              --> -4.5 db for DAC right channel volume control
+        * D7           = 1
+        *              -->  DAC right channel muted
+        * D14 - D8 = 000 1001
+        *              --> -4.5 db for DAC left channel volume control
+        * D15          = 1
+        *              --> DAC left channel muted
+        */
+       /* val  = omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL); */
+       val     = 0x8989;
+       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
+        *              --> Disaple GPIO2 for CLKOUT mode
+        * D3           = 0
+        *              --> Disable GPUI1 for interrupt detection
+        * D4           = 0
+        *              --> Disable GPIO2 for headset detection interrupt
+        * D5           = reserved, always 0
+        * D7 - D6      = 01
+        *              --> 8 ms clitch detection
+        * D8           = reserved, write only 0
+        * D10 -D9      = 00
+        *              --> 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
+        * D15          = 0
+        *              --> Disable headset detection
+        */
+       val     = 0x40;
+       omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_7, val);
+}
+
 /*
  * Checks whether the headset is detected.
  * If headset is detected, the type is returned. Type can be
@@ -431,18 +682,17 @@ u16 get_headset_detected(void)
        u16     curDetected;
        u16     curType;
        u16     curVal;
-       
+
        curType = 0;    /* not detected */
-       curVal  = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                                       TSC2101_AUDIO_CTRL_7);
+       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;
 }
@@ -452,99 +702,110 @@ 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_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                               TSC2101_HANDSET_GAIN_CTRL);
-       val     = val | HNGC_AGCEN_HND; 
+       val     = omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
+       val     = val | HNGC_AGCEN_HND;
        val     = val & ~HNGC_ADMUT_HND;
-       omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                       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_control(1, 1);
+       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_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
-                       TSC2101_AUDIO_CTRL_7,
-                       AC7_DETECT);
+       omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_7, AC7_DETECT);
 
-       init_record_sources();
+       /* Select headset to record source (MIC_INHED)*/
+       set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HED);
+       /* Init loudspeaker as a default playback target*/
        init_playback_targets();
 
        FN_OUT(0);
 }
 
-static int __pcm_playback_target_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) 
+static int __pcm_playback_target_info(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_info *uinfo)
 {
        static char *texts[PLAYBACK_TARGET_COUNT] = {
-               "Loudspeaker", "Headphone"
+               "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;
 }
 
-static int __pcm_playback_target_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+static int __pcm_playback_target_get(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_value *ucontrol)
 {
        ucontrol->value.integer.value[0] = current_playback_target;
        return 0;
 }
 
-static int __pcm_playback_target_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+static int __pcm_playback_target_put(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_value *ucontrol)
 {
        int     retVal;
        int     curVal;
-       
+
        retVal  = 0;
        curVal  = ucontrol->value.integer.value[0];
        if ((curVal >= 0) &&
            (curVal < PLAYBACK_TARGET_COUNT) &&
-           (curVal != current_playback_target)) {              
-               if (curVal == 0) {
-                       set_loudspeaker_to_playback_target();           
-               }
-               else {
+           (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) {
+                       set_record_source(REC_SRC_SINGLE_ENDED_MICIN_HND);
                        set_headphone_to_playback_target();
+               } 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(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) 
+static int __pcm_playback_volume_info(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_info *uinfo)
 {
        uinfo->type                     = SNDRV_CTL_ELEM_TYPE_INTEGER;
        uinfo->count                    = 2;
@@ -554,16 +815,17 @@ static int __pcm_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_inf
 }
 
 /*
- * 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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+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_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_DAC_GAIN_CTRL);
+
+       val     = omap_tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL);
        M_DPRINTK("registry value = %d!\n", val);
        volL    = DGC_DALVL_EXTRACT(val);
        volR    = DGC_DARVL_EXTRACT(val);
@@ -573,21 +835,26 @@ static int __pcm_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_valu
 
        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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+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(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) 
+static int __pcm_playback_switch_info(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_info *uinfo)
 {
        uinfo->type                     = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
        uinfo->count                    = 2;
@@ -596,27 +863,30 @@ static int __pcm_playback_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_inf
        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.
  */
-static int __pcm_playback_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+static int __pcm_playback_switch_get(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_value *ucontrol)
 {
-       u16 val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_DAC_GAIN_CTRL);
-       
-       ucontrol->value.integer.value[0]        = IS_DGC_DALMU_UNMUTED(val);
-       ucontrol->value.integer.value[1]        = IS_DGC_DARMU_UNMUTED(val);
+       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 */
        return 0;
 }
 
-static int __pcm_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+static int __pcm_playback_switch_put(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_value *ucontrol)
 {
-       return dac_gain_control_unmute_control(ucontrol->value.integer.value[0], 
+       return dac_gain_control_unmute(ucontrol->value.integer.value[0],
                                        ucontrol->value.integer.value[1]);
 }
 
-static int __headset_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) 
+static int __headset_playback_volume_info(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_info *uinfo)
 {
        uinfo->type                     = SNDRV_CTL_ELEM_TYPE_INTEGER;
        uinfo->count                    = 1;
@@ -625,30 +895,34 @@ static int __headset_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem
        return 0;
 }
 
-static int __headset_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+static int __headset_playback_volume_get(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_value *ucontrol)
 {
        u16 val;
        u16 vol;
-       
-       val     = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                               TSC2101_HEADSET_GAIN_CTRL);
+
+       val     = omap_tsc2101_audio_read(TSC2101_HEADSET_GAIN_CTRL);
        M_DPRINTK("registry value = %d\n", val);
        vol     = HGC_ADPGA_HED_EXTRACT(val);
        vol     = vol & ~HGC_ADMUT_HED;
 
        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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+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(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) 
+static int __headset_playback_switch_info(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_info *uinfo)
 {
        uinfo->type                     = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
        uinfo->count                    = 1;
@@ -657,46 +931,29 @@ static int __headset_playback_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem
        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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+static int __headset_playback_switch_get(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_value *ucontrol)
 {
-       u16 val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                               TSC2101_HEADSET_GAIN_CTRL);
-       ucontrol->value.integer.value[0]        = IS_DGC_HGCMU_UNMUTED(val);
+       u16 val = omap_tsc2101_audio_read(TSC2101_HEADSET_GAIN_CTRL);
+       ucontrol->value.integer.value[0]        = IS_UNMUTED(15, val);
        return 0;
 }
 
-static int __headset_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+static int __headset_playback_switch_put(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_value *ucontrol)
 {
-       int count = 0;
-       u16 val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                               TSC2101_HEADSET_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...
-        */
-       if (ucontrol->value.integer.value[0] == GET_DGC_HGCMU_BIT_VALUE(val)) {
-               if (ucontrol->value.integer.value[0] == 0) {
-                       /* mute --> turn bit on */
-                       val     = val | HGC_ADMUT_HED;
-               }
-               else {
-                       /* unmute --> turn bit off */
-                       val     = val & ~HGC_ADMUT_HED;
-               }
-               count++;
-               M_DPRINTK("changed value, is_unmuted = %d\n", IS_DGC_HGCMU_UNMUTED(val));
-       }
-       if (count) {
-               omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                               TSC2101_HEADSET_GAIN_CTRL, 
-                               val);
-       }
-       return count;
+       /* mute/unmute headset */
+       return adc_pga_unmute_control(ucontrol->value.integer.value[0],
+                               TSC2101_HEADSET_GAIN_CTRL,
+                               15);
 }
 
-static int __handset_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) 
+static int __handset_playback_volume_info(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_info *uinfo)
 {
        uinfo->type                     = SNDRV_CTL_ELEM_TYPE_INTEGER;
        uinfo->count                    = 1;
@@ -705,28 +962,33 @@ static int __handset_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem
        return 0;
 }
 
-static int __handset_playback_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+static int __handset_playback_volume_get(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_value *ucontrol)
 {
        u16 val;
        u16 vol;
-       
-       val     = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_HANDSET_GAIN_CTRL);
+
+       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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+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(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo) 
+static int __handset_playback_switch_info(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_info *uinfo)
 {
        uinfo->type                     = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
        uinfo->count                    = 1;
@@ -735,102 +997,161 @@ static int __handset_playback_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem
        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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+static int __handset_playback_switch_get(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_value *ucontrol)
 {
-       u16 val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_HANDSET_GAIN_CTRL);
-       ucontrol->value.integer.value[0]        = IS_DGC_HNGCMU_UNMUTED(val);
+       u16 val = omap_tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL);
+       ucontrol->value.integer.value[0]        = IS_UNMUTED(15, val);
        return 0;
 }
 
-static int __handset_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol) 
+static int __handset_playback_switch_put(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_value *ucontrol)
 {
-       int count = 0;
-       u16 val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, TSC2101_HANDSET_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...
-        */
-       if (ucontrol->value.integer.value[0] == GET_DGC_HNGCMU_BIT_VALUE(val)) {
-               if (ucontrol->value.integer.value[0] == 0) {
-                       /* mute --> turn bit on */
-                       val     = val | HNGC_ADMUT_HND;
-               }
-               else {
-                       /* unmute --> turn bit off */
-                       val     = val & ~HNGC_ADMUT_HND;
-               }
-               M_DPRINTK("changed value, is_unmuted = %d\n", IS_DGC_HNGCMU_UNMUTED(val));
-               count++;
-       }
-       if (count) {
-               omap_tsc2101_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 
-                               TSC2101_HANDSET_GAIN_CTRL, 
-                               val);
-       }
-       return count;
+       /* handset mute/unmute */
+       return adc_pga_unmute_control(ucontrol->value.integer.value[0],
+                               TSC2101_HANDSET_GAIN_CTRL,
+                               15);
+}
+
+static int __cellphone_input_switch_info(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_info *uinfo)
+{
+       uinfo->type                     = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+       uinfo->count                    = 1;
+       uinfo->value.integer.min        = 0;
+       uinfo->value.integer.max        = 1;
+       return 0;
+}
+
+/*
+ * 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,
+               struct snd_ctl_elem_value *ucontrol)
+{
+       u16 val = omap_tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL);
+       ucontrol->value.integer.value[0]        = IS_UNMUTED(15, val);
+       return 0;
+}
+
+static int __cellphone_input_switch_put(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_value *ucontrol)
+{
+       return adc_pga_unmute_control(ucontrol->value.integer.value[0],
+                               TSC2101_BUZZER_GAIN_CTRL,
+                               15);
 }
 
-static snd_kcontrol_new_t tsc2101_control[] __devinitdata = {
+static int __buzzer_input_switch_info(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_info *uinfo)
+{
+       uinfo->type                     = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+       uinfo->count                    = 1;
+       uinfo->value.integer.min        = 0;
+       uinfo->value.integer.max        = 1;
+       return 0;
+}
+
+/*
+ * 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,
+               struct snd_ctl_elem_value *ucontrol)
+{
+       u16 val = omap_tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL);
+       ucontrol->value.integer.value[0]        = IS_UNMUTED(6, val);
+       return 0;
+}
+
+static int __buzzer_input_switch_put(struct snd_kcontrol *kcontrol,
+               struct snd_ctl_elem_value *ucontrol)
+{
+       return adc_pga_unmute_control(ucontrol->value.integer.value[0],
+                               TSC2101_BUZZER_GAIN_CTRL,
+                               6);
+}
+
+static struct snd_kcontrol_new tsc2101_control[] __devinitdata = {
        {
-               .name  = "Playback 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 = 1,
-               .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 = 1,
-               .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 = 2,
-               .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 = 2,
-               .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   = "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,
+       }
 };
 
 #ifdef CONFIG_PM
@@ -845,20 +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;
 }