+++ /dev/null
-/*
- * 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
- * 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
- * option) any later version.
- *
- * THIS SOFTWARE IS PROVIDED ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN
- * NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
- * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
- * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF
- * USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
- * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
- * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
- * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- *
- * You should have received a copy of the GNU General Public License along
- * with this program; if not, write to the Free Software Foundation, Inc.,
- * 675 Mass Ave, Cambridge, MA 02139, USA.
- *
- * History:
- *
- * 2006-03-01 Mika Laitio - Mixer for the tsc2101 driver used in omap boards.
- * 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
- * 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>
-
-#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 HGC_ADPGA_HED_EXTRACT(ARG) ((ARG & 0x7f00) >> 8)
-#define HNGC_ADPGA_HND_EXTRACT(ARG) ((ARG & 0x7f00) >> 8)
-#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;
-
-/*
- * 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.
- */
-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;
-
- /*
- * 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 |= 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
- * 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
- * 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)
- * 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.
- * ([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;
- /* invert the value for getting the proper range 0 min and 100 max */
- retVal = OUTPUT_VOLUME_MIN - retVal;
-
- return retVal;
-}
-
-/*
- * 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)
-{
- u16 val;
- int retVal;
- int volL;
- int volR;
-
- 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);
- return -EPERM;
- }
- 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(volL) | DGC_DARVL(volR);
- retVal = 2;
- 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;
-}
-
-/*
- * 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_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 (unmuteLeft != IS_UNMUTED(15, val)) {
- if (unmuteLeft == 0) {
- /* mute --> turn bit on */
- val = val | DGC_DALMU;
- } else {
- /* unmute --> turn bit off */
- val = val & ~DGC_DALMU;
- }
- count++;
- } /* L */
- if (unmuteRight != IS_UNMUTED(7, val)) {
- if (unmuteRight == 0) {
- /* mute --> turn bit on */
- val = val | DGC_DARMU;
- } 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",
- IS_UNMUTED(15, val),
- IS_UNMUTED(7, val));
- }
- 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.
- */
-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)
-{
- u16 retVal;
-
- retVal = OUTPUT_VOLUME_MIN - vol;
- retVal = ((retVal - OUTPUT_VOLUME_MAX) * 100) / OUTPUT_VOLUME_RANGE;
- /* fix scaling error */
- if ((retVal > 0) && (retVal < 100))
- retVal++;
-
- return retVal;
-}
-
-/*
- * 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)
-{
- u16 retVal;
-
- retVal = ((registerVal * 100) / INPUT_VOLUME_RANGE);
- return retVal;
-}
-
-/*
- * 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)
-{
- return get_headset_gain_control_volume_as_mixer_volume(registerVal);
-}
-
-/*
- * 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)
-{
- u16 retVal;
-
- 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.
- */
-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);
- 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_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);
- retVal = 1;
-
- 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.
- */
-int set_mixer_volume_as_handset_gain_control_volume(int mixerVol)
-{
- int volume;
- int retVal;
- u16 val;
-
- if (mixerVol < 0 || mixerVol > 100) {
- 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
- */
- volume = get_mixer_volume_as_headset_gain_control_volume(mixerVol);
- 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_audio_write(TSC2101_HANDSET_GAIN_CTRL, val);
- retVal = 1;
-
- M_DPRINTK("to registry = %d\n", val);
- return retVal;
-}
-
-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
- * 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_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)
- * 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)
- * 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_audio_write(TSC2101_AUDIO_CTRL_6,
- AC6_SPL2LSK | AC6_AST2LSK | AC6_BUZ2LSK | AC6_KCL2LSK |
- AC6_CPI2LSK | AC6_MIC2CPO | AC6_SPL2CPO);
- current_playback_target = PLAYBACK_TARGET_LOUDSPEAKER;
-}
-
-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
- */
- 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/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
- * 0x01 = stereo headset detected
- * 0x02 = cellurar headset detected
- * 0x03 = stereo + cellurar headset detected
- * If headset is not detected 0 is returned.
- */
-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(KERN_INFO "headset detected, checking type from %d \n",
- curVal);
- curType = ((curVal & 0x6000) >> 13);
- printk(KERN_INFO "headset type detected = %d \n", curType);
- } else {
- printk(KERN_INFO "headset not detected\n");
- }
- return curType;
-}
-
-void init_playback_targets(void)
-{
- u16 val;
-
- set_loudspeaker_to_playback_target();
- /*
- * 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
- * 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_ADMUT_HND;
- 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
- * Zero-cross detect on
- */
- 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)
- */
-void snd_omap_init_mixer(void)
-{
- FN_IN;
-
- /* Headset/Hook switch detect enabled */
- omap_tsc2101_audio_write(TSC2101_AUDIO_CTRL_7, AC7_DETECT);
-
- /* 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(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- static char *texts[PLAYBACK_TARGET_COUNT] = {
- "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;
-
- strcpy(uinfo->value.enumerated.name,
- texts[uinfo->value.enumerated.item]);
- return 0;
-}
-
-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(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 == 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(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_info *uinfo)
-{
- uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
- uinfo->count = 2;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 100;
- return 0;
-}
-
-/*
- * 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 val;
-
- 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);
- /* make sure that other bits are not on */
- volL = volL & ~DGC_DALMU;
- volR = volR & ~DGC_DARMU;
-
- 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]);
- 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]);
-}
-
-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;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- 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(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 */
- 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],
- ucontrol->value.integer.value[1]);
-}
-
-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;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 100;
- return 0;
-}
-
-static int __headset_playback_volume_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- 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);
- 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]);
- 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]);
-}
-
-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;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
-
-/*
- * 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,
- struct snd_ctl_elem_value *ucontrol)
-{
- 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(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- /* mute/unmute headset */
- return adc_pga_unmute_control(ucontrol->value.integer.value[0],
- TSC2101_HEADSET_GAIN_CTRL,
- 15);
-}
-
-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;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 100;
- return 0;
-}
-
-static int __handset_playback_volume_get(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- 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]);
- 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]);
-}
-
-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;
- uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 1;
- return 0;
-}
-
-/*
- * 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,
- struct snd_ctl_elem_value *ucontrol)
-{
- 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(struct snd_kcontrol *kcontrol,
- struct snd_ctl_elem_value *ucontrol)
-{
- /* 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 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 = "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 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 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 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 = "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
-
-void snd_omap_suspend_mixer(void)
-{
-}
-
-void snd_omap_resume_mixer(void)
-{
- snd_omap_init_mixer();
-}
-#endif
-
-int snd_omap_mixer(struct snd_card_omap_codec *tsc2101)
-{
- int i = 0;
- int err = 0;
-
- if (!tsc2101)
- return -EINVAL;
-
- 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;
-}
projects / linux-2.6-omap-h63xx.git / blobdiff