--- /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/types.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+
+//#define M_DPRINTK(ARGS...) printk(KERN_INFO "<%s>: ",__FUNCTION__);printk(ARGS)
+#define M_DPRINTK(ARGS...) /* nop */
+
+#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))
+
+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.
+ */
+static int 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);
+ 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;
+
+ FN_OUT(0);
+ return 0;
+}
+
+/*
+ * 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_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 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_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);
+ return retVal;
+}
+
+int dac_gain_control_unmute_control(int muteLeft, int muteRight)
+{
+ 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.
+ */
+ if (muteLeft == GET_DGC_DALMU_BIT_VALUE(val)) {
+ if (muteLeft == 0) {
+ /* mute --> turn bit on */
+ val = val | DGC_DALMU;
+ }
+ else {
+ /* unmute --> turn bit off */
+ val = val & ~DGC_DALMU;
+ }
+ count++;
+ } /* L */
+ if (muteRight == GET_DGC_DARMU_BIT_VALUE(val)) {
+ if (muteRight == 0) {
+ /* mute --> turn bit on */
+ val = val | DGC_DARMU;
+ }
+ 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));
+ }
+ 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_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ 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);
+ 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_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 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);
+ 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);
+}
+
+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
+ * 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 SPK1/SPK2
+ * SPK1/SPK2 unmuted
+ * keyclicks routed to SPK1/SPK2
+ */
+ val = 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)
+ * 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_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ 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);
+ 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,
+ 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,
+ 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,
+ AC6_MUTLSPK | AC6_MUTSPK2 | AC6_LDSCPTC |
+ AC6_VGNDSCPTC);
+ current_playback_target = PLAYBACK_TARGET_HEADPHONE;
+}
+
+/*
+ * 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_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_AUDIO_CTRL_7);
+ curDetected = curVal & AC7_HDDETFL;
+ if (curDetected) {
+ printk("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");
+ }
+ 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_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ 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
+ */
+ 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_control(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_write(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ TSC2101_AUDIO_CTRL_7,
+ AC7_DETECT);
+
+ init_record_sources();
+ init_playback_targets();
+
+ FN_OUT(0);
+}
+
+static int __pcm_playback_target_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+ static char *texts[PLAYBACK_TARGET_COUNT] = {
+ "Loudspeaker", "Headphone"
+ };
+
+ 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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *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)
+{
+ 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 {
+ set_headphone_to_playback_target();
+ }
+ retVal = 1;
+ }
+ return retVal;
+}
+
+static int __pcm_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ u16 volL;
+ u16 volR;
+ u16 val;
+
+ val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *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(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *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);
+ return 0;
+}
+
+static int __pcm_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ return dac_gain_control_unmute_control(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)
+{
+ 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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ u16 val;
+ u16 vol;
+
+ val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2,
+ 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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ 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)
+{
+ 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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *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);
+ return 0;
+}
+
+static int __headset_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *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;
+}
+
+static int __handset_playback_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ u16 val;
+ u16 vol;
+
+ val = omap_tsc2101_read(TSC2101_AUDIO_CODEC_REGISTERS_PAGE2, 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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+ 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)
+{
+ 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(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *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);
+ return 0;
+}
+
+static int __handset_playback_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *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;
+}
+
+static snd_kcontrol_new_t 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 = "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 = 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 = 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 = 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 = 2,
+ .access= SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = __handset_playback_switch_info,
+ .get = __handset_playback_switch_get,
+ .put = __handset_playback_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++) {
+ if ((err = snd_ctl_add(tsc2101->card,
+ snd_ctl_new1(&tsc2101_control[i],
+ tsc2101->card))) < 0) {
+ return err;
+ }
+ }
+ return 0;
+}
--- /dev/null
+/*
+ * arch/arm/mach-omap1/omap-alsa-tsc2101.c
+ *
+ * Alsa codec Driver for TSC2101 chip for OMAP platform boards.
+ * Code obtained from oss omap drivers
+ *
+ * Copyright (C) 2004 Texas Instruments, Inc.
+ * Written by Nishanth Menon and Sriram Kannan
+ *
+ * Copyright (C) 2006 Instituto Nokia de Tecnologia - INdT - Manaus Brazil
+ * Alsa modularization by Daniel Petrini (d.pensator@gmail.com)
+ *
+ * Copyright (C) 2006 Mika Laitio <lamikr@cc.jyu.fi>
+ *
+ * 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.
+ */
+
+#include <linux/delay.h>
+#include <linux/soundcard.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+#include <asm/io.h>
+#include <asm/arch/mcbsp.h>
+
+#include <linux/slab.h>
+#ifdef CONFIG_PM
+#include <linux/pm.h>
+#endif
+#include <asm/mach-types.h>
+#include <asm/arch/dma.h>
+#include <asm/arch/clock.h>
+
+#include <asm/hardware/tsc2101.h>
+#include <../drivers/ssi/omap-tsc2101.h>
+
+#include <asm/arch/omap-alsa.h>
+#include "omap-alsa-tsc2101.h"
+
+static struct clk *tsc2101_mclk = 0;
+
+//#define DUMP_TSC2101_AUDIO_REGISTERS
+#undef DUMP_TSC2101_AUDIO_REGISTERS
+
+/*
+ * Hardware capabilities
+ */
+
+/*
+ * DAC USB-mode sampling rates (MCLK = 12 MHz)
+ * The rates and rate_reg_into MUST be in the same order
+ */
+static unsigned int rates[] = {
+ 7350, 8000, 8018, 8727,
+ 8820, 9600, 11025, 12000,
+ 14700, 16000, 22050, 24000,
+ 29400, 32000, 44100, 48000,
+};
+
+static snd_pcm_hw_constraint_list_t tsc2101_hw_constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+};
+
+static const struct tsc2101_samplerate_reg_info
+ rate_reg_info[NUMBER_SAMPLE_RATES_SUPPORTED] = {
+ /* Div 6 */
+ {7350, 7, 1},
+ {8000, 7, 0},
+ /* Div 5.5 */
+ {8018, 6, 1},
+ {8727, 6, 0},
+ /* Div 5 */
+ {8820, 5, 1},
+ {9600, 5, 0},
+ /* Div 4 */
+ {11025, 4, 1},
+ {12000, 4, 0},
+ /* Div 3 */
+ {14700, 3, 1},
+ {16000, 3, 0},
+ /* Div 2 */
+ {22050, 2, 1},
+ {24000, 2, 0},
+ /* Div 1.5 */
+ {29400, 1, 1},
+ {32000, 1, 0},
+ /* Div 1 */
+ {44100, 0, 1},
+ {48000, 0, 0},
+};
+
+static snd_pcm_hardware_t tsc2101_snd_omap_alsa_playback = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+ SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 7350,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8 * 1024,
+ .periods_min = 16,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+static snd_pcm_hardware_t tsc2101_snd_omap_alsa_capture = {
+ .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+ SNDRV_PCM_RATE_16000 |
+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_KNOT),
+ .rate_min = 7350,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 128 * 1024,
+ .period_bytes_min = 32,
+ .period_bytes_max = 8 * 1024,
+ .periods_min = 16,
+ .periods_max = 255,
+ .fifo_size = 0,
+};
+
+/*
+ * Simplified write for tsc Audio
+ */
+inline void tsc2101_audio_write(u8 address, u16 data)
+{
+ omap_tsc2101_write(PAGE2_AUDIO_CODEC_REGISTERS, address, data);
+}
+
+/*
+ * Simplified read for tsc Audio
+ */
+inline u16 tsc2101_audio_read(u8 address)
+{
+ return (omap_tsc2101_read(PAGE2_AUDIO_CODEC_REGISTERS, address));
+}
+
+#ifdef DUMP_TSC2101_AUDIO_REGISTERS
+void dump_tsc2101_audio_reg(void) {
+ printk("TSC2101_AUDIO_CTRL_1 = 0x%04x\n", tsc2101_audio_read(TSC2101_AUDIO_CTRL_1));
+ printk("TSC2101_HEADSET_GAIN_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_HEADSET_GAIN_CTRL));
+ printk("TSC2101_DAC_GAIN_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_DAC_GAIN_CTRL));
+ printk("TSC2101_MIXER_PGA_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_MIXER_PGA_CTRL));
+ printk("TSC2101_AUDIO_CTRL_2 = 0x%04x\n", tsc2101_audio_read(TSC2101_AUDIO_CTRL_2));
+ printk("TSC2101_CODEC_POWER_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_CODEC_POWER_CTRL));
+ printk("TSC2101_AUDIO_CTRL_3 = 0x%04x\n", tsc2101_audio_read(TSC2101_AUDIO_CTRL_3));
+ printk("TSC2101_LCH_BASS_BOOST_N0 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N0));
+ printk("TSC2101_LCH_BASS_BOOST_N1 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N1));
+ printk("TSC2101_LCH_BASS_BOOST_N2 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N2));
+ printk("TSC2101_LCH_BASS_BOOST_N3 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N3));
+ printk("TSC2101_LCH_BASS_BOOST_N4 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N4));
+ printk("TSC2101_LCH_BASS_BOOST_N5 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_N5));
+ printk("TSC2101_LCH_BASS_BOOST_D1 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D1));
+ printk("TSC2101_LCH_BASS_BOOST_D2 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D2));
+ printk("TSC2101_LCH_BASS_BOOST_D4 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D4));
+ printk("TSC2101_LCH_BASS_BOOST_D5 = 0x%04x\n", tsc2101_audio_read(TSC2101_LCH_BASS_BOOST_D5));
+
+ printk("TSC2101_RCH_BASS_BOOST_N0 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N0));
+ printk("TSC2101_RCH_BASS_BOOST_N1 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N1));
+ printk("TSC2101_RCH_BASS_BOOST_N2 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N2));
+ printk("TSC2101_RCH_BASS_BOOST_N3 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N3));
+ printk("TSC2101_RCH_BASS_BOOST_N4 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N4));
+ printk("TSC2101_RCH_BASS_BOOST_N5 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_N5));
+ printk("TSC2101_RCH_BASS_BOOST_D1 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D1));
+ printk("TSC2101_RCH_BASS_BOOST_D2 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D2));
+ printk("TSC2101_RCH_BASS_BOOST_D4 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D4));
+ printk("TSC2101_RCH_BASS_BOOST_D5 = 0x%04x\n", tsc2101_audio_read(TSC2101_RCH_BASS_BOOST_D5));
+
+ printk("TSC2101_PLL_PROG_1 = 0x%04x\n", tsc2101_audio_read(TSC2101_PLL_PROG_1));
+ printk("TSC2101_PLL_PROG_1 = 0x%04x\n", tsc2101_audio_read(TSC2101_PLL_PROG_2));
+ printk("TSC2101_AUDIO_CTRL_4 = 0x%04x\n", tsc2101_audio_read(TSC2101_AUDIO_CTRL_4));
+ printk("TSC2101_HANDSET_GAIN_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_HANDSET_GAIN_CTRL));
+ printk("TSC2101_BUZZER_GAIN_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_BUZZER_GAIN_CTRL));
+ printk("TSC2101_AUDIO_CTRL_5 = 0x%04x\n", tsc2101_audio_read(TSC2101_AUDIO_CTRL_5));
+ printk("TSC2101_AUDIO_CTRL_6 = 0x%04x\n", tsc2101_audio_read(TSC2101_AUDIO_CTRL_6));
+ printk("TSC2101_AUDIO_CTRL_7 = 0x%04x\n", tsc2101_audio_read(TSC2101_AUDIO_CTRL_7));
+ printk("TSC2101_GPIO_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_GPIO_CTRL));
+ printk("TSC2101_AGC_CTRL = 0x%04x\n", tsc2101_audio_read(TSC2101_AGC_CTRL));
+ printk("TSC2101_POWERDOWN_STS = 0x%04x\n", tsc2101_audio_read(TSC2101_POWERDOWN_STS));
+ printk("TSC2101_MIC_AGC_CONTROL = 0x%04x\n", tsc2101_audio_read(TSC2101_MIC_AGC_CONTROL));
+ printk("TSC2101_CELL_AGC_CONTROL = 0x%04x\n", tsc2101_audio_read(TSC2101_CELL_AGC_CONTROL));
+}
+#endif
+
+/*
+ * ALSA operations according to board file
+ */
+
+/*
+ * Sample rate changing
+ */
+void tsc2101_set_samplerate(long sample_rate)
+{
+ u8 count = 0;
+ u16 data = 0;
+ int clkgdv = 0;
+
+ u16 srgr1, srgr2;
+ /* wait for any frame to complete */
+ udelay(125);
+ ADEBUG();
+
+ sample_rate = sample_rate;
+ /* Search for the right sample rate */
+ while ((rate_reg_info[count].sample_rate != sample_rate) &&
+ (count < NUMBER_SAMPLE_RATES_SUPPORTED)) {
+ count++;
+ }
+ if (count == NUMBER_SAMPLE_RATES_SUPPORTED) {
+ printk(KERN_ERR "Invalid Sample Rate %d requested\n",
+ (int) sample_rate);
+ return; // -EPERM;
+ }
+
+ /* Set AC1 */
+ data = tsc2101_audio_read(TSC2101_AUDIO_CTRL_1);
+ /* Clear prev settings */
+ data &= ~(AC1_DACFS(0x07) | AC1_ADCFS(0x07));
+ data |= AC1_DACFS(rate_reg_info[count].divisor) |
+ AC1_ADCFS(rate_reg_info[count].divisor);
+ tsc2101_audio_write(TSC2101_AUDIO_CTRL_1, data);
+
+ /* Set the AC3 */
+ data = tsc2101_audio_read(TSC2101_AUDIO_CTRL_3);
+ /*Clear prev settings */
+ data &= ~(AC3_REFFS | AC3_SLVMS);
+ data |= (rate_reg_info[count].fs_44kHz) ? AC3_REFFS : 0;
+#ifdef TSC_MASTER
+ data |= AC3_SLVMS;
+#endif /* #ifdef TSC_MASTER */
+ tsc2101_audio_write(TSC2101_AUDIO_CTRL_3, data);
+
+ /* program the PLLs */
+ if (rate_reg_info[count].fs_44kHz) {
+ /* 44.1 khz - 12 MHz Mclk */
+ tsc2101_audio_write(TSC2101_PLL_PROG_1, PLL1_PLLSEL |
+ PLL1_PVAL(1) | PLL1_I_VAL(7)); /* PVAL 1; I_VAL 7 */
+ tsc2101_audio_write(TSC2101_PLL_PROG_2, PLL2_D_VAL(0x1490)); /* D_VAL 5264 */
+ } else {
+ /* 48 khz - 12 Mhz Mclk */
+ tsc2101_audio_write(TSC2101_PLL_PROG_1, PLL1_PLLSEL |
+ PLL1_PVAL(1) | PLL1_I_VAL(8)); /* PVAL 1; I_VAL 8 */
+ tsc2101_audio_write(TSC2101_PLL_PROG_2, PLL2_D_VAL(0x780)); /* D_VAL 1920 */
+ }
+
+ /* Set the sample rate */
+#ifndef TSC_MASTER
+ clkgdv = CODEC_CLOCK / (sample_rate * (DEFAULT_BITPERSAMPLE * 2 - 1));
+ if (clkgdv)
+ srgr1 = (FWID(DEFAULT_BITPERSAMPLE - 1) | CLKGDV(clkgdv));
+ else
+ return (1);
+
+ /* Stereo Mode */
+ srgr2 = (CLKSM | FSGM | FPER(DEFAULT_BITPERSAMPLE * 2 - 1));
+#else
+ srgr1 = (FWID(DEFAULT_BITPERSAMPLE - 1) | CLKGDV(clkgdv));
+ srgr2 = ((GSYNC | CLKSP | FSGM | FPER(DEFAULT_BITPERSAMPLE * 2 - 1)));
+
+#endif /* end of #ifdef TSC_MASTER */
+ OMAP_MCBSP_WRITE(OMAP1610_MCBSP1_BASE, SRGR2, srgr2);
+ OMAP_MCBSP_WRITE(OMAP1610_MCBSP1_BASE, SRGR1, srgr1);
+}
+
+void tsc2101_configure(void)
+{
+}
+
+/*
+ * Omap MCBSP clock and Power Management configuration
+ *
+ * Here we have some functions that allows clock to be enabled and
+ * disabled only when needed. Besides doing clock configuration
+ * it allows turn on/turn off audio when necessary.
+ */
+
+/*
+ * Do clock framework mclk search
+ */
+void tsc2101_clock_setup(void)
+{
+ tsc2101_mclk = clk_get(0, "mclk");
+}
+
+/*
+ * Do some sanity check, set clock rate, starts it and turn codec audio on
+ */
+int tsc2101_clock_on(void)
+{
+ int curUseCount;
+ uint curRate;
+ int err;
+
+ curUseCount = clk_get_usecount(tsc2101_mclk);
+ DPRINTK("clock use count = %d\n", curUseCount);
+ if (curUseCount > 0) {
+ // MCLK is already in use
+ printk(KERN_WARNING
+ "MCLK already in use at %d Hz. We change it to %d Hz\n",
+ (uint) clk_get_rate(tsc2101_mclk),
+ CODEC_CLOCK);
+ }
+ curRate = (uint)clk_get_rate(tsc2101_mclk);
+ DPRINTK("old clock rate = %d\n", curRate);
+ if (curRate != CODEC_CLOCK) {
+ err = clk_set_rate(tsc2101_mclk, CODEC_CLOCK);
+ if (err) {
+ printk(KERN_WARNING
+ "Cannot set MCLK clock rate for TSC2101 CODEC, error code = %d\n", err);
+ //return -ECANCELED;
+ }
+ }
+ else
+ {
+ printk(KERN_INFO
+ "omap_alsa_tsc2101_clock_on(), no need to change rate, no need to change clock rate, rate already %d Hz.\n",
+ CODEC_CLOCK);
+ }
+ err = clk_enable(tsc2101_mclk);
+ curRate = (uint)clk_get_rate(tsc2101_mclk);
+ curUseCount = clk_get_usecount(tsc2101_mclk);
+ DPRINTK("MCLK = %d [%d], usecount = %d, clk_enable retval = %d\n",
+ curRate,
+ CODEC_CLOCK,
+ curUseCount,
+ err);
+
+ // Now turn the audio on
+ omap_tsc2101_write(PAGE2_AUDIO_CODEC_REGISTERS,
+ TSC2101_CODEC_POWER_CTRL,
+ 0x0000);
+ return 0;
+}
+
+/*
+ * Do some sanity check, turn clock off and then turn
+ * codec audio off
+ */
+int tsc2101_clock_off(void)
+{
+ int curUseCount;
+ int curRate;
+
+ curUseCount = clk_get_usecount(tsc2101_mclk);
+ DPRINTK("clock use count = %d\n", curUseCount);
+ if (curUseCount > 0) {
+ curRate = clk_get_rate(tsc2101_mclk);
+ DPRINTK("clock rate = %d\n", curRate);
+ if (curRate != CODEC_CLOCK) {
+ printk(KERN_WARNING
+ "MCLK for audio should be %d Hz. But is %d Hz\n",
+ (uint) clk_get_rate(tsc2101_mclk),
+ CODEC_CLOCK);
+ }
+ clk_disable(tsc2101_mclk);
+ DPRINTK("clock disabled\n");
+ }
+ tsc2101_audio_write(TSC2101_CODEC_POWER_CTRL,
+ ~(CPC_SP1PWDN | CPC_SP2PWDN | CPC_BASSBC));
+ DPRINTK("audio codec off\n");
+#ifdef DUMP_TSC2101_AUDIO_REGISTERS
+ printk("tsc2101_clock_off()\n");
+ dump_tsc2101_audio_reg();
+#endif
+ return 0;
+}
+
+int tsc2101_get_default_samplerate(void)
+{
+ return DEFAULT_SAMPLE_RATE;
+}
+
+static int __init snd_omap_alsa_tsc2101_probe(struct platform_device *pdev)
+{
+ int ret;
+ struct omap_alsa_codec_config *codec_cfg;
+
+ codec_cfg = pdev->dev.platform_data;
+ if (codec_cfg != NULL) {
+ codec_cfg->hw_constraints_rates = &tsc2101_hw_constraints_rates;
+ codec_cfg->snd_omap_alsa_playback = &tsc2101_snd_omap_alsa_playback;
+ codec_cfg->snd_omap_alsa_capture = &tsc2101_snd_omap_alsa_capture;
+ codec_cfg->codec_configure_dev = tsc2101_configure;
+ codec_cfg->codec_set_samplerate = tsc2101_set_samplerate;
+ codec_cfg->codec_clock_setup = tsc2101_clock_setup;
+ codec_cfg->codec_clock_on = tsc2101_clock_on;
+ codec_cfg->codec_clock_off = tsc2101_clock_off;
+ codec_cfg->get_default_samplerate = tsc2101_get_default_samplerate;
+ ret = snd_omap_alsa_post_probe(pdev, codec_cfg);
+ }
+ else
+ ret = -ENODEV;
+ return ret;
+}
+
+static struct platform_driver omap_alsa_driver = {
+ .probe = snd_omap_alsa_tsc2101_probe,
+ .remove = snd_omap_alsa_remove,
+ .suspend = snd_omap_alsa_suspend,
+ .resume = snd_omap_alsa_resume,
+ .driver = {
+ .name = "omap_alsa_mcbsp",
+ },
+};
+
+static int __init omap_alsa_tsc2101_init(void)
+{
+ int err;
+
+ ADEBUG();
+ err = platform_driver_register(&omap_alsa_driver);
+
+ return err;
+}
+
+static void __exit omap_alsa_tsc2101_exit(void)
+{
+ ADEBUG();
+ platform_driver_unregister(&omap_alsa_driver);
+}
+
+module_init(omap_alsa_tsc2101_init);
+module_exit(omap_alsa_tsc2101_exit);
projects / linux-2.6-omap-h63xx.git / commitdiff