e2e8bfdf61
Convert MicBias widgets to supply widget. On tlv320aic3x, Mic bias power on/off shares the same register bits with output mic bias voltage. So, when power on mic bias, we need reclaim it to voltage value. Provide a new platform data so that the micbias voltage can be sent according to board requirement. Now since tlv320aic3x codec driver is DT aware, update dt files and functions to handle this new "micbias-vg" platform data. Because of sharing of bits, when enabling the micbias, voltage also needs to be updated. So use SND_SOC_DAPM_POST_PMU & SND_SOC_DAPM_PRE_PMD macro to create an event to handle this. Since micbias is converted to supply widget, updated machine drivers as well. This change is runtime tested on da850-evm with audio loopback (arecord|aplay) for confirmation. Signed-off-by: Hebbar Gururaja <gururaja.hebbar@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
384 lines
9.5 KiB
C
384 lines
9.5 KiB
C
/*
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* n810.c -- SoC audio for Nokia N810
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*
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* Copyright (C) 2008 Nokia Corporation
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*
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* Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* version 2 as published by the Free Software Foundation.
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*
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* This program is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
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* 02110-1301 USA
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*
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*/
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#include <linux/clk.h>
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#include <linux/i2c.h>
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#include <linux/platform_device.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/soc.h>
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#include <asm/mach-types.h>
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#include <linux/gpio.h>
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#include <linux/module.h>
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#include <linux/platform_data/asoc-ti-mcbsp.h>
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#include "omap-mcbsp.h"
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#include "omap-pcm.h"
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#define N810_HEADSET_AMP_GPIO 10
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#define N810_SPEAKER_AMP_GPIO 101
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enum {
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N810_JACK_DISABLED,
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N810_JACK_HP,
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N810_JACK_HS,
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N810_JACK_MIC,
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};
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static struct clk *sys_clkout2;
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static struct clk *sys_clkout2_src;
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static struct clk *func96m_clk;
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static int n810_spk_func;
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static int n810_jack_func;
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static int n810_dmic_func;
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static void n810_ext_control(struct snd_soc_dapm_context *dapm)
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{
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int hp = 0, line1l = 0;
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switch (n810_jack_func) {
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case N810_JACK_HS:
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line1l = 1;
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case N810_JACK_HP:
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hp = 1;
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break;
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case N810_JACK_MIC:
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line1l = 1;
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break;
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}
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if (n810_spk_func)
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snd_soc_dapm_enable_pin(dapm, "Ext Spk");
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else
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snd_soc_dapm_disable_pin(dapm, "Ext Spk");
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if (hp)
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snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
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else
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snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
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if (line1l)
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snd_soc_dapm_enable_pin(dapm, "LINE1L");
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else
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snd_soc_dapm_disable_pin(dapm, "LINE1L");
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if (n810_dmic_func)
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snd_soc_dapm_enable_pin(dapm, "DMic");
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else
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snd_soc_dapm_disable_pin(dapm, "DMic");
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snd_soc_dapm_sync(dapm);
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}
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static int n810_startup(struct snd_pcm_substream *substream)
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{
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struct snd_pcm_runtime *runtime = substream->runtime;
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_codec *codec = rtd->codec;
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snd_pcm_hw_constraint_minmax(runtime,
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SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
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n810_ext_control(&codec->dapm);
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return clk_enable(sys_clkout2);
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}
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static void n810_shutdown(struct snd_pcm_substream *substream)
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{
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clk_disable(sys_clkout2);
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}
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static int n810_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_dai *codec_dai = rtd->codec_dai;
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int err;
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/* Set the codec system clock for DAC and ADC */
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err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000,
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SND_SOC_CLOCK_IN);
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return err;
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}
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static struct snd_soc_ops n810_ops = {
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.startup = n810_startup,
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.hw_params = n810_hw_params,
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.shutdown = n810_shutdown,
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};
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static int n810_get_spk(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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ucontrol->value.integer.value[0] = n810_spk_func;
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return 0;
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}
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static int n810_set_spk(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
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if (n810_spk_func == ucontrol->value.integer.value[0])
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return 0;
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n810_spk_func = ucontrol->value.integer.value[0];
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n810_ext_control(&card->dapm);
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return 1;
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}
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static int n810_get_jack(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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ucontrol->value.integer.value[0] = n810_jack_func;
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return 0;
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}
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static int n810_set_jack(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
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if (n810_jack_func == ucontrol->value.integer.value[0])
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return 0;
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n810_jack_func = ucontrol->value.integer.value[0];
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n810_ext_control(&card->dapm);
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return 1;
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}
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static int n810_get_input(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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ucontrol->value.integer.value[0] = n810_dmic_func;
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return 0;
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}
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static int n810_set_input(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
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if (n810_dmic_func == ucontrol->value.integer.value[0])
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return 0;
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n810_dmic_func = ucontrol->value.integer.value[0];
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n810_ext_control(&card->dapm);
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return 1;
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}
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static int n810_spk_event(struct snd_soc_dapm_widget *w,
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struct snd_kcontrol *k, int event)
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{
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if (SND_SOC_DAPM_EVENT_ON(event))
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gpio_set_value(N810_SPEAKER_AMP_GPIO, 1);
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else
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gpio_set_value(N810_SPEAKER_AMP_GPIO, 0);
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return 0;
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}
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static int n810_jack_event(struct snd_soc_dapm_widget *w,
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struct snd_kcontrol *k, int event)
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{
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if (SND_SOC_DAPM_EVENT_ON(event))
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gpio_set_value(N810_HEADSET_AMP_GPIO, 1);
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else
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gpio_set_value(N810_HEADSET_AMP_GPIO, 0);
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return 0;
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}
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static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = {
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SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event),
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SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event),
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SND_SOC_DAPM_MIC("DMic", NULL),
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};
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static const struct snd_soc_dapm_route audio_map[] = {
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{"Headphone Jack", NULL, "HPLOUT"},
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{"Headphone Jack", NULL, "HPROUT"},
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{"Ext Spk", NULL, "LLOUT"},
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{"Ext Spk", NULL, "RLOUT"},
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{"DMic Rate 64", NULL, "Mic Bias"},
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{"Mic Bias", NULL, "DMic"},
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};
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static const char *spk_function[] = {"Off", "On"};
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static const char *jack_function[] = {"Off", "Headphone", "Headset", "Mic"};
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static const char *input_function[] = {"ADC", "Digital Mic"};
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static const struct soc_enum n810_enum[] = {
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SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
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SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
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SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
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};
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static const struct snd_kcontrol_new aic33_n810_controls[] = {
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SOC_ENUM_EXT("Speaker Function", n810_enum[0],
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n810_get_spk, n810_set_spk),
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SOC_ENUM_EXT("Jack Function", n810_enum[1],
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n810_get_jack, n810_set_jack),
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SOC_ENUM_EXT("Input Select", n810_enum[2],
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n810_get_input, n810_set_input),
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};
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static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd)
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{
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struct snd_soc_codec *codec = rtd->codec;
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struct snd_soc_dapm_context *dapm = &codec->dapm;
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/* Not connected */
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snd_soc_dapm_nc_pin(dapm, "MONO_LOUT");
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snd_soc_dapm_nc_pin(dapm, "HPLCOM");
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snd_soc_dapm_nc_pin(dapm, "HPRCOM");
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snd_soc_dapm_nc_pin(dapm, "MIC3L");
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snd_soc_dapm_nc_pin(dapm, "MIC3R");
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snd_soc_dapm_nc_pin(dapm, "LINE1R");
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snd_soc_dapm_nc_pin(dapm, "LINE2L");
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snd_soc_dapm_nc_pin(dapm, "LINE2R");
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return 0;
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}
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/* Digital audio interface glue - connects codec <--> CPU */
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static struct snd_soc_dai_link n810_dai = {
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.name = "TLV320AIC33",
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.stream_name = "AIC33",
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.cpu_dai_name = "omap-mcbsp.2",
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.platform_name = "omap-pcm-audio",
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.codec_name = "tlv320aic3x-codec.2-0018",
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.codec_dai_name = "tlv320aic3x-hifi",
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.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
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SND_SOC_DAIFMT_CBM_CFM,
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.init = n810_aic33_init,
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.ops = &n810_ops,
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};
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/* Audio machine driver */
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static struct snd_soc_card snd_soc_n810 = {
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.name = "N810",
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.owner = THIS_MODULE,
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.dai_link = &n810_dai,
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.num_links = 1,
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.controls = aic33_n810_controls,
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.num_controls = ARRAY_SIZE(aic33_n810_controls),
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.dapm_widgets = aic33_dapm_widgets,
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.num_dapm_widgets = ARRAY_SIZE(aic33_dapm_widgets),
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.dapm_routes = audio_map,
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.num_dapm_routes = ARRAY_SIZE(audio_map),
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};
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static struct platform_device *n810_snd_device;
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static int __init n810_soc_init(void)
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{
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int err;
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struct device *dev;
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if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax()))
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return -ENODEV;
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n810_snd_device = platform_device_alloc("soc-audio", -1);
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if (!n810_snd_device)
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return -ENOMEM;
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platform_set_drvdata(n810_snd_device, &snd_soc_n810);
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err = platform_device_add(n810_snd_device);
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if (err)
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goto err1;
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dev = &n810_snd_device->dev;
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sys_clkout2_src = clk_get(dev, "sys_clkout2_src");
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if (IS_ERR(sys_clkout2_src)) {
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dev_err(dev, "Could not get sys_clkout2_src clock\n");
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err = PTR_ERR(sys_clkout2_src);
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goto err2;
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}
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sys_clkout2 = clk_get(dev, "sys_clkout2");
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if (IS_ERR(sys_clkout2)) {
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dev_err(dev, "Could not get sys_clkout2\n");
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err = PTR_ERR(sys_clkout2);
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goto err3;
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}
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/*
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* Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use
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* 96 MHz as its parent in order to get 12 MHz
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*/
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func96m_clk = clk_get(dev, "func_96m_ck");
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if (IS_ERR(func96m_clk)) {
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dev_err(dev, "Could not get func 96M clock\n");
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err = PTR_ERR(func96m_clk);
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goto err4;
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}
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clk_set_parent(sys_clkout2_src, func96m_clk);
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clk_set_rate(sys_clkout2, 12000000);
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BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) ||
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(gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0));
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gpio_direction_output(N810_HEADSET_AMP_GPIO, 0);
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gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
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return 0;
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err4:
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clk_put(sys_clkout2);
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err3:
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clk_put(sys_clkout2_src);
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err2:
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platform_device_del(n810_snd_device);
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err1:
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platform_device_put(n810_snd_device);
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return err;
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}
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static void __exit n810_soc_exit(void)
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{
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gpio_free(N810_SPEAKER_AMP_GPIO);
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gpio_free(N810_HEADSET_AMP_GPIO);
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clk_put(sys_clkout2_src);
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clk_put(sys_clkout2);
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clk_put(func96m_clk);
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platform_device_unregister(n810_snd_device);
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}
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module_init(n810_soc_init);
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module_exit(n810_soc_exit);
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MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
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MODULE_DESCRIPTION("ALSA SoC Nokia N810");
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MODULE_LICENSE("GPL");
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