mirror of
git://projects.qi-hardware.com/openwrt-xburst.git
synced 2024-11-29 00:05:54 +02:00
dc3d3f1c49
it's basically also provided by ingenic and nativly based on 2.6.27, adjusted to fit into the OpenWrt-environment
406 lines
11 KiB
C
Executable File
406 lines
11 KiB
C
Executable File
/*
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* apus.c -- SoC audio for APUS
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*
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* This program is free software; you can redistribute it and/or modify it
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* under the terms of the GNU General Public License as published by the
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* Free Software Foundation; either version 2 of the License, or (at your
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* option) any later version.
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*
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*/
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#include <linux/module.h>
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#include <linux/moduleparam.h>
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#include <linux/timer.h>
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#include <linux/interrupt.h>
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#include <linux/platform_device.h>
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#include <sound/driver.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 <sound/soc-dapm.h>
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#include <sound/tlv.h>
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#include "../codecs/jzdlv.h"
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#include "jz4750-pcm.h"
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#include "jz4750-i2s.h"
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//static struct snd_soc_machine apus;
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#define APUS_HP 0
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#define APUS_MIC 1
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#define APUS_LINE 2
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#define APUS_HEADSET 3
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#define APUS_HP_OFF 4
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#define APUS_SPK_ON 0
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#define APUS_SPK_OFF 1
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static int apus_jack_func;
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static int apus_spk_func;
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unsigned short set_scoop_gpio(struct device *dev, unsigned short bit)
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{
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unsigned short gpio_bit = 0;
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return gpio_bit;
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}
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unsigned short reset_scoop_gpio(struct device *dev, unsigned short bit)
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{
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unsigned short gpio_bit = 0;
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return gpio_bit;
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}
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static void apus_ext_control(struct snd_soc_codec *codec)
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{
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int spk = 0, mic = 0, line = 0, hp = 0, hs = 0;
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/* set up jack connection */
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switch (apus_jack_func) {
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case APUS_HP:
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hp = 1;
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snd_soc_dapm_disable_pin(codec, "Mic Jack");
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snd_soc_dapm_disable_pin(codec, "Line Jack");
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snd_soc_dapm_enable_pin(codec, "Headphone Jack");
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snd_soc_dapm_disable_pin(codec, "Headset Jack");
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break;
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case APUS_MIC:
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mic = 1;
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snd_soc_dapm_enable_pin(codec, "Mic Jack");
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snd_soc_dapm_disable_pin(codec, "Line Jack");
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snd_soc_dapm_disable_pin(codec, "Headphone Jack");
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snd_soc_dapm_disable_pin(codec, "Headset Jack");
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break;
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case APUS_LINE:
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line = 1;
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snd_soc_dapm_disable_pin(codec, "Mic Jack");
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snd_soc_dapm_enable_pin(codec, "Line Jack");
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snd_soc_dapm_disable_pin(codec, "Headphone Jack");
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snd_soc_dapm_disable_pin(codec, "Headset Jack");
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break;
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case APUS_HEADSET:
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hs = 1;
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mic = 1;
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snd_soc_dapm_enable_pin(codec, "Mic Jack");
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snd_soc_dapm_disable_pin(codec, "Line Jack");
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snd_soc_dapm_disable_pin(codec, "Headphone Jack");
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snd_soc_dapm_enable_pin(codec, "Headset Jack");
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break;
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}
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if (apus_spk_func == APUS_SPK_ON)
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spk = 1;
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if (apus_spk_func == APUS_SPK_ON)
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snd_soc_dapm_enable_pin(codec, "Ext Spk");
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else
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snd_soc_dapm_disable_pin(codec, "Ext Spk");
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/* signal a DAPM event */
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snd_soc_dapm_sync(codec);
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}
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static int apus_startup(struct snd_pcm_substream *substream)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_codec *codec = rtd->socdev->codec;
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/* check the jack status at stream startup */
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apus_ext_control(codec);
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return 0;
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}
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/* we need to unmute the HP at shutdown as the mute burns power on apus */
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static int apus_shutdown(struct snd_pcm_substream *substream)
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{
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/*struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_codec *codec = rtd->socdev->codec;*/
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return 0;
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}
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static int apus_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->dai->codec_dai;
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struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
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int ret = 0;
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/* set codec DAI configuration */
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/*ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
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SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);*/
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ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
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SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
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if (ret < 0)
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return ret;
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/* set cpu DAI configuration */
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/*ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
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SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);*/
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ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
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SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
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if (ret < 0)
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return ret;
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/* set the codec system clock for DAC and ADC */
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/*ret = codec_dai->dai_ops.set_sysclk(codec_dai, JZDLV_SYSCLK, 111,
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SND_SOC_CLOCK_IN);*/
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ret = snd_soc_dai_set_sysclk(codec_dai, JZDLV_SYSCLK, 111,
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SND_SOC_CLOCK_IN);
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if (ret < 0)
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return ret;
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/* set the I2S system clock as input (unused) */
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/*ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, JZ4750_I2S_SYSCLK, 0,
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SND_SOC_CLOCK_IN);*/
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ret = snd_soc_dai_set_sysclk(cpu_dai, JZ4750_I2S_SYSCLK, 0,
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SND_SOC_CLOCK_IN);
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if (ret < 0)
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return ret;
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return 0;
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}
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static struct snd_soc_ops apus_ops = {
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.startup = apus_startup,
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.hw_params = apus_hw_params,
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.shutdown = apus_shutdown,
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};
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static int apus_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] = apus_jack_func;
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return 0;
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}
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static int apus_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_codec *codec = snd_kcontrol_chip(kcontrol);
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if (apus_jack_func == ucontrol->value.integer.value[0])
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return 0;
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apus_jack_func = ucontrol->value.integer.value[0];
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apus_ext_control(codec);
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return 1;
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}
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static int apus_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] = apus_spk_func;
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return 0;
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}
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static int apus_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_codec *codec = snd_kcontrol_chip(kcontrol);
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if (apus_spk_func == ucontrol->value.integer.value[0])
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return 0;
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apus_spk_func = ucontrol->value.integer.value[0];
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apus_ext_control(codec);
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return 1;
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}
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static int apus_amp_event(struct snd_soc_dapm_widget *w, 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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//set_scoop_gpio(&corgiscoop_device.dev, APUS_SCP_APM_ON);
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;
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else
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//reset_scoop_gpio(&corgiscoop_device.dev, APUS_SCP_APM_ON);
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;
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return 0;
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}
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static int apus_mic_event(struct snd_soc_dapm_widget *w, 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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//set_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS);
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;
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else
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//reset_scoop_gpio(&corgiscoop_device.dev, CORGI_SCP_MIC_BIAS);
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;
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return 0;
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}
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static int jzdlv_get_reg(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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//int reg = kcontrol->private_value & 0xFF;
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//int shift = (kcontrol->private_value >> 8) & 0x0F;
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//int mask = (kcontrol->private_value >> 16) & 0xFF;
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//ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask;
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return 0;
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}
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static int jzdlv_set_reg(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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//int reg = kcontrol->private_value & 0xFF;
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//int shift = (kcontrol->private_value >> 8) & 0x0F;
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//int mask = (kcontrol->private_value >> 16) & 0xFF;
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/*if (((lm4857_regs[reg] >> shift) & mask) ==
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ucontrol->value.integer.value[0])
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return 0;
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lm4857_regs[reg] &= ~(mask << shift);
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lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift;
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lm4857_write_regs();*/
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return 1;
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}
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/* apus machine dapm widgets */
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static const struct snd_soc_dapm_widget jzdlv_dapm_widgets[] = {
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SND_SOC_DAPM_HP("Headphone Jack", NULL),
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SND_SOC_DAPM_MIC("Mic Jack",apus_mic_event),
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SND_SOC_DAPM_SPK("Ext Spk", apus_amp_event),
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SND_SOC_DAPM_LINE("Line Jack", NULL),
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SND_SOC_DAPM_HP("Headset Jack", NULL),
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};
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/* apus machine audio map (connections to the codec pins) */
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static const struct snd_soc_dapm_route audio_map[] = {
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/* headset Jack - in = micin, out = LHPOUT*/
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{"Headset Jack", NULL, "LHPOUT"},
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/* headphone connected to LHPOUT1, RHPOUT1 */
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{"Headphone Jack", NULL, "LHPOUT"},
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{"Headphone Jack", NULL, "RHPOUT"},
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/* speaker connected to LOUT, ROUT */
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{"Ext Spk", NULL, "ROUT"},
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{"Ext Spk", NULL, "LOUT"},
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/* mic is connected to MICIN (via right channel of headphone jack) */
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{"MICIN", NULL, "Mic Jack"},
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/* Same as the above but no mic bias for line signals */
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{"MICIN", NULL, "Line Jack"},
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};
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static const char *jack_function[] = {"Headphone", "Mic", "Line", "Headset",
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"Off"};
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static const char *spk_function[] = {"On", "Off"};
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static const struct soc_enum apus_enum[] = {
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SOC_ENUM_SINGLE_EXT(5, jack_function),
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SOC_ENUM_SINGLE_EXT(2, spk_function),
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};
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static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0);
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//static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0);
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static const struct snd_kcontrol_new jzdlv_apus_controls[] = {
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/* SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", 1, 0, 31, 0,
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jzdlv_get_reg, jzdlv_set_reg, stereo_tlv),*/
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SOC_SINGLE_EXT_TLV("PCM", 1, 0, 31, 0,
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jzdlv_get_reg, jzdlv_set_reg, stereo_tlv),
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SOC_SINGLE_EXT_TLV("Capture", 2, 0, 31, 0,
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jzdlv_get_reg, jzdlv_set_reg, stereo_tlv),
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/* SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0,
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lm4857_get_reg, lm4857_set_reg, mono_tlv),*/
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SOC_ENUM_EXT("Jack Function", apus_enum[0], apus_get_jack,apus_set_jack),
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SOC_ENUM_EXT("Speaker Function", apus_enum[1], apus_get_spk,apus_set_spk),
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};
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/*
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* Apus for a jzdlv as connected on jz4750 Device
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*/
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static int apus_jzdlv_init(struct snd_soc_codec *codec)
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{
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int i, err;
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snd_soc_dapm_disable_pin(codec, "LLINEIN");
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snd_soc_dapm_disable_pin(codec, "RLINEIN");
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/* Add apus specific controls */
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for (i = 0; i < ARRAY_SIZE(jzdlv_apus_controls); i++) {
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err = snd_ctl_add(codec->card,
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snd_soc_cnew(&jzdlv_apus_controls[i],codec, NULL));
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if (err < 0)
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return err;
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}
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/* Add apus specific widgets */
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snd_soc_dapm_new_controls(codec, jzdlv_dapm_widgets,
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ARRAY_SIZE(jzdlv_dapm_widgets));
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/* Set up apus specific audio path audio_map */
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snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
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snd_soc_dapm_sync(codec);
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return 0;
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}
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/* apus digital audio interface glue - connects codec <--> CPU */
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static struct snd_soc_dai_link apus_dai = {
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.name = "JZDLV",
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.stream_name = "JZDLV",
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.cpu_dai = &jz4750_i2s_dai,
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.codec_dai = &jzdlv_dai,
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.init = apus_jzdlv_init,
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.ops = &apus_ops,
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};
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/* apus audio machine driver */
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static struct snd_soc_machine snd_soc_machine_apus = {
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.name = "Apus",
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.dai_link = &apus_dai,
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.num_links = 1,
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};
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/* apus audio subsystem */
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static struct snd_soc_device apus_snd_devdata = {
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.machine = &snd_soc_machine_apus,
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.platform = &jz4750_soc_platform,
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.codec_dev = &soc_codec_dev_jzdlv,
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//.codec_data
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};
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static struct platform_device *apus_snd_device;
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static int __init apus_init(void)
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{
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int ret;
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apus_snd_device = platform_device_alloc("soc-audio", -1);
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if (!apus_snd_device)
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return -ENOMEM;
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platform_set_drvdata(apus_snd_device, &apus_snd_devdata);
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apus_snd_devdata.dev = &apus_snd_device->dev;
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ret = platform_device_add(apus_snd_device);
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if (ret)
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platform_device_put(apus_snd_device);
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return ret;
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}
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static void __exit apus_exit(void)
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{
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platform_device_unregister(apus_snd_device);
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}
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module_init(apus_init);
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module_exit(apus_exit);
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/* Module information */
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MODULE_AUTHOR("Richard");
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MODULE_DESCRIPTION("ALSA SoC Apus");
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MODULE_LICENSE("GPL");
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