init android origin source code

1 files changed, 563 insertions, 0 deletions

diff --git a/ANDROID_3.4.5/sound/soc/pxa/magician.c b/ANDROID_3.4.5/sound/soc/pxa/magician.c
new file mode 100644
index 00000000..aace19e0
--- /dev/null
+++ b/ANDROID_3.4.5/sound/soc/pxa/magician.c
@@ -0,0 +1,563 @@
+/*
+ * SoC audio for HTC Magician
+ *
+ * Copyright (c) 2006 Philipp Zabel <philipp.zabel@gmail.com>
+ *
+ * based on spitz.c,
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
+ *          Richard Purdie <richard@openedhand.com>
+ *
+ *  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/module.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/i2c.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/uda1380.h>
+
+#include <mach/magician.h>
+#include <asm/mach-types.h>
+#include "../codecs/uda1380.h"
+#include "pxa2xx-i2s.h"
+#include "pxa-ssp.h"
+
+#define MAGICIAN_MIC       0
+#define MAGICIAN_MIC_EXT   1
+
+static int magician_hp_switch;
+static int magician_spk_switch = 1;
+static int magician_in_sel = MAGICIAN_MIC;
+
+static void magician_ext_control(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	if (magician_spk_switch)
+		snd_soc_dapm_enable_pin(dapm, "Speaker");
+	else
+		snd_soc_dapm_disable_pin(dapm, "Speaker");
+	if (magician_hp_switch)
+		snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
+	else
+		snd_soc_dapm_disable_pin(dapm, "Headphone Jack");
+
+	switch (magician_in_sel) {
+	case MAGICIAN_MIC:
+		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+		snd_soc_dapm_enable_pin(dapm, "Call Mic");
+		break;
+	case MAGICIAN_MIC_EXT:
+		snd_soc_dapm_disable_pin(dapm, "Call Mic");
+		snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+		break;
+	}
+
+	snd_soc_dapm_sync(dapm);
+}
+
+static int magician_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+
+	mutex_lock(&codec->mutex);
+
+	/* check the jack status at stream startup */
+	magician_ext_control(codec);
+
+	mutex_unlock(&codec->mutex);
+
+	return 0;
+}
+
+/*
+ * Magician uses SSP port for playback.
+ */
+static int magician_playback_hw_params(struct snd_pcm_substream *substream,
+				       struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	unsigned int acps, acds, width;
+	unsigned int div4 = PXA_SSP_CLK_SCDB_4;
+	int ret = 0;
+
+	width = snd_pcm_format_physical_width(params_format(params));
+
+	/*
+	 * rate = SSPSCLK / (2 * width(16 or 32))
+	 * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
+	 */
+	switch (params_rate(params)) {
+	case 8000:
+		/* off by a factor of 2: bug in the PXA27x audio clock? */
+		acps = 32842000;
+		switch (width) {
+		case 16:
+			/* 513156 Hz ~= _2_ * 8000 Hz * 32 (+0.23%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_16;
+			break;
+		default: /* 32 */
+			/* 1026312 Hz ~= _2_ * 8000 Hz * 64 (+0.23%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_8;
+		}
+		break;
+	case 11025:
+		acps = 5622000;
+		switch (width) {
+		case 16:
+			/* 351375 Hz ~= 11025 Hz * 32 (-0.41%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_4;
+			break;
+		default: /* 32 */
+			/* 702750 Hz ~= 11025 Hz * 64 (-0.41%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_2;
+		}
+		break;
+	case 22050:
+		acps = 5622000;
+		switch (width) {
+		case 16:
+			/* 702750 Hz ~= 22050 Hz * 32 (-0.41%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_2;
+			break;
+		default: /* 32 */
+			/* 1405500 Hz ~= 22050 Hz * 64 (-0.41%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_1;
+		}
+		break;
+	case 44100:
+		acps = 5622000;
+		switch (width) {
+		case 16:
+			/* 1405500 Hz ~= 44100 Hz * 32 (-0.41%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_2;
+			break;
+		default: /* 32 */
+			/* 2811000 Hz ~= 44100 Hz * 64 (-0.41%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_1;
+		}
+		break;
+	case 48000:
+		acps = 12235000;
+		switch (width) {
+		case 16:
+			/* 1529375 Hz ~= 48000 Hz * 32 (-0.44%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_2;
+			break;
+		default: /* 32 */
+			/* 3058750 Hz ~= 48000 Hz * 64 (-0.44%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_1;
+		}
+		break;
+	case 96000:
+	default:
+		acps = 12235000;
+		switch (width) {
+		case 16:
+			/* 3058750 Hz ~= 96000 Hz * 32 (-0.44%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_1;
+			break;
+		default: /* 32 */
+			/* 6117500 Hz ~= 96000 Hz * 64 (-0.44%) */
+			acds = PXA_SSP_CLK_AUDIO_DIV_2;
+			div4 = PXA_SSP_CLK_SCDB_1;
+			break;
+		}
+		break;
+	}
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+			SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width);
+	if (ret < 0)
+		return ret;
+
+	/* set audio clock as clock source */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0,
+			SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	/* set the SSP audio system clock ACDS divider */
+	ret = snd_soc_dai_set_clkdiv(cpu_dai,
+			PXA_SSP_AUDIO_DIV_ACDS, acds);
+	if (ret < 0)
+		return ret;
+
+	/* set the SSP audio system clock SCDB divider4 */
+	ret = snd_soc_dai_set_clkdiv(cpu_dai,
+			PXA_SSP_AUDIO_DIV_SCDB, div4);
+	if (ret < 0)
+		return ret;
+
+	/* set SSP audio pll clock */
+	ret = snd_soc_dai_set_pll(cpu_dai, 0, 0, 0, acps);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+/*
+ * Magician uses I2S for capture.
+ */
+static int magician_capture_hw_params(struct snd_pcm_substream *substream,
+				      struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret = 0;
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+			SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+			SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set the I2S system clock as output */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+			SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops magician_capture_ops = {
+	.startup = magician_startup,
+	.hw_params = magician_capture_hw_params,
+};
+
+static struct snd_soc_ops magician_playback_ops = {
+	.startup = magician_startup,
+	.hw_params = magician_playback_hw_params,
+};
+
+static int magician_get_hp(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = magician_hp_switch;
+	return 0;
+}
+
+static int magician_set_hp(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (magician_hp_switch == ucontrol->value.integer.value[0])
+		return 0;
+
+	magician_hp_switch = ucontrol->value.integer.value[0];
+	magician_ext_control(codec);
+	return 1;
+}
+
+static int magician_get_spk(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = magician_spk_switch;
+	return 0;
+}
+
+static int magician_set_spk(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (magician_spk_switch == ucontrol->value.integer.value[0])
+		return 0;
+
+	magician_spk_switch = ucontrol->value.integer.value[0];
+	magician_ext_control(codec);
+	return 1;
+}
+
+static int magician_get_input(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = magician_in_sel;
+	return 0;
+}
+
+static int magician_set_input(struct snd_kcontrol *kcontrol,
+			      struct snd_ctl_elem_value *ucontrol)
+{
+	if (magician_in_sel == ucontrol->value.integer.value[0])
+		return 0;
+
+	magician_in_sel = ucontrol->value.integer.value[0];
+
+	switch (magician_in_sel) {
+	case MAGICIAN_MIC:
+		gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 1);
+		break;
+	case MAGICIAN_MIC_EXT:
+		gpio_set_value(EGPIO_MAGICIAN_IN_SEL1, 0);
+	}
+
+	return 1;
+}
+
+static int magician_spk_power(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *k, int event)
+{
+	gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+static int magician_hp_power(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *k, int event)
+{
+	gpio_set_value(EGPIO_MAGICIAN_EP_POWER, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+static int magician_mic_bias(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *k, int event)
+{
+	gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+/* magician machine dapm widgets */
+static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", magician_hp_power),
+	SND_SOC_DAPM_SPK("Speaker", magician_spk_power),
+	SND_SOC_DAPM_MIC("Call Mic", magician_mic_bias),
+	SND_SOC_DAPM_MIC("Headset Mic", magician_mic_bias),
+};
+
+/* magician machine audio_map */
+static const struct snd_soc_dapm_route audio_map[] = {
+
+	/* Headphone connected to VOUTL, VOUTR */
+	{"Headphone Jack", NULL, "VOUTL"},
+	{"Headphone Jack", NULL, "VOUTR"},
+
+	/* Speaker connected to VOUTL, VOUTR */
+	{"Speaker", NULL, "VOUTL"},
+	{"Speaker", NULL, "VOUTR"},
+
+	/* Mics are connected to VINM */
+	{"VINM", NULL, "Headset Mic"},
+	{"VINM", NULL, "Call Mic"},
+};
+
+static const char *input_select[] = {"Call Mic", "Headset Mic"};
+static const struct soc_enum magician_in_sel_enum =
+	SOC_ENUM_SINGLE_EXT(2, input_select);
+
+static const struct snd_kcontrol_new uda1380_magician_controls[] = {
+	SOC_SINGLE_BOOL_EXT("Headphone Switch",
+			(unsigned long)&magician_hp_switch,
+			magician_get_hp, magician_set_hp),
+	SOC_SINGLE_BOOL_EXT("Speaker Switch",
+			(unsigned long)&magician_spk_switch,
+			magician_get_spk, magician_set_spk),
+	SOC_ENUM_EXT("Input Select", magician_in_sel_enum,
+			magician_get_input, magician_set_input),
+};
+
+/*
+ * Logic for a uda1380 as connected on a HTC Magician
+ */
+static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int err;
+
+	/* NC codec pins */
+	snd_soc_dapm_nc_pin(dapm, "VOUTLHP");
+	snd_soc_dapm_nc_pin(dapm, "VOUTRHP");
+
+	/* FIXME: is anything connected here? */
+	snd_soc_dapm_nc_pin(dapm, "VINL");
+	snd_soc_dapm_nc_pin(dapm, "VINR");
+
+	/* Add magician specific controls */
+	err = snd_soc_add_codec_controls(codec, uda1380_magician_controls,
+				ARRAY_SIZE(uda1380_magician_controls));
+	if (err < 0)
+		return err;
+
+	/* Add magician specific widgets */
+	snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
+				  ARRAY_SIZE(uda1380_dapm_widgets));
+
+	/* Set up magician specific audio path interconnects */
+	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
+
+	return 0;
+}
+
+/* magician digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link magician_dai[] = {
+{
+	.name = "uda1380",
+	.stream_name = "UDA1380 Playback",
+	.cpu_dai_name = "pxa-ssp-dai.0",
+	.codec_dai_name = "uda1380-hifi-playback",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "uda1380-codec.0-0018",
+	.init = magician_uda1380_init,
+	.ops = &magician_playback_ops,
+},
+{
+	.name = "uda1380",
+	.stream_name = "UDA1380 Capture",
+	.cpu_dai_name = "pxa2xx-i2s",
+	.codec_dai_name = "uda1380-hifi-capture",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "uda1380-codec.0-0018",
+	.ops = &magician_capture_ops,
+}
+};
+
+/* magician audio machine driver */
+static struct snd_soc_card snd_soc_card_magician = {
+	.name = "Magician",
+	.owner = THIS_MODULE,
+	.dai_link = magician_dai,
+	.num_links = ARRAY_SIZE(magician_dai),
+
+};
+
+static struct platform_device *magician_snd_device;
+
+/*
+ * FIXME: move into magician board file once merged into the pxa tree
+ */
+static struct uda1380_platform_data uda1380_info = {
+	.gpio_power = EGPIO_MAGICIAN_CODEC_POWER,
+	.gpio_reset = EGPIO_MAGICIAN_CODEC_RESET,
+	.dac_clk    = UDA1380_DAC_CLK_WSPLL,
+};
+
+static struct i2c_board_info i2c_board_info[] = {
+	{
+		I2C_BOARD_INFO("uda1380", 0x18),
+		.platform_data = &uda1380_info,
+	},
+};
+
+static int __init magician_init(void)
+{
+	int ret;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
+
+	if (!machine_is_magician())
+		return -ENODEV;
+
+	adapter = i2c_get_adapter(0);
+	if (!adapter)
+		return -ENODEV;
+	client = i2c_new_device(adapter, i2c_board_info);
+	i2c_put_adapter(adapter);
+	if (!client)
+		return -ENODEV;
+
+	ret = gpio_request(EGPIO_MAGICIAN_SPK_POWER, "SPK_POWER");
+	if (ret)
+		goto err_request_spk;
+	ret = gpio_request(EGPIO_MAGICIAN_EP_POWER, "EP_POWER");
+	if (ret)
+		goto err_request_ep;
+	ret = gpio_request(EGPIO_MAGICIAN_MIC_POWER, "MIC_POWER");
+	if (ret)
+		goto err_request_mic;
+	ret = gpio_request(EGPIO_MAGICIAN_IN_SEL0, "IN_SEL0");
+	if (ret)
+		goto err_request_in_sel0;
+	ret = gpio_request(EGPIO_MAGICIAN_IN_SEL1, "IN_SEL1");
+	if (ret)
+		goto err_request_in_sel1;
+
+	gpio_set_value(EGPIO_MAGICIAN_IN_SEL0, 0);
+
+	magician_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!magician_snd_device) {
+		ret = -ENOMEM;
+		goto err_pdev;
+	}
+
+	platform_set_drvdata(magician_snd_device, &snd_soc_card_magician);
+	ret = platform_device_add(magician_snd_device);
+	if (ret) {
+		platform_device_put(magician_snd_device);
+		goto err_pdev;
+	}
+
+	return 0;
+
+err_pdev:
+	gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+err_request_in_sel1:
+	gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+err_request_in_sel0:
+	gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+err_request_mic:
+	gpio_free(EGPIO_MAGICIAN_EP_POWER);
+err_request_ep:
+	gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+err_request_spk:
+	return ret;
+}
+
+static void __exit magician_exit(void)
+{
+	platform_device_unregister(magician_snd_device);
+
+	gpio_set_value(EGPIO_MAGICIAN_SPK_POWER, 0);
+	gpio_set_value(EGPIO_MAGICIAN_EP_POWER, 0);
+	gpio_set_value(EGPIO_MAGICIAN_MIC_POWER, 0);
+
+	gpio_free(EGPIO_MAGICIAN_IN_SEL1);
+	gpio_free(EGPIO_MAGICIAN_IN_SEL0);
+	gpio_free(EGPIO_MAGICIAN_MIC_POWER);
+	gpio_free(EGPIO_MAGICIAN_EP_POWER);
+	gpio_free(EGPIO_MAGICIAN_SPK_POWER);
+}
+
+module_init(magician_init);
+module_exit(magician_exit);
+
+MODULE_AUTHOR("Philipp Zabel");
+MODULE_DESCRIPTION("ALSA SoC Magician");
+MODULE_LICENSE("GPL");