[Feature]add MT2731_MP2_MR2_SVN388 baseline version
Change-Id: Ief04314834b31e27effab435d3ca8ba33b499059
diff --git a/src/kernel/linux/v4.14/sound/soc/pxa/magician.c b/src/kernel/linux/v4.14/sound/soc/pxa/magician.c
new file mode 100644
index 0000000..2fc012b
--- /dev/null
+++ b/src/kernel/linux/v4.14/sound/soc/pxa/magician.c
@@ -0,0 +1,534 @@
+/*
+ * 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_dapm_context *dapm)
+{
+
+ snd_soc_dapm_mutex_lock(dapm);
+
+ if (magician_spk_switch)
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
+ else
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
+ if (magician_hp_switch)
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+ else
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+
+ switch (magician_in_sel) {
+ case MAGICIAN_MIC:
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Mic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Call Mic");
+ break;
+ case MAGICIAN_MIC_EXT:
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Call Mic");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Mic");
+ break;
+ }
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
+}
+
+static int magician_startup(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+ /* check the jack status at stream startup */
+ magician_ext_control(&rtd->card->dapm);
+
+ 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 const struct snd_soc_ops magician_capture_ops = {
+ .startup = magician_startup,
+ .hw_params = magician_capture_hw_params,
+};
+
+static const 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_card *card = 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(&card->dapm);
+ 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_card *card = 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(&card->dapm);
+ return 1;
+}
+
+static int magician_get_input(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.enumerated.item[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.enumerated.item[0])
+ return 0;
+
+ magician_in_sel = ucontrol->value.enumerated.item[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 * const 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),
+};
+
+/* 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",
+ .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),
+
+ .controls = uda1380_magician_controls,
+ .num_controls = ARRAY_SIZE(uda1380_magician_controls),
+ .dapm_widgets = uda1380_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
+ .dapm_routes = audio_map,
+ .num_dapm_routes = ARRAY_SIZE(audio_map),
+ .fully_routed = true,
+};
+
+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");