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-rw-r--r--sound/soc/intel/boards/Makefile15
-rw-r--r--sound/soc/intel/boards/broadwell.c292
-rw-r--r--sound/soc/intel/boards/byt-max98090.c187
-rw-r--r--sound/soc/intel/boards/byt-rt5640.c229
-rw-r--r--sound/soc/intel/boards/bytcr_rt5640.c227
-rw-r--r--sound/soc/intel/boards/cht_bsw_rt5645.c324
-rw-r--r--sound/soc/intel/boards/cht_bsw_rt5672.c366
-rw-r--r--sound/soc/intel/boards/haswell.c209
-rw-r--r--sound/soc/intel/boards/mfld_machine.c430
9 files changed, 2279 insertions, 0 deletions
diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile
new file mode 100644
index 000000000000..f8237f0044eb
--- /dev/null
+++ b/sound/soc/intel/boards/Makefile
@@ -0,0 +1,15 @@
+snd-soc-sst-haswell-objs := haswell.o
+snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
+snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
+snd-soc-sst-broadwell-objs := broadwell.o
+snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o
+snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
+snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o
+
+obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
+obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
+obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
+obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
+obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o
+obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o
+obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o
diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c
new file mode 100644
index 000000000000..8bafaf6ceab1
--- /dev/null
+++ b/sound/soc/intel/boards/broadwell.c
@@ -0,0 +1,292 @@
+/*
+ * Intel Broadwell Wildcatpoint SST Audio
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License version
+ * 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/pcm_params.h>
+
+#include "../common/sst-dsp.h"
+#include "../haswell/sst-haswell-ipc.h"
+
+#include "../../codecs/rt286.h"
+
+static struct snd_soc_jack broadwell_headset;
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin broadwell_headset_pins[] = {
+ {
+ .pin = "Mic Jack",
+ .mask = SND_JACK_MICROPHONE,
+ },
+ {
+ .pin = "Headphone Jack",
+ .mask = SND_JACK_HEADPHONE,
+ },
+};
+
+static const struct snd_kcontrol_new broadwell_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Speaker"),
+ SOC_DAPM_PIN_SWITCH("Headphone Jack"),
+};
+
+static const struct snd_soc_dapm_widget broadwell_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone Jack", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+ SND_SOC_DAPM_MIC("Mic Jack", NULL),
+ SND_SOC_DAPM_MIC("DMIC1", NULL),
+ SND_SOC_DAPM_MIC("DMIC2", NULL),
+ SND_SOC_DAPM_LINE("Line Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route broadwell_rt286_map[] = {
+
+ /* speaker */
+ {"Speaker", NULL, "SPOR"},
+ {"Speaker", NULL, "SPOL"},
+
+ /* HP jack connectors - unknown if we have jack deteck */
+ {"Headphone Jack", NULL, "HPO Pin"},
+
+ /* other jacks */
+ {"MIC1", NULL, "Mic Jack"},
+ {"LINE1", NULL, "Line Jack"},
+
+ /* digital mics */
+ {"DMIC1 Pin", NULL, "DMIC1"},
+ {"DMIC2 Pin", NULL, "DMIC2"},
+
+ /* CODEC BE connections */
+ {"SSP0 CODEC IN", NULL, "AIF1 Capture"},
+ {"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
+};
+
+static int broadwell_rt286_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_codec *codec = rtd->codec;
+ int ret = 0;
+ ret = snd_soc_card_jack_new(rtd->card, "Headset",
+ SND_JACK_HEADSET | SND_JACK_BTN_0, &broadwell_headset,
+ broadwell_headset_pins, ARRAY_SIZE(broadwell_headset_pins));
+ if (ret)
+ return ret;
+
+ rt286_mic_detect(codec, &broadwell_headset);
+ return 0;
+}
+
+
+static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* The ADSP will covert the FE rate to 48k, stereo */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set SSP0 to 16 bit */
+ params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
+ return 0;
+}
+
+static int broadwell_rt286_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;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000,
+ SND_SOC_CLOCK_IN);
+
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+ return ret;
+ }
+
+ return ret;
+}
+
+static struct snd_soc_ops broadwell_rt286_ops = {
+ .hw_params = broadwell_rt286_hw_params,
+};
+
+static int broadwell_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev);
+ struct sst_hsw *broadwell = pdata->dsp;
+ int ret;
+
+ /* Set ADSP SSP port settings */
+ ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0,
+ SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
+ SST_HSW_DEVICE_CLOCK_MASTER, 9);
+ if (ret < 0) {
+ dev_err(rtd->dev, "error: failed to set device config\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+/* broadwell digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link broadwell_rt286_dais[] = {
+ /* Front End DAI links */
+ {
+ .name = "System PCM",
+ .stream_name = "System Playback/Capture",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .init = broadwell_rtd_init,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+ {
+ .name = "Offload0",
+ .stream_name = "Offload0 Playback",
+ .cpu_dai_name = "Offload0 Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Offload1",
+ .stream_name = "Offload1 Playback",
+ .cpu_dai_name = "Offload1 Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Loopback PCM",
+ .stream_name = "Loopback",
+ .cpu_dai_name = "Loopback Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 0,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ },
+ /* Back End DAI links */
+ {
+ /* SSP0 - Codec */
+ .name = "Codec",
+ .be_id = 0,
+ .cpu_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "snd-soc-dummy",
+ .no_pcm = 1,
+ .codec_name = "i2c-INT343A:00",
+ .codec_dai_name = "rt286-aif1",
+ .init = broadwell_rt286_codec_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = broadwell_ssp0_fixup,
+ .ops = &broadwell_rt286_ops,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+};
+
+static int broadwell_suspend(struct snd_soc_card *card){
+ struct snd_soc_codec *codec;
+
+ list_for_each_entry(codec, &card->codec_dev_list, card_list) {
+ if (!strcmp(codec->component.name, "i2c-INT343A:00")) {
+ dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n");
+ rt286_mic_detect(codec, NULL);
+ break;
+ }
+ }
+ return 0;
+}
+
+static int broadwell_resume(struct snd_soc_card *card){
+ struct snd_soc_codec *codec;
+
+ list_for_each_entry(codec, &card->codec_dev_list, card_list) {
+ if (!strcmp(codec->component.name, "i2c-INT343A:00")) {
+ dev_dbg(codec->dev, "enabling jack detect for resume.\n");
+ rt286_mic_detect(codec, &broadwell_headset);
+ break;
+ }
+ }
+ return 0;
+}
+
+/* broadwell audio machine driver for WPT + RT286S */
+static struct snd_soc_card broadwell_rt286 = {
+ .name = "broadwell-rt286",
+ .owner = THIS_MODULE,
+ .dai_link = broadwell_rt286_dais,
+ .num_links = ARRAY_SIZE(broadwell_rt286_dais),
+ .controls = broadwell_controls,
+ .num_controls = ARRAY_SIZE(broadwell_controls),
+ .dapm_widgets = broadwell_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(broadwell_widgets),
+ .dapm_routes = broadwell_rt286_map,
+ .num_dapm_routes = ARRAY_SIZE(broadwell_rt286_map),
+ .fully_routed = true,
+ .suspend_pre = broadwell_suspend,
+ .resume_post = broadwell_resume,
+};
+
+static int broadwell_audio_probe(struct platform_device *pdev)
+{
+ broadwell_rt286.dev = &pdev->dev;
+
+ return snd_soc_register_card(&broadwell_rt286);
+}
+
+static int broadwell_audio_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_card(&broadwell_rt286);
+ return 0;
+}
+
+static struct platform_driver broadwell_audio = {
+ .probe = broadwell_audio_probe,
+ .remove = broadwell_audio_remove,
+ .driver = {
+ .name = "broadwell-audio",
+ },
+};
+
+module_platform_driver(broadwell_audio)
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, Xingchao Wang");
+MODULE_DESCRIPTION("Intel SST Audio for WPT/Broadwell");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:broadwell-audio");
diff --git a/sound/soc/intel/boards/byt-max98090.c b/sound/soc/intel/boards/byt-max98090.c
new file mode 100644
index 000000000000..7ab8cc9fbfd5
--- /dev/null
+++ b/sound/soc/intel/boards/byt-max98090.c
@@ -0,0 +1,187 @@
+/*
+ * Intel Baytrail SST MAX98090 machine driver
+ * Copyright (c) 2014, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for
+ * more details.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/acpi.h>
+#include <linux/device.h>
+#include <linux/gpio.h>
+#include <linux/gpio/consumer.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../../codecs/max98090.h"
+
+struct byt_max98090_private {
+ struct snd_soc_jack jack;
+};
+
+static const struct snd_soc_dapm_widget byt_max98090_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Int Mic", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const struct snd_soc_dapm_route byt_max98090_audio_map[] = {
+ {"IN34", NULL, "Headset Mic"},
+ {"Headset Mic", NULL, "MICBIAS"},
+ {"DMICL", NULL, "Int Mic"},
+ {"Headphone", NULL, "HPL"},
+ {"Headphone", NULL, "HPR"},
+ {"Ext Spk", NULL, "SPKL"},
+ {"Ext Spk", NULL, "SPKR"},
+};
+
+static const struct snd_kcontrol_new byt_max98090_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Int Mic"),
+ SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+ {
+ .pin = "Headphone",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ {
+ .pin = "Headset Mic",
+ .mask = SND_JACK_MICROPHONE,
+ },
+};
+
+static struct snd_soc_jack_gpio hs_jack_gpios[] = {
+ {
+ .name = "hp-gpio",
+ .idx = 0,
+ .report = SND_JACK_HEADPHONE | SND_JACK_LINEOUT,
+ .debounce_time = 200,
+ },
+ {
+ .name = "mic-gpio",
+ .idx = 1,
+ .invert = 1,
+ .report = SND_JACK_MICROPHONE,
+ .debounce_time = 200,
+ },
+};
+
+static int byt_max98090_init(struct snd_soc_pcm_runtime *runtime)
+{
+ int ret;
+ struct snd_soc_card *card = runtime->card;
+ struct byt_max98090_private *drv = snd_soc_card_get_drvdata(card);
+ struct snd_soc_jack *jack = &drv->jack;
+
+ card->dapm.idle_bias_off = true;
+
+ ret = snd_soc_dai_set_sysclk(runtime->codec_dai,
+ M98090_REG_SYSTEM_CLOCK,
+ 25000000, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(card->dev, "Can't set codec clock %d\n", ret);
+ return ret;
+ }
+
+ /* Enable jack detection */
+ ret = snd_soc_card_jack_new(runtime->card, "Headset",
+ SND_JACK_LINEOUT | SND_JACK_HEADSET, jack,
+ hs_jack_pins, ARRAY_SIZE(hs_jack_pins));
+ if (ret)
+ return ret;
+
+ return snd_soc_jack_add_gpiods(card->dev->parent, jack,
+ ARRAY_SIZE(hs_jack_gpios),
+ hs_jack_gpios);
+}
+
+static struct snd_soc_dai_link byt_max98090_dais[] = {
+ {
+ .name = "Baytrail Audio",
+ .stream_name = "Audio",
+ .cpu_dai_name = "baytrail-pcm-audio",
+ .codec_dai_name = "HiFi",
+ .codec_name = "i2c-193C9890:00",
+ .platform_name = "baytrail-pcm-audio",
+ .init = byt_max98090_init,
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ },
+};
+
+static struct snd_soc_card byt_max98090_card = {
+ .name = "byt-max98090",
+ .dai_link = byt_max98090_dais,
+ .num_links = ARRAY_SIZE(byt_max98090_dais),
+ .dapm_widgets = byt_max98090_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(byt_max98090_widgets),
+ .dapm_routes = byt_max98090_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(byt_max98090_audio_map),
+ .controls = byt_max98090_controls,
+ .num_controls = ARRAY_SIZE(byt_max98090_controls),
+ .fully_routed = true,
+};
+
+static int byt_max98090_probe(struct platform_device *pdev)
+{
+ int ret_val = 0;
+ struct byt_max98090_private *priv;
+
+ priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC);
+ if (!priv) {
+ dev_err(&pdev->dev, "allocation failed\n");
+ return -ENOMEM;
+ }
+
+ byt_max98090_card.dev = &pdev->dev;
+ snd_soc_card_set_drvdata(&byt_max98090_card, priv);
+ ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_max98090_card);
+ if (ret_val) {
+ dev_err(&pdev->dev,
+ "snd_soc_register_card failed %d\n", ret_val);
+ return ret_val;
+ }
+
+ return ret_val;
+}
+
+static int byt_max98090_remove(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = platform_get_drvdata(pdev);
+ struct byt_max98090_private *priv = snd_soc_card_get_drvdata(card);
+
+ snd_soc_jack_free_gpios(&priv->jack, ARRAY_SIZE(hs_jack_gpios),
+ hs_jack_gpios);
+
+ return 0;
+}
+
+static struct platform_driver byt_max98090_driver = {
+ .probe = byt_max98090_probe,
+ .remove = byt_max98090_remove,
+ .driver = {
+ .name = "byt-max98090",
+ .pm = &snd_soc_pm_ops,
+ },
+};
+module_platform_driver(byt_max98090_driver)
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver");
+MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:byt-max98090");
diff --git a/sound/soc/intel/boards/byt-rt5640.c b/sound/soc/intel/boards/byt-rt5640.c
new file mode 100644
index 000000000000..ae89b9b966d9
--- /dev/null
+++ b/sound/soc/intel/boards/byt-rt5640.c
@@ -0,0 +1,229 @@
+/*
+ * Intel Baytrail SST RT5640 machine driver
+ * Copyright (c) 2014, Intel Corporation.
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for
+ * more details.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/acpi.h>
+#include <linux/device.h>
+#include <linux/dmi.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../../codecs/rt5640.h"
+
+#include "../common/sst-dsp.h"
+
+static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Internal Mic", NULL),
+ SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = {
+ {"Headset Mic", NULL, "MICBIAS1"},
+ {"IN2P", NULL, "Headset Mic"},
+ {"Headphone", NULL, "HPOL"},
+ {"Headphone", NULL, "HPOR"},
+ {"Speaker", NULL, "SPOLP"},
+ {"Speaker", NULL, "SPOLN"},
+ {"Speaker", NULL, "SPORP"},
+ {"Speaker", NULL, "SPORN"},
+};
+
+static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = {
+ {"DMIC1", NULL, "Internal Mic"},
+};
+
+static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = {
+ {"DMIC2", NULL, "Internal Mic"},
+};
+
+static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = {
+ {"Internal Mic", NULL, "MICBIAS1"},
+ {"IN1P", NULL, "Internal Mic"},
+};
+
+enum {
+ BYT_RT5640_DMIC1_MAP,
+ BYT_RT5640_DMIC2_MAP,
+ BYT_RT5640_IN1_MAP,
+};
+
+#define BYT_RT5640_MAP(quirk) ((quirk) & 0xff)
+#define BYT_RT5640_DMIC_EN BIT(16)
+
+static unsigned long byt_rt5640_quirk = BYT_RT5640_DMIC1_MAP |
+ BYT_RT5640_DMIC_EN;
+
+static const struct snd_kcontrol_new byt_rt5640_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Internal Mic"),
+ SOC_DAPM_PIN_SWITCH("Speaker"),
+};
+
+static int byt_rt5640_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;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1,
+ params_rate(params) * 256,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(codec_dai->dev, "can't set codec clock %d\n", ret);
+ return ret;
+ }
+ ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1,
+ params_rate(params) * 64,
+ params_rate(params) * 256);
+ if (ret < 0) {
+ dev_err(codec_dai->dev, "can't set codec pll: %d\n", ret);
+ return ret;
+ }
+ return 0;
+}
+
+static int byt_rt5640_quirk_cb(const struct dmi_system_id *id)
+{
+ byt_rt5640_quirk = (unsigned long)id->driver_data;
+ return 1;
+}
+
+static const struct dmi_system_id byt_rt5640_quirk_table[] = {
+ {
+ .callback = byt_rt5640_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "ASUSTeK COMPUTER INC."),
+ DMI_MATCH(DMI_PRODUCT_NAME, "T100TA"),
+ },
+ .driver_data = (unsigned long *)BYT_RT5640_IN1_MAP,
+ },
+ {
+ .callback = byt_rt5640_quirk_cb,
+ .matches = {
+ DMI_MATCH(DMI_SYS_VENDOR, "DellInc."),
+ DMI_MATCH(DMI_PRODUCT_NAME, "Venue 8 Pro 5830"),
+ },
+ .driver_data = (unsigned long *)(BYT_RT5640_DMIC2_MAP |
+ BYT_RT5640_DMIC_EN),
+ },
+ {}
+};
+
+static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
+{
+ int ret;
+ struct snd_soc_codec *codec = runtime->codec;
+ struct snd_soc_card *card = runtime->card;
+ const struct snd_soc_dapm_route *custom_map;
+ int num_routes;
+
+ card->dapm.idle_bias_off = true;
+
+ ret = snd_soc_add_card_controls(card, byt_rt5640_controls,
+ ARRAY_SIZE(byt_rt5640_controls));
+ if (ret) {
+ dev_err(card->dev, "unable to add card controls\n");
+ return ret;
+ }
+
+ dmi_check_system(byt_rt5640_quirk_table);
+ switch (BYT_RT5640_MAP(byt_rt5640_quirk)) {
+ case BYT_RT5640_IN1_MAP:
+ custom_map = byt_rt5640_intmic_in1_map;
+ num_routes = ARRAY_SIZE(byt_rt5640_intmic_in1_map);
+ break;
+ case BYT_RT5640_DMIC2_MAP:
+ custom_map = byt_rt5640_intmic_dmic2_map;
+ num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic2_map);
+ break;
+ default:
+ custom_map = byt_rt5640_intmic_dmic1_map;
+ num_routes = ARRAY_SIZE(byt_rt5640_intmic_dmic1_map);
+ }
+
+ ret = snd_soc_dapm_add_routes(&card->dapm, custom_map, num_routes);
+ if (ret)
+ return ret;
+
+ if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN) {
+ ret = rt5640_dmic_enable(codec, 0, 0);
+ if (ret)
+ return ret;
+ }
+
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone");
+ snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker");
+
+ return ret;
+}
+
+static struct snd_soc_ops byt_rt5640_ops = {
+ .hw_params = byt_rt5640_hw_params,
+};
+
+static struct snd_soc_dai_link byt_rt5640_dais[] = {
+ {
+ .name = "Baytrail Audio",
+ .stream_name = "Audio",
+ .cpu_dai_name = "baytrail-pcm-audio",
+ .codec_dai_name = "rt5640-aif1",
+ .codec_name = "i2c-10EC5640:00",
+ .platform_name = "baytrail-pcm-audio",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .init = byt_rt5640_init,
+ .ops = &byt_rt5640_ops,
+ },
+};
+
+static struct snd_soc_card byt_rt5640_card = {
+ .name = "byt-rt5640",
+ .dai_link = byt_rt5640_dais,
+ .num_links = ARRAY_SIZE(byt_rt5640_dais),
+ .dapm_widgets = byt_rt5640_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(byt_rt5640_widgets),
+ .dapm_routes = byt_rt5640_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(byt_rt5640_audio_map),
+ .fully_routed = true,
+};
+
+static int byt_rt5640_probe(struct platform_device *pdev)
+{
+ struct snd_soc_card *card = &byt_rt5640_card;
+
+ card->dev = &pdev->dev;
+ return devm_snd_soc_register_card(&pdev->dev, card);
+}
+
+static struct platform_driver byt_rt5640_audio = {
+ .probe = byt_rt5640_probe,
+ .driver = {
+ .name = "byt-rt5640",
+ .pm = &snd_soc_pm_ops,
+ },
+};
+module_platform_driver(byt_rt5640_audio)
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver");
+MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:byt-rt5640");
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
new file mode 100644
index 000000000000..7f55d59024a8
--- /dev/null
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -0,0 +1,227 @@
+/*
+ * byt_cr_dpcm_rt5640.c - ASoc Machine driver for Intel Byt CR platform
+ *
+ * Copyright (C) 2014 Intel Corp
+ * Author: Subhransu S. Prusty <subhransu.s.prusty@intel.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; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/input.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include "../../codecs/rt5640.h"
+#include "../atom/sst-atom-controls.h"
+
+static const struct snd_soc_dapm_widget byt_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Int Mic", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const struct snd_soc_dapm_route byt_audio_map[] = {
+ {"IN2P", NULL, "Headset Mic"},
+ {"IN2N", NULL, "Headset Mic"},
+ {"Headset Mic", NULL, "MICBIAS1"},
+ {"IN1P", NULL, "MICBIAS1"},
+ {"LDO2", NULL, "Int Mic"},
+ {"Headphone", NULL, "HPOL"},
+ {"Headphone", NULL, "HPOR"},
+ {"Ext Spk", NULL, "SPOLP"},
+ {"Ext Spk", NULL, "SPOLN"},
+ {"Ext Spk", NULL, "SPORP"},
+ {"Ext Spk", NULL, "SPORN"},
+
+ {"AIF1 Playback", NULL, "ssp2 Tx"},
+ {"ssp2 Tx", NULL, "codec_out0"},
+ {"ssp2 Tx", NULL, "codec_out1"},
+ {"codec_in0", NULL, "ssp2 Rx"},
+ {"codec_in1", NULL, "ssp2 Rx"},
+ {"ssp2 Rx", NULL, "AIF1 Capture"},
+};
+
+static const struct snd_kcontrol_new byt_mc_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Int Mic"),
+ SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static int byt_aif1_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;
+ int ret;
+
+ snd_soc_dai_set_bclk_ratio(codec_dai, 50);
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1,
+ params_rate(params) * 512,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec clock %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1,
+ params_rate(params) * 50,
+ params_rate(params) * 512);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_pcm_stream byt_dai_params = {
+ .formats = SNDRV_PCM_FMTBIT_S24_LE,
+ .rate_min = 48000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+};
+
+static int byt_codec_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* The DSP will covert the FE rate to 48k, stereo, 24bits */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set SSP2 to 24-bit */
+ params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
+ return 0;
+}
+
+static unsigned int rates_48000[] = {
+ 48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_48000 = {
+ .count = ARRAY_SIZE(rates_48000),
+ .list = rates_48000,
+};
+
+static int byt_aif1_startup(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_48000);
+}
+
+static struct snd_soc_ops byt_aif1_ops = {
+ .startup = byt_aif1_startup,
+};
+
+static struct snd_soc_ops byt_be_ssp2_ops = {
+ .hw_params = byt_aif1_hw_params,
+};
+
+static struct snd_soc_dai_link byt_dailink[] = {
+ [MERR_DPCM_AUDIO] = {
+ .name = "Baytrail Audio Port",
+ .stream_name = "Baytrail Audio",
+ .cpu_dai_name = "media-cpu-dai",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .platform_name = "sst-mfld-platform",
+ .ignore_suspend = 1,
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ops = &byt_aif1_ops,
+ },
+ [MERR_DPCM_COMPR] = {
+ .name = "Baytrail Compressed Port",
+ .stream_name = "Baytrail Compress",
+ .cpu_dai_name = "compress-cpu-dai",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .platform_name = "sst-mfld-platform",
+ },
+ /* back ends */
+ {
+ .name = "SSP2-Codec",
+ .be_id = 1,
+ .cpu_dai_name = "ssp2-port",
+ .platform_name = "sst-mfld-platform",
+ .no_pcm = 1,
+ .codec_dai_name = "rt5640-aif1",
+ .codec_name = "i2c-10EC5640:00",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
+ | SND_SOC_DAIFMT_CBS_CFS,
+ .be_hw_params_fixup = byt_codec_fixup,
+ .ignore_suspend = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ops = &byt_be_ssp2_ops,
+ },
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_byt = {
+ .name = "baytrailcraudio",
+ .dai_link = byt_dailink,
+ .num_links = ARRAY_SIZE(byt_dailink),
+ .dapm_widgets = byt_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(byt_dapm_widgets),
+ .dapm_routes = byt_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(byt_audio_map),
+ .controls = byt_mc_controls,
+ .num_controls = ARRAY_SIZE(byt_mc_controls),
+};
+
+static int snd_byt_mc_probe(struct platform_device *pdev)
+{
+ int ret_val = 0;
+
+ /* register the soc card */
+ snd_soc_card_byt.dev = &pdev->dev;
+
+ ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_byt);
+ if (ret_val) {
+ dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n", ret_val);
+ return ret_val;
+ }
+ platform_set_drvdata(pdev, &snd_soc_card_byt);
+ return ret_val;
+}
+
+static struct platform_driver snd_byt_mc_driver = {
+ .driver = {
+ .name = "bytt100_rt5640",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = snd_byt_mc_probe,
+};
+
+module_platform_driver(snd_byt_mc_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver");
+MODULE_AUTHOR("Subhransu S. Prusty <subhransu.s.prusty@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:bytt100_rt5640");
diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
new file mode 100644
index 000000000000..20a28b22e30f
--- /dev/null
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -0,0 +1,324 @@
+/*
+ * cht-bsw-rt5645.c - ASoc Machine driver for Intel Cherryview-based platforms
+ * Cherrytrail and Braswell, with RT5645 codec.
+ *
+ * Copyright (C) 2015 Intel Corp
+ * Author: Fang, Yang A <yang.a.fang@intel.com>
+ * N,Harshapriya <harshapriya.n@intel.com>
+ * This file is modified from cht_bsw_rt5672.c
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ * 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; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../../codecs/rt5645.h"
+#include "../atom/sst-atom-controls.h"
+
+#define CHT_PLAT_CLK_3_HZ 19200000
+#define CHT_CODEC_DAI "rt5645-aif1"
+
+struct cht_mc_private {
+ struct snd_soc_jack hp_jack;
+ struct snd_soc_jack mic_jack;
+};
+
+static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
+{
+ int i;
+
+ for (i = 0; i < card->num_rtd; i++) {
+ struct snd_soc_pcm_runtime *rtd;
+
+ rtd = card->rtd + i;
+ if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
+ strlen(CHT_CODEC_DAI)))
+ return rtd->codec_dai;
+ }
+ return NULL;
+}
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ struct snd_soc_dapm_context *dapm = w->dapm;
+ struct snd_soc_card *card = dapm->card;
+ struct snd_soc_dai *codec_dai;
+ int ret;
+
+ codec_dai = cht_get_codec_dai(card);
+ if (!codec_dai) {
+ dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
+ return -EIO;
+ }
+
+ if (!SND_SOC_DAPM_EVENT_OFF(event))
+ return 0;
+
+ /* Set codec sysclk source to its internal clock because codec PLL will
+ * be off when idle and MCLK will also be off by ACPI when codec is
+ * runtime suspended. Codec needs clock for jack detection and button
+ * press.
+ */
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_RCCLK,
+ 0, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(card->dev, "can't set codec sysclk: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Int Mic", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+ platform_clock_control, SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route cht_audio_map[] = {
+ {"IN1P", NULL, "Headset Mic"},
+ {"IN1N", NULL, "Headset Mic"},
+ {"DMIC L1", NULL, "Int Mic"},
+ {"DMIC R1", NULL, "Int Mic"},
+ {"Headphone", NULL, "HPOL"},
+ {"Headphone", NULL, "HPOR"},
+ {"Ext Spk", NULL, "SPOL"},
+ {"Ext Spk", NULL, "SPOR"},
+ {"AIF1 Playback", NULL, "ssp2 Tx"},
+ {"ssp2 Tx", NULL, "codec_out0"},
+ {"ssp2 Tx", NULL, "codec_out1"},
+ {"codec_in0", NULL, "ssp2 Rx" },
+ {"codec_in1", NULL, "ssp2 Rx" },
+ {"ssp2 Rx", NULL, "AIF1 Capture"},
+ {"Headphone", NULL, "Platform Clock"},
+ {"Headset Mic", NULL, "Platform Clock"},
+ {"Int Mic", NULL, "Platform Clock"},
+ {"Ext Spk", NULL, "Platform Clock"},
+};
+
+static const struct snd_kcontrol_new cht_mc_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Int Mic"),
+ SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static int cht_aif1_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;
+ int ret;
+
+ /* set codec PLL source to the 19.2MHz platform clock (MCLK) */
+ ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
+ CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1,
+ params_rate(params) * 512, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
+{
+ int ret;
+ struct snd_soc_codec *codec = runtime->codec;
+ struct snd_soc_dai *codec_dai = runtime->codec_dai;
+ struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
+
+ /* Select clk_i2s1_asrc as ASRC clock source */
+ rt5645_sel_asrc_clk_src(codec,
+ RT5645_DA_STEREO_FILTER |
+ RT5645_DA_MONO_L_FILTER |
+ RT5645_DA_MONO_R_FILTER |
+ RT5645_AD_STEREO_FILTER,
+ RT5645_CLK_SEL_I2S1_ASRC);
+
+ /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
+ if (ret < 0) {
+ dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_card_jack_new(runtime->card, "Headphone Jack",
+ SND_JACK_HEADPHONE, &ctx->hp_jack,
+ NULL, 0);
+ if (ret) {
+ dev_err(runtime->dev, "HP jack creation failed %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_card_jack_new(runtime->card, "Mic Jack",
+ SND_JACK_MICROPHONE, &ctx->mic_jack,
+ NULL, 0);
+ if (ret) {
+ dev_err(runtime->dev, "Mic jack creation failed %d\n", ret);
+ return ret;
+ }
+
+ rt5645_set_jack_detect(codec, &ctx->hp_jack, &ctx->mic_jack);
+
+ return ret;
+}
+
+static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* The DSP will covert the FE rate to 48k, stereo, 24bits */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set SSP2 to 24-bit */
+ params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
+ return 0;
+}
+
+static unsigned int rates_48000[] = {
+ 48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_48000 = {
+ .count = ARRAY_SIZE(rates_48000),
+ .list = rates_48000,
+};
+
+static int cht_aif1_startup(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_48000);
+}
+
+static struct snd_soc_ops cht_aif1_ops = {
+ .startup = cht_aif1_startup,
+};
+
+static struct snd_soc_ops cht_be_ssp2_ops = {
+ .hw_params = cht_aif1_hw_params,
+};
+
+static struct snd_soc_dai_link cht_dailink[] = {
+ [MERR_DPCM_AUDIO] = {
+ .name = "Audio Port",
+ .stream_name = "Audio",
+ .cpu_dai_name = "media-cpu-dai",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .platform_name = "sst-mfld-platform",
+ .ignore_suspend = 1,
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ops = &cht_aif1_ops,
+ },
+ [MERR_DPCM_COMPR] = {
+ .name = "Compressed Port",
+ .stream_name = "Compress",
+ .cpu_dai_name = "compress-cpu-dai",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .platform_name = "sst-mfld-platform",
+ },
+ /* CODEC<->CODEC link */
+ /* back ends */
+ {
+ .name = "SSP2-Codec",
+ .be_id = 1,
+ .cpu_dai_name = "ssp2-port",
+ .platform_name = "sst-mfld-platform",
+ .no_pcm = 1,
+ .codec_dai_name = "rt5645-aif1",
+ .codec_name = "i2c-10EC5645:00",
+ .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
+ | SND_SOC_DAIFMT_CBS_CFS,
+ .init = cht_codec_init,
+ .be_hw_params_fixup = cht_codec_fixup,
+ .ignore_suspend = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ops = &cht_be_ssp2_ops,
+ },
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_cht = {
+ .name = "chtrt5645",
+ .dai_link = cht_dailink,
+ .num_links = ARRAY_SIZE(cht_dailink),
+ .dapm_widgets = cht_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
+ .dapm_routes = cht_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(cht_audio_map),
+ .controls = cht_mc_controls,
+ .num_controls = ARRAY_SIZE(cht_mc_controls),
+};
+
+static int snd_cht_mc_probe(struct platform_device *pdev)
+{
+ int ret_val = 0;
+ struct cht_mc_private *drv;
+
+ drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC);
+ if (!drv)
+ return -ENOMEM;
+
+ snd_soc_card_cht.dev = &pdev->dev;
+ snd_soc_card_set_drvdata(&snd_soc_card_cht, drv);
+ ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
+ if (ret_val) {
+ dev_err(&pdev->dev,
+ "snd_soc_register_card failed %d\n", ret_val);
+ return ret_val;
+ }
+ platform_set_drvdata(pdev, &snd_soc_card_cht);
+ return ret_val;
+}
+
+static struct platform_driver snd_cht_mc_driver = {
+ .driver = {
+ .name = "cht-bsw-rt5645",
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = snd_cht_mc_probe,
+};
+
+module_platform_driver(snd_cht_mc_driver)
+
+MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver");
+MODULE_AUTHOR("Fang, Yang A,N,Harshapriya");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:cht-bsw-rt5645");
diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c
new file mode 100644
index 000000000000..2c9cc5be439e
--- /dev/null
+++ b/sound/soc/intel/boards/cht_bsw_rt5672.c
@@ -0,0 +1,366 @@
+/*
+ * cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms
+ * Cherrytrail and Braswell, with RT5672 codec.
+ *
+ * Copyright (C) 2014 Intel Corp
+ * Author: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
+ * Mengdong Lin <mengdong.lin@intel.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; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../../codecs/rt5670.h"
+#include "../atom/sst-atom-controls.h"
+
+/* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */
+#define CHT_PLAT_CLK_3_HZ 19200000
+#define CHT_CODEC_DAI "rt5670-aif1"
+
+static struct snd_soc_jack cht_bsw_headset;
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin cht_bsw_headset_pins[] = {
+ {
+ .pin = "Headset Mic",
+ .mask = SND_JACK_MICROPHONE,
+ },
+ {
+ .pin = "Headphone",
+ .mask = SND_JACK_HEADPHONE,
+ },
+};
+
+static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
+{
+ int i;
+
+ for (i = 0; i < card->num_rtd; i++) {
+ struct snd_soc_pcm_runtime *rtd;
+
+ rtd = card->rtd + i;
+ if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
+ strlen(CHT_CODEC_DAI)))
+ return rtd->codec_dai;
+ }
+ return NULL;
+}
+
+static int platform_clock_control(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *k, int event)
+{
+ struct snd_soc_dapm_context *dapm = w->dapm;
+ struct snd_soc_card *card = dapm->card;
+ struct snd_soc_dai *codec_dai;
+ int ret;
+
+ codec_dai = cht_get_codec_dai(card);
+ if (!codec_dai) {
+ dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
+ return -EIO;
+ }
+
+ if (SND_SOC_DAPM_EVENT_ON(event)) {
+ /* set codec PLL source to the 19.2MHz platform clock (MCLK) */
+ ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK,
+ CHT_PLAT_CLK_3_HZ, 48000 * 512);
+ if (ret < 0) {
+ dev_err(card->dev, "can't set codec pll: %d\n", ret);
+ return ret;
+ }
+
+ /* set codec sysclk source to PLL */
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1,
+ 48000 * 512, SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(card->dev, "can't set codec sysclk: %d\n", ret);
+ return ret;
+ }
+ } else {
+ /* Set codec sysclk source to its internal clock because codec
+ * PLL will be off when idle and MCLK will also be off by ACPI
+ * when codec is runtime suspended. Codec needs clock for jack
+ * detection and button press.
+ */
+ snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK,
+ 48000 * 512, SND_SOC_CLOCK_IN);
+ }
+ return 0;
+}
+
+static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
+ SND_SOC_DAPM_HP("Headphone", NULL),
+ SND_SOC_DAPM_MIC("Headset Mic", NULL),
+ SND_SOC_DAPM_MIC("Int Mic", NULL),
+ SND_SOC_DAPM_SPK("Ext Spk", NULL),
+ SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
+ platform_clock_control, SND_SOC_DAPM_PRE_PMU |
+ SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route cht_audio_map[] = {
+ {"IN1P", NULL, "Headset Mic"},
+ {"IN1N", NULL, "Headset Mic"},
+ {"DMIC L1", NULL, "Int Mic"},
+ {"DMIC R1", NULL, "Int Mic"},
+ {"Headphone", NULL, "HPOL"},
+ {"Headphone", NULL, "HPOR"},
+ {"Ext Spk", NULL, "SPOLP"},
+ {"Ext Spk", NULL, "SPOLN"},
+ {"Ext Spk", NULL, "SPORP"},
+ {"Ext Spk", NULL, "SPORN"},
+ {"AIF1 Playback", NULL, "ssp2 Tx"},
+ {"ssp2 Tx", NULL, "codec_out0"},
+ {"ssp2 Tx", NULL, "codec_out1"},
+ {"codec_in0", NULL, "ssp2 Rx"},
+ {"codec_in1", NULL, "ssp2 Rx"},
+ {"ssp2 Rx", NULL, "AIF1 Capture"},
+ {"Headphone", NULL, "Platform Clock"},
+ {"Headset Mic", NULL, "Platform Clock"},
+ {"Int Mic", NULL, "Platform Clock"},
+ {"Ext Spk", NULL, "Platform Clock"},
+};
+
+static const struct snd_kcontrol_new cht_mc_controls[] = {
+ SOC_DAPM_PIN_SWITCH("Headphone"),
+ SOC_DAPM_PIN_SWITCH("Headset Mic"),
+ SOC_DAPM_PIN_SWITCH("Int Mic"),
+ SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static int cht_aif1_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;
+ int ret;
+
+ /* set codec PLL source to the 19.2MHz platform clock (MCLK) */
+ ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK,
+ CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
+ return ret;
+ }
+
+ /* set codec sysclk source to PLL */
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1,
+ params_rate(params) * 512,
+ SND_SOC_CLOCK_IN);
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
+ return ret;
+ }
+ return 0;
+}
+
+static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
+{
+ int ret;
+ struct snd_soc_dai *codec_dai = runtime->codec_dai;
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
+ ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
+ if (ret < 0) {
+ dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
+ return ret;
+ }
+
+ /* Select codec ASRC clock source to track I2S1 clock, because codec
+ * is in slave mode and 100fs I2S format (BCLK = 100 * LRCLK) cannot
+ * be supported by RT5672. Otherwise, ASRC will be disabled and cause
+ * noise.
+ */
+ rt5670_sel_asrc_clk_src(codec,
+ RT5670_DA_STEREO_FILTER
+ | RT5670_DA_MONO_L_FILTER
+ | RT5670_DA_MONO_R_FILTER
+ | RT5670_AD_STEREO_FILTER
+ | RT5670_AD_MONO_L_FILTER
+ | RT5670_AD_MONO_R_FILTER,
+ RT5670_CLK_SEL_I2S1_ASRC);
+
+ ret = snd_soc_card_jack_new(runtime->card, "Headset",
+ SND_JACK_HEADSET | SND_JACK_BTN_0 |
+ SND_JACK_BTN_1 | SND_JACK_BTN_2, &cht_bsw_headset,
+ cht_bsw_headset_pins, ARRAY_SIZE(cht_bsw_headset_pins));
+ if (ret)
+ return ret;
+
+ rt5670_set_jack_detect(codec, &cht_bsw_headset);
+ return 0;
+}
+
+static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* The DSP will covert the FE rate to 48k, stereo, 24bits */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set SSP2 to 24-bit */
+ params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
+ return 0;
+}
+
+static unsigned int rates_48000[] = {
+ 48000,
+};
+
+static struct snd_pcm_hw_constraint_list constraints_48000 = {
+ .count = ARRAY_SIZE(rates_48000),
+ .list = rates_48000,
+};
+
+static int cht_aif1_startup(struct snd_pcm_substream *substream)
+{
+ return snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_48000);
+}
+
+static struct snd_soc_ops cht_aif1_ops = {
+ .startup = cht_aif1_startup,
+};
+
+static struct snd_soc_ops cht_be_ssp2_ops = {
+ .hw_params = cht_aif1_hw_params,
+};
+
+static struct snd_soc_dai_link cht_dailink[] = {
+ /* Front End DAI links */
+ [MERR_DPCM_AUDIO] = {
+ .name = "Audio Port",
+ .stream_name = "Audio",
+ .cpu_dai_name = "media-cpu-dai",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .platform_name = "sst-mfld-platform",
+ .nonatomic = true,
+ .dynamic = 1,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ops = &cht_aif1_ops,
+ },
+ [MERR_DPCM_COMPR] = {
+ .name = "Compressed Port",
+ .stream_name = "Compress",
+ .cpu_dai_name = "compress-cpu-dai",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .codec_name = "snd-soc-dummy",
+ .platform_name = "sst-mfld-platform",
+ },
+
+ /* Back End DAI links */
+ {
+ /* SSP2 - Codec */
+ .name = "SSP2-Codec",
+ .be_id = 1,
+ .cpu_dai_name = "ssp2-port",
+ .platform_name = "sst-mfld-platform",
+ .no_pcm = 1,
+ .nonatomic = true,
+ .codec_dai_name = "rt5670-aif1",
+ .codec_name = "i2c-10EC5670:00",
+ .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
+ | SND_SOC_DAIFMT_CBS_CFS,
+ .init = cht_codec_init,
+ .be_hw_params_fixup = cht_codec_fixup,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ .ops = &cht_be_ssp2_ops,
+ },
+};
+
+static int cht_suspend_pre(struct snd_soc_card *card)
+{
+ struct snd_soc_codec *codec;
+
+ list_for_each_entry(codec, &card->codec_dev_list, card_list) {
+ if (!strcmp(codec->component.name, "i2c-10EC5670:00")) {
+ dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n");
+ rt5670_jack_suspend(codec);
+ break;
+ }
+ }
+ return 0;
+}
+
+static int cht_resume_post(struct snd_soc_card *card)
+{
+ struct snd_soc_codec *codec;
+
+ list_for_each_entry(codec, &card->codec_dev_list, card_list) {
+ if (!strcmp(codec->component.name, "i2c-10EC5670:00")) {
+ dev_dbg(codec->dev, "enabling jack detect for resume.\n");
+ rt5670_jack_resume(codec);
+ break;
+ }
+ }
+
+ return 0;
+}
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_cht = {
+ .name = "cherrytrailcraudio",
+ .dai_link = cht_dailink,
+ .num_links = ARRAY_SIZE(cht_dailink),
+ .dapm_widgets = cht_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
+ .dapm_routes = cht_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(cht_audio_map),
+ .controls = cht_mc_controls,
+ .num_controls = ARRAY_SIZE(cht_mc_controls),
+ .suspend_pre = cht_suspend_pre,
+ .resume_post = cht_resume_post,
+};
+
+static int snd_cht_mc_probe(struct platform_device *pdev)
+{
+ int ret_val = 0;
+
+ /* register the soc card */
+ snd_soc_card_cht.dev = &pdev->dev;
+ ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
+ if (ret_val) {
+ dev_err(&pdev->dev,
+ "snd_soc_register_card failed %d\n", ret_val);
+ return ret_val;
+ }
+ platform_set_drvdata(pdev, &snd_soc_card_cht);
+ return ret_val;
+}
+
+static struct platform_driver snd_cht_mc_driver = {
+ .driver = {
+ .name = "cht-bsw-rt5672",
+ },
+ .probe = snd_cht_mc_probe,
+};
+
+module_platform_driver(snd_cht_mc_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver");
+MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:cht-bsw-rt5672");
diff --git a/sound/soc/intel/boards/haswell.c b/sound/soc/intel/boards/haswell.c
new file mode 100644
index 000000000000..22558572cb9c
--- /dev/null
+++ b/sound/soc/intel/boards/haswell.c
@@ -0,0 +1,209 @@
+/*
+ * Intel Haswell Lynxpoint SST Audio
+ *
+ * Copyright (C) 2013, Intel Corporation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License version
+ * 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+
+#include "../common/sst-dsp.h"
+#include "../haswell/sst-haswell-ipc.h"
+
+#include "../../codecs/rt5640.h"
+
+/* Haswell ULT platforms have a Headphone and Mic jack */
+static const struct snd_soc_dapm_widget haswell_widgets[] = {
+ SND_SOC_DAPM_HP("Headphones", NULL),
+ SND_SOC_DAPM_MIC("Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route haswell_rt5640_map[] = {
+
+ {"Headphones", NULL, "HPOR"},
+ {"Headphones", NULL, "HPOL"},
+ {"IN2P", NULL, "Mic"},
+
+ /* CODEC BE connections */
+ {"SSP0 CODEC IN", NULL, "AIF1 Capture"},
+ {"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
+};
+
+static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_interval *rate = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval *channels = hw_param_interval(params,
+ SNDRV_PCM_HW_PARAM_CHANNELS);
+
+ /* The ADSP will covert the FE rate to 48k, stereo */
+ rate->min = rate->max = 48000;
+ channels->min = channels->max = 2;
+
+ /* set SSP0 to 16 bit */
+ params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
+ return 0;
+}
+
+static int haswell_rt5640_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;
+ int ret;
+
+ ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000,
+ SND_SOC_CLOCK_IN);
+
+ if (ret < 0) {
+ dev_err(rtd->dev, "can't set codec sysclk configuration\n");
+ return ret;
+ }
+
+ /* set correct codec filter for DAI format and clock config */
+ snd_soc_update_bits(rtd->codec, 0x83, 0xffff, 0x8000);
+
+ return ret;
+}
+
+static struct snd_soc_ops haswell_rt5640_ops = {
+ .hw_params = haswell_rt5640_hw_params,
+};
+
+static int haswell_rtd_init(struct snd_soc_pcm_runtime *rtd)
+{
+ struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev);
+ struct sst_hsw *haswell = pdata->dsp;
+ int ret;
+
+ /* Set ADSP SSP port settings */
+ ret = sst_hsw_device_set_config(haswell, SST_HSW_DEVICE_SSP_0,
+ SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
+ SST_HSW_DEVICE_CLOCK_MASTER, 9);
+ if (ret < 0) {
+ dev_err(rtd->dev, "failed to set device config\n");
+ return ret;
+ }
+
+ return 0;
+}
+
+static struct snd_soc_dai_link haswell_rt5640_dais[] = {
+ /* Front End DAI links */
+ {
+ .name = "System",
+ .stream_name = "System Playback/Capture",
+ .cpu_dai_name = "System Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .init = haswell_rtd_init,
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+ {
+ .name = "Offload0",
+ .stream_name = "Offload0 Playback",
+ .cpu_dai_name = "Offload0 Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Offload1",
+ .stream_name = "Offload1 Playback",
+ .cpu_dai_name = "Offload1 Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 1,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_playback = 1,
+ },
+ {
+ .name = "Loopback",
+ .stream_name = "Loopback",
+ .cpu_dai_name = "Loopback Pin",
+ .platform_name = "haswell-pcm-audio",
+ .dynamic = 0,
+ .codec_name = "snd-soc-dummy",
+ .codec_dai_name = "snd-soc-dummy-dai",
+ .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
+ .dpcm_capture = 1,
+ },
+
+ /* Back End DAI links */
+ {
+ /* SSP0 - Codec */
+ .name = "Codec",
+ .be_id = 0,
+ .cpu_dai_name = "snd-soc-dummy-dai",
+ .platform_name = "snd-soc-dummy",
+ .no_pcm = 1,
+ .codec_name = "i2c-INT33CA:00",
+ .codec_dai_name = "rt5640-aif1",
+ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBS_CFS,
+ .ignore_suspend = 1,
+ .ignore_pmdown_time = 1,
+ .be_hw_params_fixup = haswell_ssp0_fixup,
+ .ops = &haswell_rt5640_ops,
+ .dpcm_playback = 1,
+ .dpcm_capture = 1,
+ },
+};
+
+/* audio machine driver for Haswell Lynxpoint DSP + RT5640 */
+static struct snd_soc_card haswell_rt5640 = {
+ .name = "haswell-rt5640",
+ .owner = THIS_MODULE,
+ .dai_link = haswell_rt5640_dais,
+ .num_links = ARRAY_SIZE(haswell_rt5640_dais),
+ .dapm_widgets = haswell_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(haswell_widgets),
+ .dapm_routes = haswell_rt5640_map,
+ .num_dapm_routes = ARRAY_SIZE(haswell_rt5640_map),
+ .fully_routed = true,
+};
+
+static int haswell_audio_probe(struct platform_device *pdev)
+{
+ haswell_rt5640.dev = &pdev->dev;
+
+ return devm_snd_soc_register_card(&pdev->dev, &haswell_rt5640);
+}
+
+static struct platform_driver haswell_audio = {
+ .probe = haswell_audio_probe,
+ .driver = {
+ .name = "haswell-audio",
+ },
+};
+
+module_platform_driver(haswell_audio)
+
+/* Module information */
+MODULE_AUTHOR("Liam Girdwood, Xingchao Wang");
+MODULE_DESCRIPTION("Intel SST Audio for Haswell Lynxpoint");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:haswell-audio");
diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c
new file mode 100644
index 000000000000..49c09a0add79
--- /dev/null
+++ b/sound/soc/intel/boards/mfld_machine.c
@@ -0,0 +1,430 @@
+/*
+ * mfld_machine.c - ASoc Machine driver for Intel Medfield MID platform
+ *
+ * Copyright (C) 2010 Intel Corp
+ * Author: Vinod Koul <vinod.koul@intel.com>
+ * Author: Harsha Priya <priya.harsha@intel.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; version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * 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.,
+ * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+
+#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt
+
+#include <linux/init.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include "../codecs/sn95031.h"
+
+#define MID_MONO 1
+#define MID_STEREO 2
+#define MID_MAX_CAP 5
+#define MFLD_JACK_INSERT 0x04
+
+enum soc_mic_bias_zones {
+ MFLD_MV_START = 0,
+ /* mic bias volutage range for Headphones*/
+ MFLD_MV_HP = 400,
+ /* mic bias volutage range for American Headset*/
+ MFLD_MV_AM_HS = 650,
+ /* mic bias volutage range for Headset*/
+ MFLD_MV_HS = 2000,
+ MFLD_MV_UNDEFINED,
+};
+
+static unsigned int hs_switch;
+static unsigned int lo_dac;
+static struct snd_soc_codec *mfld_codec;
+
+struct mfld_mc_private {
+ void __iomem *int_base;
+ u8 interrupt_status;
+};
+
+struct snd_soc_jack mfld_jack;
+
+/*Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin mfld_jack_pins[] = {
+ {
+ .pin = "Headphones",
+ .mask = SND_JACK_HEADPHONE,
+ },
+ {
+ .pin = "AMIC1",
+ .mask = SND_JACK_MICROPHONE,
+ },
+};
+
+/* jack detection voltage zones */
+static struct snd_soc_jack_zone mfld_zones[] = {
+ {MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE},
+ {MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET},
+};
+
+/* sound card controls */
+static const char *headset_switch_text[] = {"Earpiece", "Headset"};
+
+static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"};
+
+static const struct soc_enum headset_enum =
+ SOC_ENUM_SINGLE_EXT(2, headset_switch_text);
+
+static const struct soc_enum lo_enum =
+ SOC_ENUM_SINGLE_EXT(4, lo_text);
+
+static int headset_get_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = hs_switch;
+ return 0;
+}
+
+static int headset_set_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_context *dapm = &card->dapm;
+
+ if (ucontrol->value.integer.value[0] == hs_switch)
+ return 0;
+
+ snd_soc_dapm_mutex_lock(dapm);
+
+ if (ucontrol->value.integer.value[0]) {
+ pr_debug("hs_set HS path\n");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
+ } else {
+ pr_debug("hs_set EP path\n");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT");
+ }
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
+
+ hs_switch = ucontrol->value.integer.value[0];
+
+ return 0;
+}
+
+static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm)
+{
+ snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTL");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTR");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTL");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTR");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "VIB1OUT");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "VIB2OUT");
+ if (hs_switch) {
+ snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
+ } else {
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT");
+ }
+}
+
+static int lo_get_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ ucontrol->value.integer.value[0] = lo_dac;
+ return 0;
+}
+
+static int lo_set_switch(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_dapm_context *dapm = &card->dapm;
+
+ if (ucontrol->value.integer.value[0] == lo_dac)
+ return 0;
+
+ snd_soc_dapm_mutex_lock(dapm);
+
+ /* we dont want to work with last state of lineout so just enable all
+ * pins and then disable pins not required
+ */
+ lo_enable_out_pins(dapm);
+
+ switch (ucontrol->value.integer.value[0]) {
+ case 0:
+ pr_debug("set vibra path\n");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "VIB2OUT");
+ snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0);
+ break;
+
+ case 1:
+ pr_debug("set hs path\n");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT");
+ snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x22);
+ break;
+
+ case 2:
+ pr_debug("set spkr path\n");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTL");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTR");
+ snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x44);
+ break;
+
+ case 3:
+ pr_debug("set null path\n");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTL");
+ snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTR");
+ snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x66);
+ break;
+ }
+
+ snd_soc_dapm_sync_unlocked(dapm);
+
+ snd_soc_dapm_mutex_unlock(dapm);
+
+ lo_dac = ucontrol->value.integer.value[0];
+ return 0;
+}
+
+static const struct snd_kcontrol_new mfld_snd_controls[] = {
+ SOC_ENUM_EXT("Playback Switch", headset_enum,
+ headset_get_switch, headset_set_switch),
+ SOC_ENUM_EXT("Lineout Mux", lo_enum,
+ lo_get_switch, lo_set_switch),
+};
+
+static const struct snd_soc_dapm_widget mfld_widgets[] = {
+ SND_SOC_DAPM_HP("Headphones", NULL),
+ SND_SOC_DAPM_MIC("Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route mfld_map[] = {
+ {"Headphones", NULL, "HPOUTR"},
+ {"Headphones", NULL, "HPOUTL"},
+ {"Mic", NULL, "AMIC1"},
+};
+
+static void mfld_jack_check(unsigned int intr_status)
+{
+ struct mfld_jack_data jack_data;
+
+ if (!mfld_codec)
+ return;
+
+ jack_data.mfld_jack = &mfld_jack;
+ jack_data.intr_id = intr_status;
+
+ sn95031_jack_detection(mfld_codec, &jack_data);
+ /* TODO: add american headset detection post gpiolib support */
+}
+
+static int mfld_init(struct snd_soc_pcm_runtime *runtime)
+{
+ struct snd_soc_dapm_context *dapm = &runtime->card->dapm;
+ int ret_val;
+
+ /* default is earpiece pin, userspace sets it explcitly */
+ snd_soc_dapm_disable_pin(dapm, "Headphones");
+ /* default is lineout NC, userspace sets it explcitly */
+ snd_soc_dapm_disable_pin(dapm, "LINEOUTL");
+ snd_soc_dapm_disable_pin(dapm, "LINEOUTR");
+ lo_dac = 3;
+ hs_switch = 0;
+ /* we dont use linein in this so set to NC */
+ snd_soc_dapm_disable_pin(dapm, "LINEINL");
+ snd_soc_dapm_disable_pin(dapm, "LINEINR");
+
+ /* Headset and button jack detection */
+ ret_val = snd_soc_card_jack_new(runtime->card,
+ "Intel(R) MID Audio Jack", SND_JACK_HEADSET |
+ SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack,
+ mfld_jack_pins, ARRAY_SIZE(mfld_jack_pins));
+ if (ret_val) {
+ pr_err("jack creation failed\n");
+ return ret_val;
+ }
+
+ ret_val = snd_soc_jack_add_zones(&mfld_jack,
+ ARRAY_SIZE(mfld_zones), mfld_zones);
+ if (ret_val) {
+ pr_err("adding jack zones failed\n");
+ return ret_val;
+ }
+
+ mfld_codec = runtime->codec;
+
+ /* we want to check if anything is inserted at boot,
+ * so send a fake event to codec and it will read adc
+ * to find if anything is there or not */
+ mfld_jack_check(MFLD_JACK_INSERT);
+ return ret_val;
+}
+
+static struct snd_soc_dai_link mfld_msic_dailink[] = {
+ {
+ .name = "Medfield Headset",
+ .stream_name = "Headset",
+ .cpu_dai_name = "Headset-cpu-dai",
+ .codec_dai_name = "SN95031 Headset",
+ .codec_name = "sn95031",
+ .platform_name = "sst-platform",
+ .init = mfld_init,
+ },
+ {
+ .name = "Medfield Speaker",
+ .stream_name = "Speaker",
+ .cpu_dai_name = "Speaker-cpu-dai",
+ .codec_dai_name = "SN95031 Speaker",
+ .codec_name = "sn95031",
+ .platform_name = "sst-platform",
+ .init = NULL,
+ },
+ {
+ .name = "Medfield Vibra",
+ .stream_name = "Vibra1",
+ .cpu_dai_name = "Vibra1-cpu-dai",
+ .codec_dai_name = "SN95031 Vibra1",
+ .codec_name = "sn95031",
+ .platform_name = "sst-platform",
+ .init = NULL,
+ },
+ {
+ .name = "Medfield Haptics",
+ .stream_name = "Vibra2",
+ .cpu_dai_name = "Vibra2-cpu-dai",
+ .codec_dai_name = "SN95031 Vibra2",
+ .codec_name = "sn95031",
+ .platform_name = "sst-platform",
+ .init = NULL,
+ },
+ {
+ .name = "Medfield Compress",
+ .stream_name = "Speaker",
+ .cpu_dai_name = "Compress-cpu-dai",
+ .codec_dai_name = "SN95031 Speaker",
+ .codec_name = "sn95031",
+ .platform_name = "sst-platform",
+ .init = NULL,
+ },
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_mfld = {
+ .name = "medfield_audio",
+ .owner = THIS_MODULE,
+ .dai_link = mfld_msic_dailink,
+ .num_links = ARRAY_SIZE(mfld_msic_dailink),
+
+ .controls = mfld_snd_controls,
+ .num_controls = ARRAY_SIZE(mfld_snd_controls),
+ .dapm_widgets = mfld_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(mfld_widgets),
+ .dapm_routes = mfld_map,
+ .num_dapm_routes = ARRAY_SIZE(mfld_map),
+};
+
+static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev)
+{
+ struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev;
+
+ memcpy_fromio(&mc_private->interrupt_status,
+ ((void *)(mc_private->int_base)),
+ sizeof(u8));
+ return IRQ_WAKE_THREAD;
+}
+
+static irqreturn_t snd_mfld_jack_detection(int irq, void *data)
+{
+ struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data;
+
+ mfld_jack_check(mc_drv_ctx->interrupt_status);
+
+ return IRQ_HANDLED;
+}
+
+static int snd_mfld_mc_probe(struct platform_device *pdev)
+{
+ int ret_val = 0, irq;
+ struct mfld_mc_private *mc_drv_ctx;
+ struct resource *irq_mem;
+
+ pr_debug("snd_mfld_mc_probe called\n");
+
+ /* retrive the irq number */
+ irq = platform_get_irq(pdev, 0);
+
+ /* audio interrupt base of SRAM location where
+ * interrupts are stored by System FW */
+ mc_drv_ctx = devm_kzalloc(&pdev->dev, sizeof(*mc_drv_ctx), GFP_ATOMIC);
+ if (!mc_drv_ctx) {
+ pr_err("allocation failed\n");
+ return -ENOMEM;
+ }
+
+ irq_mem = platform_get_resource_byname(
+ pdev, IORESOURCE_MEM, "IRQ_BASE");
+ if (!irq_mem) {
+ pr_err("no mem resource given\n");
+ return -ENODEV;
+ }
+ mc_drv_ctx->int_base = devm_ioremap_nocache(&pdev->dev, irq_mem->start,
+ resource_size(irq_mem));
+ if (!mc_drv_ctx->int_base) {
+ pr_err("Mapping of cache failed\n");
+ return -ENOMEM;
+ }
+ /* register for interrupt */
+ ret_val = devm_request_threaded_irq(&pdev->dev, irq,
+ snd_mfld_jack_intr_handler,
+ snd_mfld_jack_detection,
+ IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx);
+ if (ret_val) {
+ pr_err("cannot register IRQ\n");
+ return ret_val;
+ }
+ /* register the soc card */
+ snd_soc_card_mfld.dev = &pdev->dev;
+ ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_mfld);
+ if (ret_val) {
+ pr_debug("snd_soc_register_card failed %d\n", ret_val);
+ return ret_val;
+ }
+ platform_set_drvdata(pdev, mc_drv_ctx);
+ pr_debug("successfully exited probe\n");
+ return 0;
+}
+
+static struct platform_driver snd_mfld_mc_driver = {
+ .driver = {
+ .name = "msic_audio",
+ },
+ .probe = snd_mfld_mc_probe,
+};
+
+module_platform_driver(snd_mfld_mc_driver);
+
+MODULE_DESCRIPTION("ASoC Intel(R) MID Machine driver");
+MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>");
+MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:msic-audio");
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-28 17:35:59 +0100