ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_rt5672 (026da220) · Commits · e / devices / android_kernel_teracube_2e

sound/soc/intel/Kconfig

+12 −0

Original line number	Diff line number	Diff line
		@@ -98,3 +98,15 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH
		used as alsa device in audio substem in Intel(R) MID devices
		Say Y if you have such a device
		If unsure select "N".

		config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
		tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec"
		depends on X86_INTEL_LPSS
		select SND_SOC_RT5670
		select SND_SST_MFLD_PLATFORM
		select SND_SST_IPC_ACPI
		help
		This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
		platforms with RT5672 audio codec.
		Say Y if you have such a device
		If unsure select "N".

sound/soc/intel/Makefile

+2 −0

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		@@ -27,12 +27,14 @@ 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-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o
		snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.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-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o
		obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o

		# DSP driver
		obj-$(CONFIG_SND_SST_IPC) += sst/

sound/soc/intel/cht_bsw_rt5672.c

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+285 −0

Original line number	Diff line number	Diff line
		/*
		* 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 "../codecs/rt5670.h"
		#include "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 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;

		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.
		*/
		snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK,
		0, 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_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;

		/* 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;
		}

		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 */
		snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT -
		SNDRV_PCM_HW_PARAM_FIRST_MASK],
		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",
		.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",
		},

		/* 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,
		.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,
		.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 = "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),
		};

		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 = {
		.owner = THIS_MODULE,
		.name = "cht-bsw-rt5672",
		.pm = &snd_soc_pm_ops,
		},
		.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");