As of commit: ASoC: soc-core: care .ignore_suspend for Component suspend function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend' flag for dai links. While BE dai link for System Pin is supposed to follow standard suspend-resume flow, appended 'ignore_suspend' flag disturbs that flow and causes audio to break right after resume. Remove the flag to address this. Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Cc: Dominik Brodowski <linux@dominikbrodowski.net> Cc: Mark Brown <broonie@kernel.org> Link: https://lore.kernel.org/r/20200319204947.18963-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
		
			
				
	
	
		
			328 lines
		
	
	
		
			8.7 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			328 lines
		
	
	
		
			8.7 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| // SPDX-License-Identifier: GPL-2.0-only
 | |
| /*
 | |
|  * ASoC machine driver for Intel Broadwell platforms with RT5650 codec
 | |
|  *
 | |
|  * Copyright 2019, The Chromium OS Authors.  All rights reserved.
 | |
|  */
 | |
| 
 | |
| #include <linux/delay.h>
 | |
| #include <linux/gpio/consumer.h>
 | |
| #include <linux/module.h>
 | |
| #include <linux/platform_device.h>
 | |
| #include <sound/core.h>
 | |
| #include <sound/jack.h>
 | |
| #include <sound/pcm.h>
 | |
| #include <sound/pcm_params.h>
 | |
| #include <sound/soc.h>
 | |
| #include <sound/soc-acpi.h>
 | |
| 
 | |
| #include "../common/sst-dsp.h"
 | |
| #include "../haswell/sst-haswell-ipc.h"
 | |
| 
 | |
| #include "../../codecs/rt5645.h"
 | |
| 
 | |
| struct bdw_rt5650_priv {
 | |
| 	struct gpio_desc *gpio_hp_en;
 | |
| 	struct snd_soc_component *component;
 | |
| };
 | |
| 
 | |
| static const struct snd_soc_dapm_widget bdw_rt5650_widgets[] = {
 | |
| 	SND_SOC_DAPM_HP("Headphone", NULL),
 | |
| 	SND_SOC_DAPM_SPK("Speaker", NULL),
 | |
| 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
 | |
| 	SND_SOC_DAPM_MIC("DMIC Pair1", NULL),
 | |
| 	SND_SOC_DAPM_MIC("DMIC Pair2", NULL),
 | |
| };
 | |
| 
 | |
| static const struct snd_soc_dapm_route bdw_rt5650_map[] = {
 | |
| 	/* Speakers */
 | |
| 	{"Speaker", NULL, "SPOL"},
 | |
| 	{"Speaker", NULL, "SPOR"},
 | |
| 
 | |
| 	/* Headset jack connectors */
 | |
| 	{"Headphone", NULL, "HPOL"},
 | |
| 	{"Headphone", NULL, "HPOR"},
 | |
| 	{"IN1P", NULL, "Headset Mic"},
 | |
| 	{"IN1N", NULL, "Headset Mic"},
 | |
| 
 | |
| 	/* Digital MICs
 | |
| 	 * DMIC Pair1 are the two DMICs connected on the DMICN1 connector.
 | |
| 	 * DMIC Pair2 are the two DMICs connected on the DMICN2 connector.
 | |
| 	 * Facing the camera, DMIC Pair1 are on the left side, DMIC Pair2
 | |
| 	 * are on the right side.
 | |
| 	 */
 | |
| 	{"DMIC L1", NULL, "DMIC Pair1"},
 | |
| 	{"DMIC R1", NULL, "DMIC Pair1"},
 | |
| 	{"DMIC L2", NULL, "DMIC Pair2"},
 | |
| 	{"DMIC R2", NULL, "DMIC Pair2"},
 | |
| 
 | |
| 	/* CODEC BE connections */
 | |
| 	{"SSP0 CODEC IN", NULL, "AIF1 Capture"},
 | |
| 	{"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
 | |
| };
 | |
| 
 | |
| static const struct snd_kcontrol_new bdw_rt5650_controls[] = {
 | |
| 	SOC_DAPM_PIN_SWITCH("Speaker"),
 | |
| 	SOC_DAPM_PIN_SWITCH("Headphone"),
 | |
| 	SOC_DAPM_PIN_SWITCH("Headset Mic"),
 | |
| 	SOC_DAPM_PIN_SWITCH("DMIC Pair1"),
 | |
| 	SOC_DAPM_PIN_SWITCH("DMIC Pair2"),
 | |
| };
 | |
| 
 | |
| 
 | |
| static struct snd_soc_jack headphone_jack;
 | |
| static struct snd_soc_jack mic_jack;
 | |
| 
 | |
| static struct snd_soc_jack_pin headphone_jack_pin = {
 | |
| 	.pin	= "Headphone",
 | |
| 	.mask	= SND_JACK_HEADPHONE,
 | |
| };
 | |
| 
 | |
| static struct snd_soc_jack_pin mic_jack_pin = {
 | |
| 	.pin	= "Headset Mic",
 | |
| 	.mask	= SND_JACK_MICROPHONE,
 | |
| };
 | |
| 
 | |
| 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, max 4-channels */
 | |
| 	rate->min = rate->max = 48000;
 | |
| 	channels->min = 2;
 | |
| 	channels->max = 4;
 | |
| 
 | |
| 	/* set SSP0 to 24 bit */
 | |
| 	snd_mask_set_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT),
 | |
| 			    SNDRV_PCM_FORMAT_S24_LE);
 | |
| 
 | |
| 	return 0;
 | |
| }
 | |
| 
 | |
| static int bdw_rt5650_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 = asoc_rtd_to_codec(rtd, 0);
 | |
| 	int ret;
 | |
| 
 | |
| 	/* Workaround: set codec PLL to 19.2MHz that PLL source is
 | |
| 	 * from MCLK(24MHz) to conform 2.4MHz DMIC clock.
 | |
| 	 */
 | |
| 	ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
 | |
| 		24000000, 19200000);
 | |
| 	if (ret < 0) {
 | |
| 		dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
 | |
| 		return ret;
 | |
| 	}
 | |
| 
 | |
| 	/* The actual MCLK freq is 24MHz. The codec is told that MCLK is
 | |
| 	 * 24.576MHz to satisfy the requirement of rl6231_get_clk_info.
 | |
| 	 * ASRC is enabled on AD and DA filters to ensure good audio quality.
 | |
| 	 */
 | |
| 	ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, 24576000,
 | |
| 		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 bdw_rt5650_ops = {
 | |
| 	.hw_params = bdw_rt5650_hw_params,
 | |
| };
 | |
| 
 | |
| #if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
 | |
| static int bdw_rt5650_rtd_init(struct snd_soc_pcm_runtime *rtd)
 | |
| {
 | |
| 	struct snd_soc_component *component =
 | |
| 		snd_soc_rtdcom_lookup(rtd, DRV_NAME);
 | |
| 	struct sst_pdata *pdata = dev_get_platdata(component->dev);
 | |
| 	struct sst_hsw *broadwell = pdata->dsp;
 | |
| 	int ret;
 | |
| 
 | |
| 	/* Set ADSP SSP port settings
 | |
| 	 * clock_divider = 4 means BCLK = MCLK/5 = 24MHz/5 = 4.8MHz
 | |
| 	 */
 | |
| 	ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0,
 | |
| 		SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
 | |
| 		SST_HSW_DEVICE_TDM_CLOCK_MASTER, 4);
 | |
| 	if (ret < 0) {
 | |
| 		dev_err(rtd->dev, "error: failed to set device config\n");
 | |
| 		return ret;
 | |
| 	}
 | |
| 
 | |
| 	return 0;
 | |
| }
 | |
| #endif
 | |
| 
 | |
| static int bdw_rt5650_init(struct snd_soc_pcm_runtime *rtd)
 | |
| {
 | |
| 	struct bdw_rt5650_priv *bdw_rt5650 =
 | |
| 		snd_soc_card_get_drvdata(rtd->card);
 | |
| 	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
 | |
| 	struct snd_soc_component *component = codec_dai->component;
 | |
| 	int ret;
 | |
| 
 | |
| 	/* Enable codec ASRC function for Stereo DAC/Stereo1 ADC/DMIC/I2S1.
 | |
| 	 * The ASRC clock source is clk_i2s1_asrc.
 | |
| 	 */
 | |
| 	rt5645_sel_asrc_clk_src(component,
 | |
| 				RT5645_DA_STEREO_FILTER |
 | |
| 				RT5645_DA_MONO_L_FILTER |
 | |
| 				RT5645_DA_MONO_R_FILTER |
 | |
| 				RT5645_AD_STEREO_FILTER |
 | |
| 				RT5645_AD_MONO_L_FILTER |
 | |
| 				RT5645_AD_MONO_R_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(rtd->dev, "can't set codec TDM slot %d\n", ret);
 | |
| 		return ret;
 | |
| 	}
 | |
| 
 | |
| 	/* Create and initialize headphone jack */
 | |
| 	if (snd_soc_card_jack_new(rtd->card, "Headphone Jack",
 | |
| 			SND_JACK_HEADPHONE, &headphone_jack,
 | |
| 			&headphone_jack_pin, 1)) {
 | |
| 		dev_err(component->dev, "Can't create headphone jack\n");
 | |
| 	}
 | |
| 
 | |
| 	/* Create and initialize mic jack */
 | |
| 	if (snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE,
 | |
| 			&mic_jack, &mic_jack_pin, 1)) {
 | |
| 		dev_err(component->dev, "Can't create mic jack\n");
 | |
| 	}
 | |
| 
 | |
| 	rt5645_set_jack_detect(component, &headphone_jack, &mic_jack, NULL);
 | |
| 
 | |
| 	bdw_rt5650->component = component;
 | |
| 
 | |
| 	return 0;
 | |
| }
 | |
| 
 | |
| /* broadwell digital audio interface glue - connects codec <--> CPU */
 | |
| SND_SOC_DAILINK_DEF(dummy,
 | |
| 	DAILINK_COMP_ARRAY(COMP_DUMMY()));
 | |
| 
 | |
| SND_SOC_DAILINK_DEF(fe,
 | |
| 	DAILINK_COMP_ARRAY(COMP_CPU("System Pin")));
 | |
| 
 | |
| SND_SOC_DAILINK_DEF(platform,
 | |
| 	DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio")));
 | |
| 
 | |
| SND_SOC_DAILINK_DEF(be,
 | |
| 	DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5650:00", "rt5645-aif1")));
 | |
| 
 | |
| #if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
 | |
| SND_SOC_DAILINK_DEF(ssp0_port,
 | |
| 	    DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port")));
 | |
| #endif
 | |
| 
 | |
| static struct snd_soc_dai_link bdw_rt5650_dais[] = {
 | |
| 	/* Front End DAI links */
 | |
| 	{
 | |
| 		.name = "System PCM",
 | |
| 		.stream_name = "System Playback",
 | |
| 		.dynamic = 1,
 | |
| #if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
 | |
| 		.init = bdw_rt5650_rtd_init,
 | |
| #endif
 | |
| 		.trigger = {
 | |
| 			SND_SOC_DPCM_TRIGGER_POST,
 | |
| 			SND_SOC_DPCM_TRIGGER_POST
 | |
| 		},
 | |
| 		.dpcm_playback = 1,
 | |
| 		.dpcm_capture = 1,
 | |
| 		SND_SOC_DAILINK_REG(fe, dummy, platform),
 | |
| 	},
 | |
| 
 | |
| 	/* Back End DAI links */
 | |
| 	{
 | |
| 		/* SSP0 - Codec */
 | |
| 		.name = "Codec",
 | |
| 		.id = 0,
 | |
| 		.no_pcm = 1,
 | |
| 		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
 | |
| 			SND_SOC_DAIFMT_CBS_CFS,
 | |
| 		.ignore_pmdown_time = 1,
 | |
| 		.be_hw_params_fixup = broadwell_ssp0_fixup,
 | |
| 		.ops = &bdw_rt5650_ops,
 | |
| 		.dpcm_playback = 1,
 | |
| 		.dpcm_capture = 1,
 | |
| 		.init = bdw_rt5650_init,
 | |
| #if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
 | |
| 		SND_SOC_DAILINK_REG(dummy, be, dummy),
 | |
| #else
 | |
| 		SND_SOC_DAILINK_REG(ssp0_port, be, platform),
 | |
| #endif
 | |
| 	},
 | |
| };
 | |
| 
 | |
| /* ASoC machine driver for Broadwell DSP + RT5650 */
 | |
| static struct snd_soc_card bdw_rt5650_card = {
 | |
| 	.name = "bdw-rt5650",
 | |
| 	.owner = THIS_MODULE,
 | |
| 	.dai_link = bdw_rt5650_dais,
 | |
| 	.num_links = ARRAY_SIZE(bdw_rt5650_dais),
 | |
| 	.dapm_widgets = bdw_rt5650_widgets,
 | |
| 	.num_dapm_widgets = ARRAY_SIZE(bdw_rt5650_widgets),
 | |
| 	.dapm_routes = bdw_rt5650_map,
 | |
| 	.num_dapm_routes = ARRAY_SIZE(bdw_rt5650_map),
 | |
| 	.controls = bdw_rt5650_controls,
 | |
| 	.num_controls = ARRAY_SIZE(bdw_rt5650_controls),
 | |
| 	.fully_routed = true,
 | |
| };
 | |
| 
 | |
| static int bdw_rt5650_probe(struct platform_device *pdev)
 | |
| {
 | |
| 	struct bdw_rt5650_priv *bdw_rt5650;
 | |
| 	struct snd_soc_acpi_mach *mach;
 | |
| 	int ret;
 | |
| 
 | |
| 	bdw_rt5650_card.dev = &pdev->dev;
 | |
| 
 | |
| 	/* Allocate driver private struct */
 | |
| 	bdw_rt5650 = devm_kzalloc(&pdev->dev, sizeof(struct bdw_rt5650_priv),
 | |
| 		GFP_KERNEL);
 | |
| 	if (!bdw_rt5650)
 | |
| 		return -ENOMEM;
 | |
| 
 | |
| 	/* override plaform name, if required */
 | |
| 	mach = pdev->dev.platform_data;
 | |
| 	ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5650_card,
 | |
| 						    mach->mach_params.platform);
 | |
| 
 | |
| 	if (ret)
 | |
| 		return ret;
 | |
| 
 | |
| 	snd_soc_card_set_drvdata(&bdw_rt5650_card, bdw_rt5650);
 | |
| 
 | |
| 	return devm_snd_soc_register_card(&pdev->dev, &bdw_rt5650_card);
 | |
| }
 | |
| 
 | |
| static struct platform_driver bdw_rt5650_audio = {
 | |
| 	.probe = bdw_rt5650_probe,
 | |
| 	.driver = {
 | |
| 		.name = "bdw-rt5650",
 | |
| 		.pm = &snd_soc_pm_ops,
 | |
| 	},
 | |
| };
 | |
| 
 | |
| module_platform_driver(bdw_rt5650_audio)
 | |
| 
 | |
| /* Module information */
 | |
| MODULE_AUTHOR("Ben Zhang <benzh@chromium.org>");
 | |
| MODULE_DESCRIPTION("Intel Broadwell RT5650 machine driver");
 | |
| MODULE_LICENSE("GPL v2");
 | |
| MODULE_ALIAS("platform:bdw-rt5650");
 |