| /* |
| * 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" |
| #include "../common/sst-acpi.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; |
| static char cht_bsw_codec_name[16]; |
| |
| /* 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) |
| { |
| struct snd_soc_pcm_runtime *rtd; |
| |
| list_for_each_entry(rtd, &card->rtd_list, list) { |
| 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 int cht_aif1_startup(struct snd_pcm_substream *substream) |
| { |
| return snd_pcm_hw_constraint_single(substream->runtime, |
| SNDRV_PCM_HW_PARAM_RATE, 48000); |
| } |
| |
| static const struct snd_soc_ops cht_aif1_ops = { |
| .startup = cht_aif1_startup, |
| }; |
| |
| static const 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_DEEP_BUFFER] = { |
| .name = "Deep-Buffer Audio Port", |
| .stream_name = "Deep-Buffer Audio", |
| .cpu_dai_name = "deepbuffer-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, |
| .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", |
| .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_component *component; |
| |
| list_for_each_entry(component, &card->component_dev_list, card_list) { |
| if (!strcmp(component->name, "i2c-10EC5670:00")) { |
| struct snd_soc_codec *codec = snd_soc_component_to_codec(component); |
| |
| 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_component *component; |
| |
| list_for_each_entry(component, &card->component_dev_list, card_list) { |
| if (!strcmp(component->name, "i2c-10EC5670:00")) { |
| struct snd_soc_codec *codec = snd_soc_component_to_codec(component); |
| |
| 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", |
| .owner = THIS_MODULE, |
| .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, |
| }; |
| |
| #define RT5672_I2C_DEFAULT "i2c-10EC5670:00" |
| |
| static int snd_cht_mc_probe(struct platform_device *pdev) |
| { |
| int ret_val = 0; |
| struct sst_acpi_mach *mach = pdev->dev.platform_data; |
| const char *i2c_name; |
| int i; |
| |
| strcpy(cht_bsw_codec_name, RT5672_I2C_DEFAULT); |
| |
| /* fixup codec name based on HID */ |
| if (mach) { |
| i2c_name = sst_acpi_find_name_from_hid(mach->id); |
| if (i2c_name) { |
| snprintf(cht_bsw_codec_name, sizeof(cht_bsw_codec_name), |
| "i2c-%s", i2c_name); |
| for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) { |
| if (!strcmp(cht_dailink[i].codec_name, |
| RT5672_I2C_DEFAULT)) { |
| cht_dailink[i].codec_name = |
| cht_bsw_codec_name; |
| break; |
| } |
| } |
| } |
| } |
| |
| /* 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"); |