From: Mark Brown Date: Thu, 5 Mar 2015 01:07:23 +0000 (+0000) Subject: Merge remote-tracking branches 'asoc/topic/jack', 'asoc/topic/max98357a', 'asoc/topic... X-Git-Url: https://git.stricted.de/?a=commitdiff_plain;h=256fca9247c4e62408923a0fe861398510cf6bc1;p=GitHub%2FLineageOS%2FG12%2Fandroid_kernel_amlogic_linux-4.9.git Merge remote-tracking branches 'asoc/topic/jack', 'asoc/topic/max98357a', 'asoc/topic/omap' and 'asoc/topic/rt286' into asoc-next --- 256fca9247c4e62408923a0fe861398510cf6bc1 diff --cc sound/soc/intel/cht_bsw_rt5645.c index dd935255a020,0bfca2192ca0,bd29617a9ab9,000000000000,bd29617a9ab9..012227997ed9 mode 100644,100644,100644,000000,100644..100644 --- a/sound/soc/intel/cht_bsw_rt5645.c +++ b/sound/soc/intel/cht_bsw_rt5645.c @@@@@@ -1,324 -1,326 -1,326 -1,0 -1,326 +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 + * N,Harshapriya + * 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 + #include + #include + #include + #include + #include + #include + #include "../codecs/rt5645.h" + #include "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_jack_new(codec, "Headphone Jack", - - - SND_JACK_HEADPHONE, - - - &ctx->hp_jack); + +++ 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_jack_new(codec, "Mic Jack", - - - SND_JACK_MICROPHONE, - - - &ctx->mic_jack); + +++ 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 */ -- - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - -- - SNDRV_PCM_HW_PARAM_FIRST_MASK], -- - SNDRV_PCM_FORMAT_S24_LE); ++++ 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");