--- /dev/null
- - - ret = snd_soc_jack_new(codec, "Headphone Jack",
- - - SND_JACK_HEADPHONE,
- - - &ctx->hp_jack);
+ /*
+ * 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 "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, "Mic Jack",
- - - SND_JACK_MICROPHONE,
- - - &ctx->mic_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;
+ }
+
-- - snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT -
-- - SNDRV_PCM_HW_PARAM_FIRST_MASK],
-- - SNDRV_PCM_FORMAT_S24_LE);
+ +++ 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");
projects / GitHub / moto-9609 / android_kernel_motorola_exynos9610.git / commitdiff