1 files changed, 324 insertions, 0 deletions

diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c
new file mode 100644
index 000000000000..93bb6711ba3d
--- /dev/null
+++ b/sound/soc/intel/boards/cht_bsw_rt5645.c
@@ -0,0 +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 <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_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_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");