mirror of https://gitee.com/openkylin/linux.git
ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_rt5672
Add machine driver for two Intel Cherryview-based platforms, Cherrytrail and Braswell, with RT5672 codec. Signed-off-by: Mengdong Lin <mengdong.lin@intel.com> Acked-by: Vinod Koul <vinod.koul@intel.com> Signed-off-by: Mark Brown <broonie@kernel.org>
This commit is contained in:
Mengdong Lin
Mark Brown
parent
1a28fc190c
commit
026da220c5
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@ -98,3 +98,15 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH
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used as alsa device in audio substem in Intel(R) MID devices
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Say Y if you have such a device
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If unsure select "N".
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config SND_SOC_INTEL_CHT_BSW_RT5672_MACH
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tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec"
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depends on X86_INTEL_LPSS
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select SND_SOC_RT5670
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select SND_SST_MFLD_PLATFORM
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select SND_SST_IPC_ACPI
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help
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This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell
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platforms with RT5672 audio codec.
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Say Y if you have such a device
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If unsure select "N".
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@ -27,12 +27,14 @@ snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
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snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
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snd-soc-sst-broadwell-objs := broadwell.o
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snd-soc-sst-bytcr-dpcm-rt5640-objs := bytcr_dpcm_rt5640.o
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snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
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obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
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obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o
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obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o
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obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
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obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-dpcm-rt5640.o
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obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o
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# DSP driver
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obj-$(CONFIG_SND_SST_IPC) += sst/
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@ -0,0 +1,285 @@
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/*
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* cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms
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* Cherrytrail and Braswell, with RT5672 codec.
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*
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* Copyright (C) 2014 Intel Corp
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* Author: Subhransu S. Prusty <subhransu.s.prusty@intel.com>
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* Mengdong Lin <mengdong.lin@intel.com>
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; version 2 of the License.
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*
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* This program is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* General Public License for more details.
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*/
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#include <linux/module.h>
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#include <linux/platform_device.h>
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#include <linux/slab.h>
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#include <sound/pcm.h>
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#include <sound/pcm_params.h>
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#include <sound/soc.h>
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#include "../codecs/rt5670.h"
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#include "sst-atom-controls.h"
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/* The platform clock #3 outputs 19.2Mhz clock to codec as I2S MCLK */
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#define CHT_PLAT_CLK_3_HZ 19200000
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#define CHT_CODEC_DAI "rt5670-aif1"
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static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
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{
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int i;
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for (i = 0; i < card->num_rtd; i++) {
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struct snd_soc_pcm_runtime *rtd;
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rtd = card->rtd + i;
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if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
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strlen(CHT_CODEC_DAI)))
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return rtd->codec_dai;
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}
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return NULL;
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}
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static int platform_clock_control(struct snd_soc_dapm_widget *w,
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struct snd_kcontrol *k, int event)
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{
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struct snd_soc_dapm_context *dapm = w->dapm;
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struct snd_soc_card *card = dapm->card;
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struct snd_soc_dai *codec_dai;
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codec_dai = cht_get_codec_dai(card);
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if (!codec_dai) {
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dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n");
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return -EIO;
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}
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if (!SND_SOC_DAPM_EVENT_OFF(event))
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return 0;
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/* Set codec sysclk source to its internal clock because codec PLL will
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* be off when idle and MCLK will also be off by ACPI when codec is
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* runtime suspended. Codec needs clock for jack detection and button
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* press.
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*/
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snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_RCCLK,
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0, SND_SOC_CLOCK_IN);
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return 0;
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}
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static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
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SND_SOC_DAPM_HP("Headphone", NULL),
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SND_SOC_DAPM_MIC("Headset Mic", NULL),
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SND_SOC_DAPM_MIC("Int Mic", NULL),
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SND_SOC_DAPM_SPK("Ext Spk", NULL),
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SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0,
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platform_clock_control, SND_SOC_DAPM_POST_PMD),
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};
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static const struct snd_soc_dapm_route cht_audio_map[] = {
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{"IN1P", NULL, "Headset Mic"},
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{"IN1N", NULL, "Headset Mic"},
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{"DMIC L1", NULL, "Int Mic"},
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{"DMIC R1", NULL, "Int Mic"},
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{"Headphone", NULL, "HPOL"},
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{"Headphone", NULL, "HPOR"},
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{"Ext Spk", NULL, "SPOLP"},
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{"Ext Spk", NULL, "SPOLN"},
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{"Ext Spk", NULL, "SPORP"},
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{"Ext Spk", NULL, "SPORN"},
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{"AIF1 Playback", NULL, "ssp2 Tx"},
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{"ssp2 Tx", NULL, "codec_out0"},
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{"ssp2 Tx", NULL, "codec_out1"},
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{"codec_in0", NULL, "ssp2 Rx"},
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{"codec_in1", NULL, "ssp2 Rx"},
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{"ssp2 Rx", NULL, "AIF1 Capture"},
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{"Headphone", NULL, "Platform Clock"},
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{"Headset Mic", NULL, "Platform Clock"},
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{"Int Mic", NULL, "Platform Clock"},
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{"Ext Spk", NULL, "Platform Clock"},
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};
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static const struct snd_kcontrol_new cht_mc_controls[] = {
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SOC_DAPM_PIN_SWITCH("Headphone"),
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SOC_DAPM_PIN_SWITCH("Headset Mic"),
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SOC_DAPM_PIN_SWITCH("Int Mic"),
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SOC_DAPM_PIN_SWITCH("Ext Spk"),
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};
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static int cht_aif1_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_dai *codec_dai = rtd->codec_dai;
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int ret;
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/* set codec PLL source to the 19.2MHz platform clock (MCLK) */
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ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK,
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CHT_PLAT_CLK_3_HZ, params_rate(params) * 512);
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if (ret < 0) {
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dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
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return ret;
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}
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/* set codec sysclk source to PLL */
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ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1,
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params_rate(params) * 512,
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SND_SOC_CLOCK_IN);
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if (ret < 0) {
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dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
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return ret;
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}
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return 0;
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}
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static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
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{
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int ret;
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struct snd_soc_dai *codec_dai = runtime->codec_dai;
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/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
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ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
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if (ret < 0) {
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dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret);
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return ret;
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}
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return 0;
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}
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static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
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struct snd_pcm_hw_params *params)
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{
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struct snd_interval *rate = hw_param_interval(params,
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SNDRV_PCM_HW_PARAM_RATE);
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struct snd_interval *channels = hw_param_interval(params,
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SNDRV_PCM_HW_PARAM_CHANNELS);
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/* The DSP will covert the FE rate to 48k, stereo, 24bits */
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rate->min = rate->max = 48000;
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channels->min = channels->max = 2;
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/* set SSP2 to 24-bit */
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snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT -
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SNDRV_PCM_HW_PARAM_FIRST_MASK],
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SNDRV_PCM_FORMAT_S24_LE);
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return 0;
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}
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static unsigned int rates_48000[] = {
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48000,
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};
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static struct snd_pcm_hw_constraint_list constraints_48000 = {
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.count = ARRAY_SIZE(rates_48000),
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.list = rates_48000,
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};
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static int cht_aif1_startup(struct snd_pcm_substream *substream)
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{
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return snd_pcm_hw_constraint_list(substream->runtime, 0,
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SNDRV_PCM_HW_PARAM_RATE,
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&constraints_48000);
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}
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static struct snd_soc_ops cht_aif1_ops = {
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.startup = cht_aif1_startup,
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};
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static struct snd_soc_ops cht_be_ssp2_ops = {
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.hw_params = cht_aif1_hw_params,
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};
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static struct snd_soc_dai_link cht_dailink[] = {
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/* Front End DAI links */
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[MERR_DPCM_AUDIO] = {
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.name = "Audio Port",
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.stream_name = "Audio",
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.cpu_dai_name = "media-cpu-dai",
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.codec_dai_name = "snd-soc-dummy-dai",
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.codec_name = "snd-soc-dummy",
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.platform_name = "sst-mfld-platform",
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.ignore_suspend = 1,
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.dynamic = 1,
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.dpcm_playback = 1,
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.dpcm_capture = 1,
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.ops = &cht_aif1_ops,
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},
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[MERR_DPCM_COMPR] = {
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.name = "Compressed Port",
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.stream_name = "Compress",
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.cpu_dai_name = "compress-cpu-dai",
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.codec_dai_name = "snd-soc-dummy-dai",
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.codec_name = "snd-soc-dummy",
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.platform_name = "sst-mfld-platform",
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},
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/* Back End DAI links */
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{
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/* SSP2 - Codec */
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.name = "SSP2-Codec",
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.be_id = 1,
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.cpu_dai_name = "ssp2-port",
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.platform_name = "sst-mfld-platform",
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.no_pcm = 1,
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.codec_dai_name = "rt5670-aif1",
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.codec_name = "i2c-10EC5670:00",
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.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF
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| SND_SOC_DAIFMT_CBS_CFS,
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.init = cht_codec_init,
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.be_hw_params_fixup = cht_codec_fixup,
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.ignore_suspend = 1,
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.dpcm_playback = 1,
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.dpcm_capture = 1,
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.ops = &cht_be_ssp2_ops,
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},
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};
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/* SoC card */
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static struct snd_soc_card snd_soc_card_cht = {
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.name = "cherrytrailcraudio",
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.dai_link = cht_dailink,
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.num_links = ARRAY_SIZE(cht_dailink),
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.dapm_widgets = cht_dapm_widgets,
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.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
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.dapm_routes = cht_audio_map,
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.num_dapm_routes = ARRAY_SIZE(cht_audio_map),
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.controls = cht_mc_controls,
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.num_controls = ARRAY_SIZE(cht_mc_controls),
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};
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static int snd_cht_mc_probe(struct platform_device *pdev)
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{
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int ret_val = 0;
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/* register the soc card */
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snd_soc_card_cht.dev = &pdev->dev;
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ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
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if (ret_val) {
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dev_err(&pdev->dev,
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"snd_soc_register_card failed %d\n", ret_val);
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return ret_val;
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}
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platform_set_drvdata(pdev, &snd_soc_card_cht);
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return ret_val;
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}
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static struct platform_driver snd_cht_mc_driver = {
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.driver = {
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.owner = THIS_MODULE,
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.name = "cht-bsw-rt5672",
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.pm = &snd_soc_pm_ops,
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},
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.probe = snd_cht_mc_probe,
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};
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module_platform_driver(snd_cht_mc_driver);
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MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver");
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MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin");
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MODULE_LICENSE("GPL v2");
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MODULE_ALIAS("platform:cht-bsw-rt5672");
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