linux/sound/soc/intel/boards/cht_bsw_max98090_ti.c
Linus Torvalds f4c80d5a16 sound updates for 4.7-rc1
This time was again a relatively calm development cycle; most of
 updates are about drivers, and no radical changes are seen in any
 core code.  Here are some highlights:
 
 ALSA core:
 - Continued hardening of ALSA hrtimer
 - A few leak fixes in timer interface
 - Fix poll error handling in PCM and compress
 - Add error propagation in compress API
 - Removal of dead rtctimer driver
 
 HD-audio:
 - Native ELD notify support for i915 HDMI
 - Realtek ALC234 & co support
 - Code refactoring to standardize chmap support
 - Continued development for SKL HDMI core support
 
 Firewire:
 - Apply delayed card registration to all drivers
 - Improved / stabilized the handling of PCM stream start / stop
 - Add tracepoints to dump a part of isochronous packet data
 - Fixed incoming/outgoing packet parameter usages
 - Add support for M-Audio profire series
 
 USB-audio:
 - Fixes for UAC2 clock source
 - SS+ support
 - Workaround for oft-seen repeated sample rate read errors
 
 ASoC:
 - Further slow progress on the topology code
 - Substantial updates and improvements for the da7219, es8328,
   fsl-ssi, Intel and rcar drivers.
 - Compress error handling in WM ADSP driver
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Merge tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This time was again a relatively calm development cycle; most of
  updates are about drivers, and no radical changes are seen in any core
  code.  Here are some highlights:

  ALSA core:
   - Continued hardening of ALSA hrtimer
   - A few leak fixes in timer interface
   - Fix poll error handling in PCM and compress
   - Add error propagation in compress API
   - Removal of dead rtctimer driver

  HD-audio:
   - Native ELD notify support for i915 HDMI
   - Realtek ALC234 & co support
   - Code refactoring to standardize chmap support
   - Continued development for SKL HDMI core support

  Firewire:
   - Apply delayed card registration to all drivers
   - Improved / stabilized the handling of PCM stream start / stop
   - Add tracepoints to dump a part of isochronous packet data
   - Fixed incoming/outgoing packet parameter usages
   - Add support for M-Audio profire series

  USB-audio:
   - Fixes for UAC2 clock source
   - SS+ support
   - Workaround for oft-seen repeated sample rate read errors

  ASoC:
   - Further slow progress on the topology code
   - Substantial updates and improvements for the da7219, es8328,
     fsl-ssi, Intel and rcar drivers.
   - Compress error handling in WM ADSP driver"

* tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (177 commits)
  ALSA: firewire-lib: change a member of event structure to suppress sparse wanings to bool type
  sound: oss: Use setup_timer and mod_timer.
  ASoC: hdac_hdmi: Remove the unused 'timeout' variable
  ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex.
  ASoC: fsl_ssi: Fix channel slipping in Playback at startup
  ASoC: fsl_ssi: Fix samples being dropped at Playback startup
  ASoC: fsl_ssi: Save a dev reference for dev_err() purpose.
  ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk.
  ASoC: fsl_ssi: Real hardware channels max number is 32
  ASoC: pcm5102a: Add support for PCM5102A codec
  ASoC: hdac_hdmi: add link management
  ASoC: Intel: Skylake: add link management
  ALSA: hdac: add link pm and ref counting
  ALSA: au88x0: Fix zero clear of stream->resources
  ASoC: rt298: Add DMI match for Broxton-P reference platform
  ASoC: rt298: fix null deref on acpi driver data
  ASoC: dapm: deprecate MICBIAS widget type
  ALSA: firewire-lib: drop skip argument from helper functions to queue a packet
  ALSA: firewire-lib: add context information to tracepoints
  ALSA: firewire-lib: permit to flush queued packets only in process context for better PCM period granularity
  ...
2016-05-19 13:41:32 -07:00

332 lines
8.9 KiB
C

/*
* cht-bsw-max98090.c - ASoc Machine driver for Intel Cherryview-based
* platforms Cherrytrail and Braswell, with max98090 & TI codec.
*
* Copyright (C) 2015 Intel Corp
* Author: Fang, Yang A <yang.a.fang@intel.com>
* This file is modified from cht_bsw_rt5645.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 <linux/acpi.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include "../../codecs/max98090.h"
#include "../atom/sst-atom-controls.h"
#include "../../codecs/ts3a227e.h"
#define CHT_PLAT_CLK_3_HZ 19200000
#define CHT_CODEC_DAI "HiFi"
struct cht_mc_private {
struct snd_soc_jack jack;
bool ts3a227e_present;
};
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 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),
};
static const struct snd_soc_dapm_route cht_audio_map[] = {
{"IN34", NULL, "Headset Mic"},
{"Headset Mic", NULL, "MICBIAS"},
{"DMICL", NULL, "Int Mic"},
{"Headphone", NULL, "HPL"},
{"Headphone", NULL, "HPR"},
{"Ext Spk", NULL, "SPKL"},
{"Ext Spk", NULL, "SPKR"},
{"HiFi 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, "HiFi Capture"},
};
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;
ret = snd_soc_dai_set_sysclk(codec_dai, M98090_REG_SYSTEM_CLOCK,
CHT_PLAT_CLK_3_HZ, 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_ti_jack_event(struct notifier_block *nb,
unsigned long event, void *data)
{
struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
struct snd_soc_dapm_context *dapm = &jack->card->dapm;
if (event & SND_JACK_MICROPHONE) {
snd_soc_dapm_force_enable_pin(dapm, "SHDN");
snd_soc_dapm_force_enable_pin(dapm, "MICBIAS");
snd_soc_dapm_sync(dapm);
} else {
snd_soc_dapm_disable_pin(dapm, "MICBIAS");
snd_soc_dapm_disable_pin(dapm, "SHDN");
snd_soc_dapm_sync(dapm);
}
return 0;
}
static struct notifier_block cht_jack_nb = {
.notifier_call = cht_ti_jack_event,
};
static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
{
int ret;
int jack_type;
struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
struct snd_soc_jack *jack = &ctx->jack;
/**
* TI supports 4 butons headset detection
* KEY_MEDIA
* KEY_VOICECOMMAND
* KEY_VOLUMEUP
* KEY_VOLUMEDOWN
*/
if (ctx->ts3a227e_present)
jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
SND_JACK_BTN_0 | SND_JACK_BTN_1 |
SND_JACK_BTN_2 | SND_JACK_BTN_3;
else
jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE;
ret = snd_soc_card_jack_new(runtime->card, "Headset Jack",
jack_type, jack, NULL, 0);
if (ret) {
dev_err(runtime->dev, "Headset Jack creation failed %d\n", ret);
return ret;
}
if (ctx->ts3a227e_present)
snd_soc_jack_notifier_register(jack, &cht_jack_nb);
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);
int ret = 0;
unsigned int fmt = 0;
ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16);
if (ret < 0) {
dev_err(rtd->dev, "can't set cpu_dai slot fmt: %d\n", ret);
return ret;
}
fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBS_CFS;
ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt);
if (ret < 0) {
dev_err(rtd->dev, "can't set cpu_dai set fmt: %d\n", ret);
return ret;
}
/* 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 int cht_max98090_headset_init(struct snd_soc_component *component)
{
struct snd_soc_card *card = component->card;
struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card);
return ts3a227e_enable_jack_detect(component, &ctx->jack);
}
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_aux_dev cht_max98090_headset_dev = {
.name = "Headset Chip",
.init = cht_max98090_headset_init,
.codec_name = "i2c-104C227E:00",
};
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",
.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 ends */
{
.name = "SSP2-Codec",
.id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
.codec_dai_name = "HiFi",
.codec_name = "i2c-193C9890:00",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_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,
},
};
/* SoC card */
static struct snd_soc_card snd_soc_card_cht = {
.name = "chtmax98090",
.owner = THIS_MODULE,
.dai_link = cht_dailink,
.num_links = ARRAY_SIZE(cht_dailink),
.aux_dev = &cht_max98090_headset_dev,
.num_aux_devs = 1,
.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;
drv->ts3a227e_present = acpi_dev_found("104C227E");
if (!drv->ts3a227e_present) {
/* no need probe TI jack detection chip */
snd_soc_card_cht.aux_dev = NULL;
snd_soc_card_cht.num_aux_devs = 0;
}
/* register the soc card */
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-max98090",
},
.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 <yang.a.fang@intel.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:cht-bsw-max98090");