linux/sound/soc/intel/boards/cht_bsw_rt5645.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

400 lines
11 KiB
C

/*
* 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/acpi.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 "../atom/sst-atom-controls.h"
#define CHT_PLAT_CLK_3_HZ 19200000
#define CHT_CODEC_DAI "rt5645-aif1"
struct cht_acpi_card {
char *codec_id;
int codec_type;
struct snd_soc_card *soc_card;
};
struct cht_mc_private {
struct snd_soc_jack jack;
struct cht_acpi_card *acpi_card;
};
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 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_rt5645_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_soc_dapm_route cht_rt5650_audio_map[] = {
{"IN1P", NULL, "Headset Mic"},
{"IN1N", NULL, "Headset Mic"},
{"DMIC L2", NULL, "Int Mic"},
{"DMIC R2", 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 struct snd_soc_jack_pin cht_bsw_jack_pins[] = {
{
.pin = "Headphone",
.mask = SND_JACK_HEADPHONE,
},
{
.pin = "Headset Mic",
.mask = SND_JACK_MICROPHONE,
},
};
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;
int jack_type;
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;
}
if (ctx->acpi_card->codec_type == CODEC_TYPE_RT5650)
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_type, &ctx->jack,
cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins));
if (ret) {
dev_err(runtime->dev, "Headset jack creation failed %d\n", ret);
return ret;
}
rt5645_set_jack_detect(codec, &ctx->jack, &ctx->jack, &ctx->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 int cht_aif1_startup(struct snd_pcm_substream *substream)
{
return snd_pcm_hw_constraint_single(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE, 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",
.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",
},
/* CODEC<->CODEC link */
/* back ends */
{
.name = "SSP2-Codec",
.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,
.nonatomic = true,
.dpcm_playback = 1,
.dpcm_capture = 1,
.ops = &cht_be_ssp2_ops,
},
};
/* SoC card */
static struct snd_soc_card snd_soc_card_chtrt5645 = {
.name = "chtrt5645",
.owner = THIS_MODULE,
.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_rt5645_audio_map,
.num_dapm_routes = ARRAY_SIZE(cht_rt5645_audio_map),
.controls = cht_mc_controls,
.num_controls = ARRAY_SIZE(cht_mc_controls),
};
static struct snd_soc_card snd_soc_card_chtrt5650 = {
.name = "chtrt5650",
.owner = THIS_MODULE,
.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_rt5650_audio_map,
.num_dapm_routes = ARRAY_SIZE(cht_rt5650_audio_map),
.controls = cht_mc_controls,
.num_controls = ARRAY_SIZE(cht_mc_controls),
};
static struct cht_acpi_card snd_soc_cards[] = {
{"10EC5645", CODEC_TYPE_RT5645, &snd_soc_card_chtrt5645},
{"10EC5650", CODEC_TYPE_RT5650, &snd_soc_card_chtrt5650},
};
static int snd_cht_mc_probe(struct platform_device *pdev)
{
int ret_val = 0;
int i;
struct cht_mc_private *drv;
struct snd_soc_card *card = snd_soc_cards[0].soc_card;
char codec_name[16];
drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC);
if (!drv)
return -ENOMEM;
for (i = 0; i < ARRAY_SIZE(snd_soc_cards); i++) {
if (acpi_dev_found(snd_soc_cards[i].codec_id)) {
dev_dbg(&pdev->dev,
"found codec %s\n", snd_soc_cards[i].codec_id);
card = snd_soc_cards[i].soc_card;
drv->acpi_card = &snd_soc_cards[i];
break;
}
}
card->dev = &pdev->dev;
sprintf(codec_name, "i2c-%s:00", drv->acpi_card->codec_id);
/* set correct codec name */
for (i = 0; i < ARRAY_SIZE(cht_dailink); i++)
if (!strcmp(card->dai_link[i].codec_name, "i2c-10EC5645:00"))
card->dai_link[i].codec_name = kstrdup(codec_name, GFP_KERNEL);
snd_soc_card_set_drvdata(card, drv);
ret_val = devm_snd_soc_register_card(&pdev->dev, card);
if (ret_val) {
dev_err(&pdev->dev,
"snd_soc_register_card failed %d\n", ret_val);
return ret_val;
}
platform_set_drvdata(pdev, card);
return ret_val;
}
static struct platform_driver snd_cht_mc_driver = {
.driver = {
.name = "cht-bsw-rt5645",
},
.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");