From ace9429bb58fd418f0c81d4c2835699bddf6bde6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Thu, 11 Apr 2024 10:27:49 +0200 Subject: Adding upstream version 6.6.15. Signed-off-by: Daniel Baumann --- sound/soc/intel/boards/bdw_rt286.c | 263 +++++++++++++++++++++++++++++++++++++ 1 file changed, 263 insertions(+) create mode 100644 sound/soc/intel/boards/bdw_rt286.c (limited to 'sound/soc/intel/boards/bdw_rt286.c') diff --git a/sound/soc/intel/boards/bdw_rt286.c b/sound/soc/intel/boards/bdw_rt286.c new file mode 100644 index 0000000000..036579331d --- /dev/null +++ b/sound/soc/intel/boards/bdw_rt286.c @@ -0,0 +1,263 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Sound card driver for Intel Broadwell Wildcat Point with Realtek 286 + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "../../codecs/rt286.h" + +static struct snd_soc_jack card_headset; + +static struct snd_soc_jack_pin card_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new card_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), +}; + +static const struct snd_soc_dapm_widget card_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC1", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_LINE("Line Jack", NULL), +}; + +static const struct snd_soc_dapm_route card_routes[] = { + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + {"Headphone Jack", NULL, "HPO Pin"}, + + {"MIC1", NULL, "Mic Jack"}, + {"LINE1", NULL, "Line Jack"}, + + {"DMIC1 Pin", NULL, "DMIC1"}, + {"DMIC2 Pin", NULL, "DMIC2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static int codec_link_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component; + int ret; + + ret = snd_soc_card_jack_new_pins(rtd->card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, + &card_headset, card_headset_pins, + ARRAY_SIZE(card_headset_pins)); + if (ret) + return ret; + + return snd_soc_component_set_jack(codec, &card_headset, NULL); +} + +static void codec_link_exit(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *codec = asoc_rtd_to_codec(rtd, 0)->component; + + snd_soc_component_set_jack(codec, NULL, NULL); +} + +static int codec_link_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + + /* The ADSP will convert the FE rate to 48kHz, stereo. */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + /* Set SSP0 to 16 bit. */ + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static int codec_link_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "set codec sysclk failed: %d\n", ret); + return ret; + } + + return ret; +} + +static const struct snd_soc_ops codec_link_ops = { + .hw_params = codec_link_hw_params, +}; + +SND_SOC_DAILINK_DEF(system, DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); +SND_SOC_DAILINK_DEF(offload0, DAILINK_COMP_ARRAY(COMP_CPU("Offload0 Pin"))); +SND_SOC_DAILINK_DEF(offload1, DAILINK_COMP_ARRAY(COMP_CPU("Offload1 Pin"))); +SND_SOC_DAILINK_DEF(loopback, DAILINK_COMP_ARRAY(COMP_CPU("Loopback Pin"))); + +SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); +SND_SOC_DAILINK_DEF(platform, DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); +SND_SOC_DAILINK_DEF(codec, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT343A:00", "rt286-aif1"))); +SND_SOC_DAILINK_DEF(ssp0_port, DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port"))); + +static struct snd_soc_dai_link card_dai_links[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback/Capture", + .nonatomic = 1, + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(system, dummy, platform), + }, + { + .name = "Offload0", + .stream_name = "Offload0 Playback", + .nonatomic = 1, + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + SND_SOC_DAILINK_REG(offload0, dummy, platform), + }, + { + .name = "Offload1", + .stream_name = "Offload1 Playback", + .nonatomic = 1, + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + SND_SOC_DAILINK_REG(offload1, dummy, platform), + }, + { + .name = "Loopback PCM", + .stream_name = "Loopback", + .nonatomic = 1, + .dynamic = 1, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(loopback, dummy, platform), + }, + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .id = 0, + .nonatomic = 1, + .no_pcm = 1, + .init = codec_link_init, + .exit = codec_link_exit, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBC_CFC, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = codec_link_hw_params_fixup, + .ops = &codec_link_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(ssp0_port, codec, platform), + }, +}; + +static int card_suspend_pre(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, "rt286-aif1"); + + if (!codec_dai) + return 0; + + return snd_soc_component_set_jack(codec_dai->component, NULL, NULL); +} + +static int card_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, "rt286-aif1"); + + if (!codec_dai) + return 0; + + return snd_soc_component_set_jack(codec_dai->component, &card_headset, NULL); +} + +static struct snd_soc_card bdw_rt286_card = { + .owner = THIS_MODULE, + .suspend_pre = card_suspend_pre, + .resume_post = card_resume_post, + .dai_link = card_dai_links, + .num_links = ARRAY_SIZE(card_dai_links), + .controls = card_controls, + .num_controls = ARRAY_SIZE(card_controls), + .dapm_widgets = card_widgets, + .num_dapm_widgets = ARRAY_SIZE(card_widgets), + .dapm_routes = card_routes, + .num_dapm_routes = ARRAY_SIZE(card_routes), + .fully_routed = true, +}; + +/* Use space before codec name to simplify card ID, and simplify driver name. */ +#define SOF_CARD_NAME "bdw rt286" /* card name will be 'sof-bdw rt286' */ +#define SOF_DRIVER_NAME "SOF" + +#define CARD_NAME "broadwell-rt286" + +static int bdw_rt286_probe(struct platform_device *pdev) +{ + struct snd_soc_acpi_mach *mach; + struct device *dev = &pdev->dev; + int ret; + + bdw_rt286_card.dev = dev; + mach = dev_get_platdata(dev); + + ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt286_card, mach->mach_params.platform); + if (ret) + return ret; + + if (snd_soc_acpi_sof_parent(dev)) { + bdw_rt286_card.name = SOF_CARD_NAME; + bdw_rt286_card.driver_name = SOF_DRIVER_NAME; + } else { + bdw_rt286_card.name = CARD_NAME; + } + + return devm_snd_soc_register_card(dev, &bdw_rt286_card); +} + +static struct platform_driver bdw_rt286_driver = { + .probe = bdw_rt286_probe, + .driver = { + .name = "bdw_rt286", + .pm = &snd_soc_pm_ops + }, +}; + +module_platform_driver(bdw_rt286_driver) + +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Sound card driver for Intel Broadwell Wildcat Point with Realtek 286"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:bdw_rt286"); -- cgit v1.2.3